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New upstream version 3.5~git20180113

Browse Source
This commit is contained in:
James Cowgill
2018-01-14 00:26:02 +00:00
parent 5cc1c3e3e7
commit 443004e62a
1030 changed files with 64537 additions and 41149 deletions
Download Patch File Download Diff File Expand all files Collapse all files
.gitignore
+38
View File
@@ -0,0 +1,38 @@
*.a
*.o
*.o.*
*.d
*.def
*.dll
*.dylib
*.exe
*.exp
*.gcda
*.gcno
*.h.c
*.ilk
*.lib
*.pc
*.pdb
*.so
*.so.*
*.swp
*.ver
*.version
*.ptx
*.ptx.c
*_g
\#*
.\#*
/.config
/.version
/ffmpeg
/ffplay
/ffprobe
/config.asm
/config.h
/coverage.info
/avversion.h
/lcov/
/src
/mapfile
.travis.yml
+4
View File
@@ -11,6 +11,10 @@ addons:
compiler:
- clang
- gcc
matrix:
exclude:
- os: osx
compiler: gcc
cache:
directories:
- ffmpeg-samples
Changelog
+39
View File
@@ -1,6 +1,45 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>:
- Bitstream filters for editing metadata in H.264, HEVC and MPEG-2 streams
- Dropped support for OpenJPEG versions 2.0 and below. Using OpenJPEG now
requires 2.1 (or later) and pkg-config.
- VDA dropped (use VideoToolbox instead)
- MagicYUV encoder
- Raw AMR-NB and AMR-WB demuxers
- TiVo ty/ty+ demuxer
- Intel QSV-accelerated MJPEG encoding
- PCE support for extended channel layouts in the AAC encoder
- native aptX encoder and decoder
- Raw aptX muxer and demuxer
- NVIDIA NVDEC-accelerated H.264, HEVC, MPEG-1/2/4, VC1, VP8/9 hwaccel decoding
- Intel QSV-accelerated overlay filter
- mcompand audio filter
- acontrast audio filter
- OpenCL overlay filter
- video mix filter
- video normalize filter
- audio lv2 wrapper filter
- VAAPI VP8 decoding
- AMD AMF H.264 and HEVC encoders
- video fillborders filter
- video setrange filter
- nsp demuxer
- support LibreSSL (via libtls)
- AVX-512/ZMM support added
- Dropped support for building for Windows XP. The minimum supported Windows
version is Windows Vista.
- deconvolve video filter
- entropy video filter
- hilbert audio filter source
- aiir audio filter
- aiff: add support for CD-ROM XA ADPCM
- Removed the ffserver program
- Removed the ffmenc and ffmdec muxer and demuxer
- VideoToolbox HEVC encoder and hwaccel
version 3.4:
- deflicker video filter
- doubleweave video filter
MAINTAINERS
+6 -8
View File
@@ -29,9 +29,6 @@ ffplay:
ffprobe:
ffprobe.c Stefano Sabatini
ffserver:
ffserver.c Reynaldo H. Verdejo Pinochet
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
@@ -158,7 +155,7 @@ Codecs:
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
cuvid.c Timo Rothenpieler
cuviddec.c Timo Rothenpieler
dca* foo86
dirac* Rostislav Pehlivanov
dnxhd* Baptiste Coudurier
@@ -170,6 +167,7 @@ Codecs:
eacmv*, eaidct*, eat* Peter Ross
evrc* Paul B Mahol
exif.c, exif.h Thilo Borgmann
exr.c Martin Vignali
ffv1* Michael Niedermayer
ffwavesynth.c Nicolas George
fifo.c Jan Sebechlebsky
@@ -212,7 +210,7 @@ Codecs:
msrle.c Mike Melanson
msvideo1.c Mike Melanson
nuv.c Reimar Doeffinger
nvenc* Timo Rothenpieler
nvdec*, nvenc* Timo Rothenpieler
opus* Rostislav Pehlivanov
paf.* Paul B Mahol
pcx.c Ivo van Poorten
@@ -244,8 +242,7 @@ Codecs:
txd.c Ivo van Poorten
vc2* Rostislav Pehlivanov
vcr1.c Michael Niedermayer
vda_h264_dec.c Xidorn Quan
videotoolboxenc.c Rick Kern
videotoolboxenc.c Rick Kern, Aman Gupta
vima.c Paul B Mahol
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
@@ -272,7 +269,7 @@ Hardware acceleration:
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern
videotoolbox* Rick Kern, Aman Gupta
libavdevice
@@ -587,6 +584,7 @@ FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Ganesh Ajjanagadde C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
James Almer 7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Makefile
+7 -9
View File
@@ -45,12 +45,11 @@ FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
@@ -128,21 +127,18 @@ install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
uninstall: uninstall-libs uninstall-headers uninstall-data
uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(CLEANSUFFIXES)
$(RM) $(CLEANSUFFIXES:%=compat/msvcrt/%)
$(RM) $(CLEANSUFFIXES:%=compat/atomics/pthread/%)
$(RM) $(CLEANSUFFIXES:%=compat/%)
$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES))
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
distclean::
$(RM) $(DISTCLEANSUFFIXES)
distclean:: clean
$(RM) .version avversion.h config.asm config.h mapfile \
ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
version.h libavutil/ffversion.h libavcodec/codec_names.h \
@@ -155,6 +151,7 @@ endif
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
build: all alltools examples testprogs
check: all alltools examples testprogs fate
include $(SRC_PATH)/tests/Makefile
@@ -170,4 +167,5 @@ $(sort $(OBJDIRS)):
# so this saves some time on slow systems.
.SUFFIXES:
.PHONY: all all-yes alltools check *clean config install* testprogs uninstall*
.PHONY: all all-yes alltools build check config testprogs
.PHONY: *clean install* uninstall*
README.md
-2
View File
@@ -21,8 +21,6 @@ such as audio, video, subtitles and related metadata.
* [ffplay](https://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
* [ffprobe](https://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
multimedia content.
* [ffserver](https://ffmpeg.org/ffserver.html) is a multimedia streaming server
for live broadcasts.
* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
## Documentation
RELEASE
+1 -1
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@@ -1 +1 @@
3.4
3.4.git
RELEASE_NOTES
-15
View File
@@ -1,15 +0,0 @@
┌───────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 3.4 "Cantor" │
└───────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 3.4 "Cantor", about 6
months after the release of FFmpeg 3.3.
A complete Changelog is available at the root of the project, and the
complete Git history on http://source.ffmpeg.org.
We hope you will like this release as much as we enjoyed working on it, and
as usual, if you have any questions about it, or any FFmpeg related topic,
feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
on the mailing-lists.
VERSION
-1
View File
@@ -1 +0,0 @@
3.4
compat/cuda/dynlink_loader.h
+9 -9
View File
@@ -59,29 +59,29 @@
#define LOAD_LIBRARY(l, path) \
do { \
if (!((l) = dlopen(path, RTLD_LAZY))) { \
av_log(NULL, AV_LOG_ERROR, "Cannot load %s\n", path); \
av_log(logctx, AV_LOG_ERROR, "Cannot load %s\n", path); \
ret = AVERROR_UNKNOWN; \
goto error; \
} \
av_log(NULL, AV_LOG_TRACE, "Loaded lib: %s\n", path); \
av_log(logctx, AV_LOG_TRACE, "Loaded lib: %s\n", path); \
} while (0)
#define LOAD_SYMBOL(fun, tp, symbol) \
do { \
if (!((f->fun) = (tp*)dlsym(f->lib, symbol))) { \
av_log(NULL, AV_LOG_ERROR, "Cannot load %s\n", symbol); \
av_log(logctx, AV_LOG_ERROR, "Cannot load %s\n", symbol); \
ret = AVERROR_UNKNOWN; \
goto error; \
} \
av_log(NULL, AV_LOG_TRACE, "Loaded sym: %s\n", symbol); \
av_log(logctx, AV_LOG_TRACE, "Loaded sym: %s\n", symbol); \
} while (0)
#define LOAD_SYMBOL_OPT(fun, tp, symbol) \
do { \
if (!((f->fun) = (tp*)dlsym(f->lib, symbol))) { \
av_log(NULL, AV_LOG_DEBUG, "Cannot load optional %s\n", symbol); \
av_log(logctx, AV_LOG_DEBUG, "Cannot load optional %s\n", symbol); \
} else { \
av_log(NULL, AV_LOG_TRACE, "Loaded sym: %s\n", symbol); \
av_log(logctx, AV_LOG_TRACE, "Loaded sym: %s\n", symbol); \
} \
} while (0)
@@ -187,7 +187,7 @@ static inline void nvenc_free_functions(NvencFunctions **functions)
}
#ifdef AV_COMPAT_DYNLINK_CUDA_H
static inline int cuda_load_functions(CudaFunctions **functions)
static inline int cuda_load_functions(CudaFunctions **functions, void *logctx)
{
GENERIC_LOAD_FUNC_PREAMBLE(CudaFunctions, cuda, CUDA_LIBNAME);
@@ -210,7 +210,7 @@ static inline int cuda_load_functions(CudaFunctions **functions)
}
#endif
static inline int cuvid_load_functions(CuvidFunctions **functions)
static inline int cuvid_load_functions(CuvidFunctions **functions, void *logctx)
{
GENERIC_LOAD_FUNC_PREAMBLE(CuvidFunctions, cuvid, NVCUVID_LIBNAME);
@@ -244,7 +244,7 @@ static inline int cuvid_load_functions(CuvidFunctions **functions)
GENERIC_LOAD_FUNC_FINALE(cuvid);
}
static inline int nvenc_load_functions(NvencFunctions **functions)
static inline int nvenc_load_functions(NvencFunctions **functions, void *logctx)
{
GENERIC_LOAD_FUNC_PREAMBLE(NvencFunctions, nvenc, NVENC_LIBNAME);
compat/nvenc/nvEncodeAPI.h
+1 -1
View File
@@ -77,7 +77,7 @@ extern "C" {
typedef RECT NVENC_RECT;
#else
// =========================================================================================
#ifndef GUID
#if !defined(GUID) && !defined(GUID_DEFINED)
/*!
* \struct GUID
* Abstracts the GUID structure for non-windows platforms.
compat/os2threads.h
+8 -6
View File
@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
* Copyright (c) 2011-2017 KO Myung-Hun <komh@chollian.net>
*
* This file is part of FFmpeg.
*
@@ -46,9 +46,11 @@ typedef struct {
typedef void pthread_attr_t;
typedef HMTX pthread_mutex_t;
typedef _fmutex pthread_mutex_t;
typedef void pthread_mutexattr_t;
#define PTHREAD_MUTEX_INITIALIZER _FMUTEX_INITIALIZER
typedef struct {
HEV event_sem;
HEV ack_sem;
@@ -98,28 +100,28 @@ static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
const pthread_mutexattr_t *attr)
{
DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
_fmutex_create(mutex, 0);
return 0;
}
static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
{
DosCloseMutexSem(*(PHMTX)mutex);
_fmutex_close(mutex);
return 0;
}
static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
{
DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
_fmutex_request(mutex, 0);
return 0;
}
static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
{
DosReleaseMutexSem(*(PHMTX)mutex);
_fmutex_release(mutex);
return 0;
}
compat/w32dlfcn.h
+1
View File
@@ -21,6 +21,7 @@
#ifdef _WIN32
#include <windows.h>
#include "config.h"
#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
#include "libavutil/wchar_filename.h"
#endif
compat/w32pthreads.h
+10 -263
View File
@@ -39,11 +39,6 @@
#include <windows.h>
#include <process.h>
#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
#undef MemoryBarrier
#define MemoryBarrier __sync_synchronize
#endif
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/internal.h"
@@ -56,24 +51,15 @@ typedef struct pthread_t {
void *ret;
} pthread_t;
/* the conditional variable api for windows 6.0+ uses critical sections and
* not mutexes */
typedef CRITICAL_SECTION pthread_mutex_t;
/* This is the CONDITION_VARIABLE typedef for using Windows' native
* conditional variables on kernels 6.0+. */
#if HAVE_CONDITION_VARIABLE_PTR
/* use light weight mutex/condition variable API for Windows Vista and later */
typedef SRWLOCK pthread_mutex_t;
typedef CONDITION_VARIABLE pthread_cond_t;
#else
typedef struct pthread_cond_t {
void *Ptr;
} pthread_cond_t;
#endif
#if _WIN32_WINNT >= 0x0600
#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
#endif
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
@@ -114,26 +100,25 @@ static av_unused int pthread_join(pthread_t thread, void **value_ptr)
static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
{
InitializeCriticalSection(m);
InitializeSRWLock(m);
return 0;
}
static inline int pthread_mutex_destroy(pthread_mutex_t *m)
{
DeleteCriticalSection(m);
/* Unlocked SWR locks use no resources */
return 0;
}
static inline int pthread_mutex_lock(pthread_mutex_t *m)
{
EnterCriticalSection(m);
AcquireSRWLockExclusive(m);
return 0;
}
static inline int pthread_mutex_unlock(pthread_mutex_t *m)
{
LeaveCriticalSection(m);
ReleaseSRWLockExclusive(m);
return 0;
}
#if _WIN32_WINNT >= 0x0600
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
@@ -167,7 +152,7 @@ static inline int pthread_cond_broadcast(pthread_cond_t *cond)
static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
SleepConditionVariableCS(cond, mutex, INFINITE);
SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
return 0;
}
@@ -177,242 +162,4 @@ static inline int pthread_cond_signal(pthread_cond_t *cond)
return 0;
}
#else // _WIN32_WINNT < 0x0600
/* atomic init state of dynamically loaded functions */
static LONG w32thread_init_state = 0;
static av_unused void w32thread_init(void);
/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
* compatible to the one used in the native API */
typedef union pthread_once_t {
void * Ptr; ///< For the Windows 6.0+ native functions
LONG state; ///< For the pre-Windows 6.0 compat code
} pthread_once_t;
#define PTHREAD_ONCE_INIT {0}
/* function pointers to init once API on windows 6.0+ kernels */
static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
/* pre-Windows 6.0 compat using a spin-lock */
static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
{
switch (InterlockedCompareExchange(state, 1, 0)) {
/* Initial run */
case 0:
init_routine();
InterlockedExchange(state, 2);
break;
/* Another thread is running init */
case 1:
while (1) {
MemoryBarrier();
if (*state == 2)
break;
Sleep(0);
}
break;
/* Initialization complete */
case 2:
break;
}
}
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
/* Use native functions on Windows 6.0+ */
if (initonce_begin && initonce_complete) {
BOOL pending = FALSE;
initonce_begin(once_control, 0, &pending, NULL);
if (pending)
init_routine();
initonce_complete(once_control, 0, NULL);
return 0;
}
w32thread_once_fallback(&once_control->state, init_routine);
return 0;
}
/* for pre-Windows 6.0 platforms we need to define and use our own condition
* variable and api */
typedef struct win32_cond_t {
pthread_mutex_t mtx_broadcast;
pthread_mutex_t mtx_waiter_count;
volatile int waiter_count;
HANDLE semaphore;
HANDLE waiters_done;
volatile int is_broadcast;
} win32_cond_t;
/* function pointers to conditional variable API on windows 6.0+ kernels */
static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
static void (WINAPI *cond_init)(pthread_cond_t *cond);
static void (WINAPI *cond_signal)(pthread_cond_t *cond);
static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
DWORD milliseconds);
static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
win32_cond_t *win32_cond = NULL;
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
if (cond_init) {
cond_init(cond);
return 0;
}
/* non native condition variables */
win32_cond = (win32_cond_t*)av_mallocz(sizeof(win32_cond_t));
if (!win32_cond)
return ENOMEM;
cond->Ptr = win32_cond;
win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
if (!win32_cond->semaphore)
return ENOMEM;
win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
if (!win32_cond->waiters_done)
return ENOMEM;
pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
return 0;
}
static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
/* native condition variables do not destroy */
if (cond_init)
return 0;
/* non native condition variables */
CloseHandle(win32_cond->semaphore);
CloseHandle(win32_cond->waiters_done);
pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
pthread_mutex_destroy(&win32_cond->mtx_broadcast);
av_freep(&win32_cond);
cond->Ptr = NULL;
return 0;
}
static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
int have_waiter;
if (cond_broadcast) {
cond_broadcast(cond);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = 0;
if (win32_cond->waiter_count) {
win32_cond->is_broadcast = 1;
have_waiter = 1;
}
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
win32_cond->is_broadcast = 0;
} else
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
int last_waiter;
if (cond_wait) {
cond_wait(cond, mutex, INFINITE);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count++;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
// unlock the external mutex
pthread_mutex_unlock(mutex);
WaitForSingleObject(win32_cond->semaphore, INFINITE);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count--;
last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (last_waiter)
SetEvent(win32_cond->waiters_done);
// lock the external mutex
return pthread_mutex_lock(mutex);
}
static av_unused int pthread_cond_signal(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
int have_waiter;
if (cond_signal) {
cond_signal(cond);
return 0;
}
pthread_mutex_lock(&win32_cond->mtx_broadcast);
/* non-native condition variables */
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = win32_cond->waiter_count;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
}
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
#endif
static av_unused void w32thread_init(void)
{
#if _WIN32_WINNT < 0x0600
HMODULE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
/* if one is available, then they should all be available */
cond_init = (void (WINAPI*)(pthread_cond_t *))
GetProcAddress(kernel_dll, "InitializeConditionVariable");
cond_broadcast = (void (WINAPI*)(pthread_cond_t *))
GetProcAddress(kernel_dll, "WakeAllConditionVariable");
cond_signal = (void (WINAPI*)(pthread_cond_t *))
GetProcAddress(kernel_dll, "WakeConditionVariable");
cond_wait = (BOOL (WINAPI*)(pthread_cond_t *, pthread_mutex_t *, DWORD))
GetProcAddress(kernel_dll, "SleepConditionVariableCS");
initonce_begin = (BOOL (WINAPI*)(pthread_once_t *, DWORD, BOOL *, void **))
GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
initonce_complete = (BOOL (WINAPI*)(pthread_once_t *, DWORD, void *))
GetProcAddress(kernel_dll, "InitOnceComplete");
#endif
}
#endif /* COMPAT_W32PTHREADS_H */
compat/windows/makedef
+18 -21
View File
@@ -45,7 +45,11 @@ libname=$(mktemp -u "library").lib
trap 'rm -f -- $libname' EXIT
if [ -n "$AR" ]; then
$AR rcs ${libname} $@ >/dev/null
else
lib -out:${libname} $@ >/dev/null
fi
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
exit 1
@@ -54,23 +58,7 @@ fi
IFS='
'
# Determine if we're building for x86 or x86_64 and
# set the symbol prefix accordingly.
prefix=""
arch=$(dumpbin -headers ${libname} |
tr '\t' ' ' |
grep '^ \+.\+machine \+(.\+)' |
head -1 |
sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
if [ "${arch}" = "x86" ]; then
prefix="_"
else
if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
echo "Unknown machine type." >&2
exit 1
fi
fi
prefix="$EXTERN_PREFIX"
started=0
regex="none"
@@ -112,7 +100,19 @@ for line in $(cat ${vscript} | tr '\t' ' '); do
'
done
dump=$(dumpbin -linkermember:1 ${libname})
if [ -n "$NM" ]; then
# Use eval, since NM="nm -g"
dump=$(eval "$NM --defined-only -g ${libname}" |
grep -v : |
grep -v ^$ |
cut -d' ' -f3 |
sed -e "s/^${prefix}//")
else
dump=$(dumpbin -linkermember:1 ${libname} |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
cut -d' ' -f3)
fi
rm ${libname}
@@ -121,9 +121,6 @@ list=""
for exp in ${regex}; do
list="${list}"'
'$(echo "${dump}" |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
cut -d' ' -f3 |
grep "^${exp}" |
sed -e 's/^/ /')
done
configure
Vendored
+628 -680
View File
File diff suppressed because it is too large Load Diff
doc/.gitignore
+9
View File
@@ -0,0 +1,9 @@
/*.1
/*.3
/*.html
/*.pod
/config.texi
/avoptions_codec.texi
/avoptions_format.texi
/fate.txt
/print_options
doc/APIchanges
+66 -9
View File
@@ -2,19 +2,76 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
since the last major version increase or the API was added.
The last version increases were:
libavcodec: 2015-08-28
libavdevice: 2015-08-28
libavfilter: 2015-08-28
libavformat: 2015-08-28
libavresample: 2015-08-28
libpostproc: 2015-08-28
libswresample: 2015-08-28
libswscale: 2015-08-28
libavutil: 2015-08-28
libavcodec: 2017-10-21
libavdevice: 2017-10-21
libavfilter: 2017-10-21
libavformat: 2017-10-21
libavresample: 2017-10-21
libpostproc: 2017-10-21
libswresample: 2017-10-21
libswscale: 2017-10-21
libavutil: 2017-10-21
API changes, most recent first:
2018-01-xx - xxxxxxx - lavfi 7.11.101 - avfilter.h
Deprecate avfilter_link_get_channels(). Use av_buffersink_get_channels().
2017-xx-xx - xxxxxxx - lavr 4.0.0 - avresample.h
Deprecate the entire library. Merged years ago to provide compatibility
with Libav, it remained unmaintained by the FFmpeg project and duplicated
functionality provided by libswresample.
In order to improve consistency and reduce attack surface, it has been deprecated.
Users of this library are asked to migrate to libswresample, which, as well as
providing more functionality, is faster and has higher accuracy.
2017-xx-xx - xxxxxxx - lavc 58.9.100 - avcodec.h
Deprecate av_lockmgr_register(). You need to build FFmpeg with threading
support enabled to get basic thread-safety (which is the default build
configuration).
2017-12-xx - xxxxxxx - lavu 56.7.100 - cpu.h
AVX-512 flags added.
2017-xx-xx - xxxxxxx - lavc 58.8.100 - avcodec.h
The MediaCodec decoders now support AVCodecContext.hw_device_ctx.
2017-xx-xx - xxxxxxx - lavu 56.6.100 - hwcontext.h hwcontext_mediacodec.h
Add AV_HWDEVICE_TYPE_MEDIACODEC and a new installed header with
MediaCodec-specific hwcontext definitions.
2017-xx-xx - xxxxxxc - lavc 58.7.100 - avcodec.h
Add AV_CODEC_CAP_HARDWARE, AV_CODEC_CAP_HYBRID, and AVCodec.wrapper_name,
and mark all AVCodecs accordingly.
2017-xx-xx - xxxxxxx - lavu 56.4.100 / 56.7.0 - stereo3d.h
Add view field to AVStereo3D structure and AVStereo3DView enum.
2017-xx-xx - xxxxxxx - lavc 58.6.100 - avcodec.h
Add const to AVCodecContext.hwaccel.
2017-11-xx - xxxxxxx - lavc 58.5.100 - avcodec.h
Deprecate user visibility of the AVHWAccel structure and the functions
av_register_hwaccel() and av_hwaccel_next().
2017-11-xx - xxxxxxx - lavc 58.4.100 - avcodec.h
Add AVCodecHWConfig and avcodec_get_hw_config().
2017-11-22 - 3650cb2dfa - lavu 55.3.100 - opencl.h
Remove experimental OpenCL API (av_opencl_*).
2017-11-22 - b25d8ef0a7 - lavu 55.2.100 - hwcontext.h hwcontext_opencl.h
Add AV_HWDEVICE_TYPE_OPENCL and a new installed header with
OpenCL-specific hwcontext definitions.
2017-11-22 - a050f56c09 - lavu 55.1.100 - pixfmt.h
Add AV_PIX_FMT_OPENCL.
2017-11-11 - 48e4eda11d - lavc 58.3.100 - avcodec.h
Add avcodec_get_hw_frames_parameters().
-------- 8< --------- FFmpeg 3.4 was cut here -------- 8< ---------
2017-09-28 - b6cf66ae1c - lavc 57.106.104 - avcodec.h
doc/Doxyfile
+1 -1
View File
@@ -38,7 +38,7 @@ PROJECT_NAME = FFmpeg
# could be handy for archiving the generated documentation or if some version
# control system is used.
PROJECT_NUMBER = 3.4
PROJECT_NUMBER =
# Using the PROJECT_BRIEF tag one can provide an optional one line description
# for a project that appears at the top of each page and should give viewer a
doc/bitstream_filters.texi
+170
View File
@@ -92,6 +92,69 @@ When this option is enabled, the long-term headers are removed from the
bitstream after extraction.
@end table
@section h264_metadata
Modify metadata embedded in an H.264 stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.264 section E.2.1 and
table E-2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.264 section E.2.1
and tables E-3, E-4 and E-5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.264 section
E.2.1 and figure E-1).
@item tick_rate
Set the tick rate (num_units_in_tick / time_scale) in the VUI
parameters. This is the smallest time unit representable in the
stream, and in many cases represents the field rate of the stream
(double the frame rate).
@item fixed_frame_rate_flag
Set whether the stream has fixed framerate - typically this indicates
that the framerate is exactly half the tick rate, but the exact
meaning is dependent on interlacing and the picture structure (see
H.264 section E.2.1 and table E-6).
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the frame cropping offsets in the SPS. These values will replace
the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled or the stream is interlaced
(see H.264 section 7.4.2.1.1).
@item sei_user_data
Insert a string as SEI unregistered user data. The argument must
be of the form @emph{UUID+string}, where the UUID is as hex digits
possibly separated by hyphens, and the string can be anything.
For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
insert the string ``hello'' associated with the given UUID.
@end table
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
@@ -111,6 +174,69 @@ ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
@section h264_redundant_pps
This applies a specific fixup to some Blu-ray streams which contain
redundant PPSs modifying irrelevant parameters of the stream which
confuse other transformations which require correct extradata.
A new single global PPS is created, and all of the redundant PPSs
within the stream are removed.
@section hevc_metadata
Modify metadata embedded in an HEVC stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio in the stream in the VUI parameters.
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.265 section E.3.1 and
table E.2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.265 section E.3.1
and tables E.3, E.4 and E.5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.265 section
E.3.1 and figure E.1).
@item tick_rate
Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
time_scale). Combined with @option{num_ticks_poc_diff_one}, this can
set a constant framerate in the stream. Note that it is likely to be
overridden by container parameters when the stream is in a container.
@item num_ticks_poc_diff_one
Set poc_proportional_to_timing_flag in VPS and VUI and use this value
to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and
E.3.1). Ignored if @option{tick_rate} is not also set.
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the conformance window cropping offsets in the SPS. These values
will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
@end table
@section hevc_mp4toannexb
Convert an HEVC/H.265 bitstream from length prefixed mode to start code
@@ -198,6 +324,42 @@ See also the @ref{text2movsub} filter.
Decompress non-standard compressed MP3 audio headers.
@section mpeg2_metadata
Modify metadata embedded in an MPEG-2 stream.
@table @option
@item display_aspect_ratio
Set the display aspect ratio in the stream.
The following fixed values are supported:
@table @option
@item 4/3
@item 16/9
@item 221/100
@end table
Any other value will result in square pixels being signalled instead
(see H.262 section 6.3.3 and table 6-3).
@item frame_rate
Set the frame rate in the stream. This is constructed from a table
of known values combined with a small multiplier and divisor - if
the supplied value is not exactly representable, the nearest
representable value will be used instead (see H.262 section 6.3.3
and table 6-4).
@item video_format
Set the video format in the stream (see H.262 section 6.3.6 and
table 6-6).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.262 section 6.3.6
and tables 6-7, 6-8 and 6-9).
@end table
@section mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
@@ -275,6 +437,14 @@ codec) with metadata headers.
See also the @ref{mov2textsub} filter.
@section trace_headers
Log trace output containing all syntax elements in the coded stream
headers (everything above the level of individual coded blocks).
This can be useful for debugging low-level stream issues.
Supports H.264, H.265 and MPEG-2.
@section vp9_superframe
Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
doc/codecs.texi
-25
View File
@@ -44,12 +44,6 @@ Use 1/4 pel motion compensation.
Use loop filter.
@item qscale
Use fixed qscale.
@item gmc
Use gmc.
@item mv0
Always try a mb with mv=<0,0>.
@item input_preserved
@item pass1
Use internal 2pass ratecontrol in first pass mode.
@item pass2
@@ -62,8 +56,6 @@ Do not draw edges.
Set error[?] variables during encoding.
@item truncated
@item naq
Normalize adaptive quantization.
@item ildct
Use interlaced DCT.
@item low_delay
@@ -506,21 +498,6 @@ threading operations
skip motion compensation
@end table
@item vismv @var{integer} (@emph{decoding,video})
Visualize motion vectors (MVs).
This option is deprecated, see the codecview filter instead.
Possible values:
@table @samp
@item pf
forward predicted MVs of P-frames
@item bf
forward predicted MVs of B-frames
@item bb
backward predicted MVs of B-frames
@end table
@item cmp @var{integer} (@emph{encoding,video})
Set full pel me compare function.
@@ -757,8 +734,6 @@ Set context model.
@item slice_flags @var{integer}
@item xvmc_acceleration @var{integer}
@item mbd @var{integer} (@emph{encoding,video})
Set macroblock decision algorithm (high quality mode).
doc/decoders.texi
-7
View File
@@ -25,13 +25,6 @@ enabled decoders.
A description of some of the currently available video decoders
follows.
@section hevc
HEVC / H.265 decoder.
Note: the @option{skip_loop_filter} option has effect only at level
@code{all}.
@section rawvideo
Raw video decoder.
doc/demuxers.texi
+8
View File
@@ -316,6 +316,14 @@ segment index to start live streams at (negative values are from the end).
@item max_reload
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@item http_persistent
Use persistent HTTP connections. Applicable only for HTTP streams.
Enabled by default.
@item http_multiple
Use multiple HTTP connections for downloading HTTP segments.
Enabled by default for HTTP/1.1 servers.
@end table
@section image2
doc/developer.texi
+47 -32
View File
@@ -10,9 +10,7 @@
@contents
@chapter Developers Guide
@section Notes for external developers
@chapter Notes for external developers
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
@@ -30,7 +28,7 @@ For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
@section Contributing
@chapter Contributing
There are 3 ways by which code gets into FFmpeg.
@itemize @bullet
@@ -47,9 +45,9 @@ The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@section Coding Rules
@chapter Coding Rules
@subsection Code formatting conventions
@section Code formatting conventions
There are the following guidelines regarding the indentation in files:
@@ -74,7 +72,7 @@ The presentation is one inspired by 'indent -i4 -kr -nut'.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@subsection Comments
@section Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
@@ -114,7 +112,7 @@ int myfunc(int my_parameter)
...
@end example
@subsection C language features
@section C language features
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@@ -160,7 +158,7 @@ mixing statements and declarations;
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@subsection Naming conventions
@section Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
@@ -184,7 +182,7 @@ e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_aac_parse_header}.
@samp{avpriv_report_missing_feature}.
@item
Each library has its own prefix for public symbols, in addition to the
@@ -204,7 +202,7 @@ letter as they are reserved by the C standard. Names starting with @code{_}
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@subsection Miscellaneous conventions
@section Miscellaneous conventions
@itemize @bullet
@item
@@ -216,7 +214,7 @@ Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@end itemize
@subsection Editor configuration
@section Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
@@ -249,9 +247,9 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
(setq c-default-style "ffmpeg")
@end lisp
@section Development Policy
@chapter Development Policy
@subsection Patches/Committing
@section Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
@@ -350,7 +348,7 @@ time-frame (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@subsection Code
@section Code
@subheading API/ABI changes should be discussed before they are made.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
@@ -381,12 +379,29 @@ Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@subsection Documentation/Other
@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
It is important to be subscribed to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Almost any non-trivial patch is to be sent there for review.
Other developers may have comments about your contribution. We expect you see
those comments, and to improve it if requested. (N.B. Experienced committers
have other channels, and may sometimes skip review for trivial fixes.) Also,
discussion here about bug fixes and FFmpeg improvements by other developers may
be helpful information for you. Finally, by being a list subscriber, your
contribution will be posted immediately to the list, without the moderation
hold which messages from non-subscribers experience.
However, it is more important to the project that we receive your patch than
that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
want to subscribe and discuss the patch, then please do send it to the list
anyway.
@subheading Subscribe to the ffmpeg-cvslog mailing list.
It is important to do this as the diffs of all commits are sent there and
reviewed by all the other developers. Bugs and possible improvements or
general questions regarding commits are discussed there. We expect you to
react if problems with your code are uncovered.
Diffs of all commits are sent to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
mailing list. Some developers read this list to review all code base changes
from all sources. Subscribing to this list is not mandatory.
@subheading Keep the documentation up to date.
Update the documentation if you change behavior or add features. If you are
@@ -406,7 +421,7 @@ finding a new maintainer and also don't forget to update the @file{MAINTAINERS}
We think our rules are not too hard. If you have comments, contact us.
@section Code of conduct
@chapter Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
@@ -436,7 +451,7 @@ Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@section Submitting patches
@chapter Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
@@ -485,7 +500,7 @@ Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@section New codecs or formats checklist
@chapter New codecs or formats checklist
@enumerate
@item
@@ -537,7 +552,7 @@ Did you make sure it compiles standalone, i.e. with
@end enumerate
@section patch submission checklist
@chapter Patch submission checklist
@enumerate
@item
@@ -650,7 +665,7 @@ Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
@end enumerate
@section Patch review process
@chapter Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
@@ -681,7 +696,7 @@ to be reviewed, please consider helping to review other patches, that is a great
way to get everyone's patches reviewed sooner.
@anchor{Regression tests}
@section Regression tests
@chapter Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
@@ -692,7 +707,7 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
this case, the reference results of the regression tests shall be modified
accordingly].
@subsection Adding files to the fate-suite dataset
@section Adding files to the fate-suite dataset
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
@@ -703,7 +718,7 @@ Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@subsection Visualizing Test Coverage
@section Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
@@ -730,7 +745,7 @@ You can use the command @code{make lcov-reset} to reset the coverage
measurements. You will need to rerun @code{make lcov} after running a
new test.
@subsection Using Valgrind
@section Using Valgrind
The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
@@ -744,7 +759,7 @@ In case you need finer control over how valgrind is invoked, use the
your configure line instead.
@anchor{Release process}
@section Release process
@chapter Release process
FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
@@ -776,7 +791,7 @@ adjustments to the symbol versioning file. Please discuss such changes
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
@anchor{Criteria for Point Releases}
@subsection Criteria for Point Releases
@section Criteria for Point Releases
Changes that match the following criteria are valid candidates for
inclusion into a point release:
@@ -800,7 +815,7 @@ point releases of the same release branch.
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@subsection Release Checklist
@section Release Checklist
The release process involves the following steps:
doc/doxy/.gitignore
+1
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@@ -0,0 +1 @@
/html/
doc/encoders.texi
+11 -5
View File
@@ -64,7 +64,6 @@ to find an optimal combination by adding or subtracting a specific value from
all quantizers and adjusting some individual quantizer a little. Will tune
itself based on whether @option{aac_is}, @option{aac_ms} and @option{aac_pns}
are enabled.
This is the default choice for a coder.
@item anmr
Average noise to mask ratio (ANMR) trellis-based solution.
@@ -77,10 +76,10 @@ Not currently recommended.
@item fast
Constant quantizer method.
This method sets a constant quantizer for all bands. This is the fastest of all
the methods and has no rate control or support for @option{aac_is} or
@option{aac_pns}.
Not recommended.
Uses a cheaper version of twoloop algorithm that doesn't try to do as many
clever adjustments. Worse with low bitrates (less than 64kbps), but is better
and much faster at higher bitrates.
This is the default choice for a coder
@end table
@@ -1681,6 +1680,13 @@ colorspaces:
@end table
@item row-mt @var{boolean}
Enable row based multi-threading.
@item tune-content
Set content type: default (0), screen (1), film (2).
@item corpus-complexity
Corpus VBR mode is a variant of standard VBR where the complexity distribution
midpoint is passed in rather than calculated for a specific clip or chunk.
The valid range is [0, 10000]. 0 (default) uses standard VBR.
@end table
@end table
doc/examples/.gitignore
+22
View File
@@ -0,0 +1,22 @@
/avio_dir_cmd
/avio_reading
/decode_audio
/decode_video
/demuxing_decoding
/encode_audio
/encode_video
/extract_mvs
/filter_audio
/filtering_audio
/filtering_video
/http_multiclient
/hw_decode
/metadata
/muxing
/pc-uninstalled
/qsvdec
/remuxing
/resampling_audio
/scaling_video
/transcode_aac
/transcoding
doc/examples/Makefile
+1
View File
@@ -19,6 +19,7 @@ EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
EXAMPLES := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
EXAMPLES_G := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
doc/examples/avio_reading.c
+2
View File
@@ -44,6 +44,8 @@ static int read_packet(void *opaque, uint8_t *buf, int buf_size)
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
if (!buf_size)
return AVERROR_EOF;
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
doc/examples/filter_audio.c
+4 -4
View File
@@ -64,13 +64,13 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
AVFilter *abuffer;
const AVFilter *abuffer;
AVFilterContext *volume_ctx;
AVFilter *volume;
const AVFilter *volume;
AVFilterContext *aformat_ctx;
AVFilter *aformat;
const AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
AVFilter *abuffersink;
const AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
doc/examples/filtering_audio.c
+2 -3
View File
@@ -32,7 +32,6 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -90,8 +89,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
doc/examples/filtering_video.c
+2 -3
View File
@@ -32,7 +32,6 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -93,8 +92,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
doc/examples/hw_decode.c
+22 -32
View File
@@ -44,34 +44,6 @@ static AVBufferRef *hw_device_ctx = NULL;
static enum AVPixelFormat hw_pix_fmt;
static FILE *output_file = NULL;
static enum AVPixelFormat find_fmt_by_hw_type(const enum AVHWDeviceType type)
{
enum AVPixelFormat fmt;
switch (type) {
case AV_HWDEVICE_TYPE_VAAPI:
fmt = AV_PIX_FMT_VAAPI;
break;
case AV_HWDEVICE_TYPE_DXVA2:
fmt = AV_PIX_FMT_DXVA2_VLD;
break;
case AV_HWDEVICE_TYPE_D3D11VA:
fmt = AV_PIX_FMT_D3D11;
break;
case AV_HWDEVICE_TYPE_VDPAU:
fmt = AV_PIX_FMT_VDPAU;
break;
case AV_HWDEVICE_TYPE_VIDEOTOOLBOX:
fmt = AV_PIX_FMT_VIDEOTOOLBOX;
break;
default:
fmt = AV_PIX_FMT_NONE;
break;
}
return fmt;
}
static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
{
int err = 0;
@@ -184,18 +156,22 @@ int main(int argc, char *argv[])
AVCodec *decoder = NULL;
AVPacket packet;
enum AVHWDeviceType type;
int i;
if (argc < 4) {
fprintf(stderr, "Usage: %s <vaapi|vdpau|dxva2|d3d11va> <input file> <output file>\n", argv[0]);
fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
return -1;
}
av_register_all();
type = av_hwdevice_find_type_by_name(argv[1]);
hw_pix_fmt = find_fmt_by_hw_type(type);
if (hw_pix_fmt == -1) {
fprintf(stderr, "Cannot support '%s' in this example.\n", argv[1]);
if (type == AV_HWDEVICE_TYPE_NONE) {
fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
fprintf(stderr, "Available device types:");
while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
fprintf(stderr, "\n");
return -1;
}
@@ -218,6 +194,20 @@ int main(int argc, char *argv[])
}
video_stream = ret;
for (i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
if (!config) {
fprintf(stderr, "Decoder %s does not support device type %s.\n",
decoder->name, av_hwdevice_get_type_name(type));
return -1;
}
if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
config->device_type == type) {
hw_pix_fmt = config->pix_fmt;
break;
}
}
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
doc/examples/muxing.c
+3 -3
View File
@@ -488,9 +488,9 @@ static AVFrame *get_video_frame(OutputStream *ost)
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx,
(const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
0, c->height, ost->frame->data, ost->frame->linesize);
sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
doc/examples/transcode_aac.c
+208 -151
View File
@@ -1,4 +1,6 @@
/*
* Copyright (c) 2013-2017 Andreas Unterweger
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -18,10 +20,11 @@
/**
* @file
* simple audio converter
* Simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
@@ -40,12 +43,18 @@
#include "libswresample/swresample.h"
/** The output bit rate in kbit/s */
/* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
/** The number of output channels */
/* The number of output channels */
#define OUTPUT_CHANNELS 2
/** Open an input file and the required decoder. */
/**
* Open an input file and the required decoder.
* @param filename File to be opened
* @param[out] input_format_context Format context of opened file
* @param[out] input_codec_context Codec context of opened file
* @return Error code (0 if successful)
*/
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
@@ -54,7 +63,7 @@ static int open_input_file(const char *filename,
AVCodec *input_codec;
int error;
/** Open the input file to read from it. */
/* Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
@@ -63,7 +72,7 @@ static int open_input_file(const char *filename,
return error;
}
/** Get information on the input file (number of streams etc.). */
/* Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
fprintf(stderr, "Could not open find stream info (error '%s')\n",
av_err2str(error));
@@ -71,7 +80,7 @@ static int open_input_file(const char *filename,
return error;
}
/** Make sure that there is only one stream in the input file. */
/* Make sure that there is only one stream in the input file. */
if ((*input_format_context)->nb_streams != 1) {
fprintf(stderr, "Expected one audio input stream, but found %d\n",
(*input_format_context)->nb_streams);
@@ -79,14 +88,14 @@ static int open_input_file(const char *filename,
return AVERROR_EXIT;
}
/** Find a decoder for the audio stream. */
/* Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** allocate a new decoding context */
/* Allocate a new decoding context. */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate a decoding context\n");
@@ -94,7 +103,7 @@ static int open_input_file(const char *filename,
return AVERROR(ENOMEM);
}
/** initialize the stream parameters with demuxer information */
/* Initialize the stream parameters with demuxer information. */
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
@@ -102,7 +111,7 @@ static int open_input_file(const char *filename,
return error;
}
/** Open the decoder for the audio stream to use it later. */
/* Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
av_err2str(error));
@@ -111,7 +120,7 @@ static int open_input_file(const char *filename,
return error;
}
/** Save the decoder context for easier access later. */
/* Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
@@ -121,6 +130,11 @@ static int open_input_file(const char *filename,
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
* @param filename File to be opened
* @param input_codec_context Codec context of input file
* @param[out] output_format_context Format context of output file
* @param[out] output_codec_context Codec context of output file
* @return Error code (0 if successful)
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
@@ -133,7 +147,7 @@ static int open_output_file(const char *filename,
AVCodec *output_codec = NULL;
int error;
/** Open the output file to write to it. */
/* Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
@@ -141,16 +155,16 @@ static int open_output_file(const char *filename,
return error;
}
/** Create a new format context for the output container format. */
/* Create a new format context for the output container format. */
if (!(*output_format_context = avformat_alloc_context())) {
fprintf(stderr, "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
/** Associate the output file (pointer) with the container format context. */
/* Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
/** Guess the desired container format based on the file extension. */
/* Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
@@ -160,13 +174,13 @@ static int open_output_file(const char *filename,
av_strlcpy((*output_format_context)->filename, filename,
sizeof((*output_format_context)->filename));
/** Find the encoder to be used by its name. */
/* Find the encoder to be used by its name. */
if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
fprintf(stderr, "Could not find an AAC encoder.\n");
goto cleanup;
}
/** Create a new audio stream in the output file container. */
/* Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
@@ -180,31 +194,27 @@ static int open_output_file(const char *filename,
goto cleanup;
}
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
/* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion. */
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
/** Allow the use of the experimental AAC encoder */
/* Allow the use of the experimental AAC encoder. */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/** Set the sample rate for the container. */
/* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
/**
* Some container formats (like MP4) require global headers to be present
* Mark the encoder so that it behaves accordingly.
*/
/* Some container formats (like MP4) require global headers to be present.
* Mark the encoder so that it behaves accordingly. */
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/** Open the encoder for the audio stream to use it later. */
/* Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
av_err2str(error));
@@ -217,7 +227,7 @@ static int open_output_file(const char *filename,
goto cleanup;
}
/** Save the encoder context for easier access later. */
/* Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
@@ -230,16 +240,23 @@ cleanup:
return error < 0 ? error : AVERROR_EXIT;
}
/** Initialize one data packet for reading or writing. */
/**
* Initialize one data packet for reading or writing.
* @param packet Packet to be initialized
*/
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
/** Set the packet data and size so that it is recognized as being empty. */
/* Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/** Initialize one audio frame for reading from the input file */
/**
* Initialize one audio frame for reading from the input file.
* @param[out] frame Frame to be initialized
* @return Error code (0 if successful)
*/
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
@@ -253,6 +270,10 @@ static int init_input_frame(AVFrame **frame)
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param[out] resample_context Resample context for the required conversion
* @return Error code (0 if successful)
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
@@ -260,7 +281,7 @@ static int init_resampler(AVCodecContext *input_codec_context,
{
int error;
/**
/*
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
@@ -279,14 +300,14 @@ static int init_resampler(AVCodecContext *input_codec_context,
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/**
/*
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/** Open the resampler with the specified parameters. */
/* Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
@@ -295,10 +316,15 @@ static int init_resampler(AVCodecContext *input_codec_context,
return 0;
}
/** Initialize a FIFO buffer for the audio samples to be encoded. */
/**
* Initialize a FIFO buffer for the audio samples to be encoded.
* @param[out] fifo Sample buffer
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
/** Create the FIFO buffer based on the specified output sample format. */
/* Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
@@ -307,7 +333,11 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
return 0;
}
/** Write the header of the output file container. */
/**
* Write the header of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
@@ -319,20 +349,32 @@ static int write_output_file_header(AVFormatContext *output_format_context)
return 0;
}
/** Decode one audio frame from the input file. */
/**
* Decode one audio frame from the input file.
* @param frame Audio frame to be decoded
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param[out] data_present Indicates whether data has been decoded
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false, there
* is more data to be decoded, i.e., this
* function has to be called again.
* @return Error code (0 if successful)
*/
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/** Packet used for temporary storage. */
/* Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
/** Read one audio frame from the input file into a temporary packet. */
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
/* If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
@@ -342,12 +384,10 @@ static int decode_audio_frame(AVFrame *frame,
}
}
/**
* Decode the audio frame stored in the temporary packet.
/* Decode the audio frame stored in the temporary packet.
* The input audio stream decoder is used to do this.
* If we are at the end of the file, pass an empty packet to the decoder
* to flush it.
*/
* to flush it. */
if ((error = avcodec_decode_audio4(input_codec_context, frame,
data_present, &input_packet)) < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
@@ -356,10 +396,8 @@ static int decode_audio_frame(AVFrame *frame,
return error;
}
/**
* If the decoder has not been flushed completely, we are not finished,
* so that this function has to be called again.
*/
/* If the decoder has not been flushed completely, we are not finished,
* so that this function has to be called again. */
if (*finished && *data_present)
*finished = 0;
av_packet_unref(&input_packet);
@@ -370,6 +408,13 @@ static int decode_audio_frame(AVFrame *frame,
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
* @param[out] converted_input_samples Array of converted samples. The
* dimensions are reference, channel
* (for multi-channel audio), sample.
* @param output_codec_context Codec context of the output file
* @param frame_size Number of samples to be converted in
* each round
* @return Error code (0 if successful)
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
@@ -377,8 +422,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
{
int error;
/**
* Allocate as many pointers as there are audio channels.
/* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
@@ -388,10 +432,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
return AVERROR(ENOMEM);
}
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
@@ -408,8 +450,15 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is specified
* by frame_size.
* The conversion happens on a per-frame basis, the size of which is
* specified by frame_size.
* @param input_data Samples to be decoded. The dimensions are
* channel (for multi-channel audio), sample.
* @param[out] converted_data Converted samples. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @param resample_context Resample context for the conversion
* @return Error code (0 if successful)
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
@@ -417,7 +466,7 @@ static int convert_samples(const uint8_t **input_data,
{
int error;
/** Convert the samples using the resampler. */
/* Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
@@ -429,23 +478,28 @@ static int convert_samples(const uint8_t **input_data,
return 0;
}
/** Add converted input audio samples to the FIFO buffer for later processing. */
/**
* Add converted input audio samples to the FIFO buffer for later processing.
* @param fifo Buffer to add the samples to
* @param converted_input_samples Samples to be added. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @return Error code (0 if successful)
*/
static int add_samples_to_fifo(AVAudioFifo *fifo,
uint8_t **converted_input_samples,
const int frame_size)
{
int error;
/**
* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples.
*/
/* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples. */
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
/** Store the new samples in the FIFO buffer. */
/* Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
@@ -455,8 +509,20 @@ static int add_samples_to_fifo(AVAudioFifo *fifo,
}
/**
* Read one audio frame from the input file, decodes, converts and stores
* Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer.
* @param fifo Buffer used for temporary storage
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param resampler_context Resample context for the conversion
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false,
* there is more data to be decoded,
* i.e., this function has to be called
* again.
* @return Error code (0 if successful)
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
@@ -465,45 +531,41 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
SwrContext *resampler_context,
int *finished)
{
/** Temporary storage of the input samples of the frame read from the file. */
/* Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/** Temporary storage for the converted input samples. */
/* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int ret = AVERROR_EXIT;
/** Initialize temporary storage for one input frame. */
/* Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/** Decode one frame worth of audio samples. */
/* Decode one frame worth of audio samples. */
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/**
* If we are at the end of the file and there are no more samples
/* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error.
*/
* This must not be treated as an error. */
if (*finished && !data_present) {
ret = 0;
goto cleanup;
}
/** If there is decoded data, convert and store it */
/* If there is decoded data, convert and store it. */
if (data_present) {
/** Initialize the temporary storage for the converted input samples. */
/* Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
/**
* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples.
*/
/* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples. */
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
/** Add the converted input samples to the FIFO buffer for later processing. */
/* Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
@@ -524,6 +586,10 @@ cleanup:
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
* @param[out] frame Frame to be initialized
* @param output_codec_context Codec context of the output file
* @param frame_size Size of the frame
* @return Error code (0 if successful)
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
@@ -531,28 +597,24 @@ static int init_output_frame(AVFrame **frame,
{
int error;
/** Create a new frame to store the audio samples. */
/* Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/**
* Set the frame's parameters, especially its size and format.
/* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity.
*/
* are assumed for simplicity. */
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/**
* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified.
*/
/* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified. */
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
av_err2str(error));
@@ -563,30 +625,36 @@ static int init_output_frame(AVFrame **frame,
return 0;
}
/** Global timestamp for the audio frames */
/* Global timestamp for the audio frames. */
static int64_t pts = 0;
/** Encode one frame worth of audio to the output file. */
/**
* Encode one frame worth of audio to the output file.
* @param frame Samples to be encoded
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @param[out] data_present Indicates whether data has been
* decoded
* @return Error code (0 if successful)
*/
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/** Packet used for temporary storage. */
/* Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
/** Set a timestamp based on the sample rate for the container. */
/* Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
/* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this. */
if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
frame, data_present)) < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
@@ -595,7 +663,7 @@ static int encode_audio_frame(AVFrame *frame,
return error;
}
/** Write one audio frame from the temporary packet to the output file. */
/* Write one audio frame from the temporary packet to the output file. */
if (*data_present) {
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
@@ -613,37 +681,37 @@ static int encode_audio_frame(AVFrame *frame,
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
* @param fifo Buffer used for temporary storage
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
/** Temporary storage of the output samples of the frame written to the file. */
/* Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/**
* Use the maximum number of possible samples per frame.
/* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size
*/
* buffer use this number. Otherwise, use the maximum possible frame size. */
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
/** Initialize temporary storage for one output frame. */
/* Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
/**
* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily.
*/
/* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily. */
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
fprintf(stderr, "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
/** Encode one frame worth of audio samples. */
/* Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
@@ -653,7 +721,11 @@ static int load_encode_and_write(AVAudioFifo *fifo,
return 0;
}
/** Write the trailer of the output file container. */
/**
* Write the trailer of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
@@ -665,7 +737,6 @@ static int write_output_file_trailer(AVFormatContext *output_format_context)
return 0;
}
/** Convert an audio file to an AAC file in an MP4 container. */
int main(int argc, char **argv)
{
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
@@ -674,89 +745,75 @@ int main(int argc, char **argv)
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc < 3) {
if (argc != 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/** Register all codecs and formats so that they can be used. */
/* Register all codecs and formats so that they can be used. */
av_register_all();
/** Open the input file for reading. */
/* Open the input file for reading. */
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/** Open the output file for writing. */
/* Open the output file for writing. */
if (open_output_file(argv[2], input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
/** Initialize the resampler to be able to convert audio sample formats. */
/* Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/** Initialize the FIFO buffer to store audio samples to be encoded. */
/* Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
/** Write the header of the output file container. */
/* Write the header of the output file container. */
if (write_output_file_header(output_format_context))
goto cleanup;
/**
* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither.
*/
/* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither. */
while (1) {
/** Use the encoder's desired frame size for processing. */
/* Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/**
* Make sure that there is one frame worth of samples in the FIFO
/* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples.
*/
* that they make up at least one frame worth of output samples. */
while (av_audio_fifo_size(fifo) < output_frame_size) {
/**
* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer.
*/
/* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer. */
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/**
* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file.
*/
/* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file. */
if (finished)
break;
}
/**
* If we have enough samples for the encoder, we encode them.
/* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder.
*/
* the encoder. */
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/**
* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish.
*/
/* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish. */
if (finished) {
int data_written;
/** Flush the encoder as it may have delayed frames. */
/* Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
@@ -766,7 +823,7 @@ int main(int argc, char **argv)
}
}
/** Write the trailer of the output file container. */
/* Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
doc/examples/transcoding.c
+2 -3
View File
@@ -30,7 +30,6 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -228,8 +227,8 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
{
char args[512];
int ret = 0;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
const AVFilter *buffersrc = NULL;
const AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
doc/examples/vaapi_encode.c
+224
View File
@@ -0,0 +1,224 @@
/*
* Video Acceleration API (video encoding) encode sample
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Intel VAAPI-accelerated encoding example.
*
* @example vaapi_encode.c
* This example shows how to do VAAPI-accelerated encoding. now only support NV12
* raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
*
*/
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <libavcodec/avcodec.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
static int width, height;
static AVBufferRef *hw_device_ctx = NULL;
static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
{
AVBufferRef *hw_frames_ref;
AVHWFramesContext *frames_ctx = NULL;
int err = 0;
if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
fprintf(stderr, "Failed to create VAAPI frame context.\n");
return -1;
}
frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
frames_ctx->format = AV_PIX_FMT_VAAPI;
frames_ctx->sw_format = AV_PIX_FMT_NV12;
frames_ctx->width = width;
frames_ctx->height = height;
frames_ctx->initial_pool_size = 20;
if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
fprintf(stderr, "Failed to initialize VAAPI frame context."
"Error code: %s\n",av_err2str(err));
av_buffer_unref(&hw_frames_ref);
return err;
}
ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
if (!ctx->hw_frames_ctx)
err = AVERROR(ENOMEM);
av_buffer_unref(&hw_frames_ref);
return err;
}
static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
{
int ret = 0;
AVPacket enc_pkt;
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
fprintf(stderr, "Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(avctx, &enc_pkt);
if (ret)
break;
enc_pkt.stream_index = 0;
ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
av_packet_unref(&enc_pkt);
}
end:
ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
return ret;
}
int main(int argc, char *argv[])
{
int size, err;
FILE *fin = NULL, *fout = NULL;
AVFrame *sw_frame = NULL, *hw_frame = NULL;
AVCodecContext *avctx = NULL;
AVCodec *codec = NULL;
const char *enc_name = "h264_vaapi";
if (argc < 5) {
fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
return -1;
}
width = atoi(argv[1]);
height = atoi(argv[2]);
size = width * height;
if (!(fin = fopen(argv[3], "r"))) {
fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
return -1;
}
if (!(fout = fopen(argv[4], "w+b"))) {
fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
err = -1;
goto close;
}
avcodec_register_all();
err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
NULL, NULL, 0);
if (err < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
goto close;
}
if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
fprintf(stderr, "Could not find encoder.\n");
err = -1;
goto close;
}
if (!(avctx = avcodec_alloc_context3(codec))) {
err = AVERROR(ENOMEM);
goto close;
}
avctx->width = width;
avctx->height = height;
avctx->time_base = (AVRational){1, 25};
avctx->framerate = (AVRational){25, 1};
avctx->sample_aspect_ratio = (AVRational){1, 1};
avctx->pix_fmt = AV_PIX_FMT_VAAPI;
/* set hw_frames_ctx for encoder's AVCodecContext */
if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
fprintf(stderr, "Failed to set hwframe context.\n");
goto close;
}
if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
goto close;
}
while (1) {
if (!(sw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
/* read data into software frame, and transfer them into hw frame */
sw_frame->width = width;
sw_frame->height = height;
sw_frame->format = AV_PIX_FMT_NV12;
if ((err = av_frame_get_buffer(sw_frame, 32)) < 0)
goto close;
if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
break;
if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
break;
if (!(hw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
fprintf(stderr, "Error code: %s.\n", av_err2str(err));
goto close;
}
if (!hw_frame->hw_frames_ctx) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
fprintf(stderr, "Error while transferring frame data to surface."
"Error code: %s.\n", av_err2str(err));
goto close;
}
if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
fprintf(stderr, "Failed to encode.\n");
goto close;
}
av_frame_free(&hw_frame);
av_frame_free(&sw_frame);
}
/* flush encoder */
err = encode_write(avctx, NULL, fout);
if (err == AVERROR_EOF)
err = 0;
close:
if (fin)
fclose(fin);
if (fout)
fclose(fout);
av_frame_free(&sw_frame);
av_frame_free(&hw_frame);
avcodec_free_context(&avctx);
av_buffer_unref(&hw_device_ctx);
return err;
}
doc/faq.texi
+65
View File
@@ -501,6 +501,71 @@ ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
@end example
@anchor{background task}
@section How do I run ffmpeg as a background task?
ffmpeg normally checks the console input, for entries like "q" to stop
and "?" to give help, while performing operations. ffmpeg does not have a way of
detecting when it is running as a background task.
When it checks the console input, that can cause the process running ffmpeg
in the background to suspend.
To prevent those input checks, allowing ffmpeg to run as a background task,
use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
or invoke ffmpeg in its own process via an operating system API.
As an alternative, when you are running ffmpeg in a shell, you can redirect
standard input to @code{/dev/null} (on Linux and Mac OS)
or @code{NUL} (on Windows). You can do this redirect either
on the ffmpeg invocation, or from a shell script which calls ffmpeg.
For example:
@example
ffmpeg -nostdin -i INPUT OUTPUT
@end example
or (on Linux, Mac OS, and other UNIX-like shells):
@example
ffmpeg -i INPUT OUTPUT </dev/null
@end example
or (on Windows):
@example
ffmpeg -i INPUT OUTPUT <NUL
@end example
@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
If you run ffmpeg in the background, you may find that its process suspends.
There may be a message like @emph{suspended (tty output)}. The question is how
to prevent the process from being suspended.
For example:
@example
% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
[1] 93352
%
[1] + suspended (tty output) ffmpeg -i INPUT OUTPUT &>
@end example
The message "tty output" notwithstanding, the problem here is that
ffmpeg normally checks the console input when it runs. The operating system
detects this, and suspends the process until you can bring it to the
foreground and attend to it.
The solution is to use the right techniques to tell ffmpeg not to consult
console input. You can use the
@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
or redirect standard input with @code{< /dev/null}.
See FAQ
@ref{background task, @emph{How do I run ffmpeg as a background task?}}
for details.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
doc/fate.texi
+20
View File
@@ -147,6 +147,26 @@ process.
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@chapter Uploading new samples to the fate suite
This is for developers who have an account on the fate suite server.
If you upload new samples, please make sure they are as small as possible,
space on each client, network bandwidth and so on benefit from smaller test cases.
Also keep in mind older checkouts use existing sample files, that means in
practice generally do not replace, remove or overwrite files as it likely would
break older checkouts or releases.
@example
#First update your local samples copy:
rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
#Then do a dry run checking what would be uploaded:
rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
#Upload the files:
rsync -vaL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
@end example
@chapter FATE makefile targets and variables
doc/ffmpeg-bitstream-filters.texi
+2 -2
View File
@@ -26,12 +26,12 @@ bitstream level modifications without performing decoding.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-codecs.texi
+2 -2
View File
@@ -23,12 +23,12 @@ the libavcodec library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-devices.texi
+2 -2
View File
@@ -23,12 +23,12 @@ libavdevice library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-filters.texi
+2 -2
View File
@@ -23,12 +23,12 @@ libavfilter library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavfilter.html,libavfilter}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
ffmpeg(1), ffplay(1), ffprobe(1), libavfilter(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-formats.texi
+2 -2
View File
@@ -23,12 +23,12 @@ provided by the libavformat library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-protocols.texi
+2 -2
View File
@@ -23,12 +23,12 @@ libavformat library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-resampler.texi
+2 -2
View File
@@ -25,12 +25,12 @@ and convert audio format and packing layout.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libswresample.html,libswresample}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-scaler.texi
+2 -2
View File
@@ -24,12 +24,12 @@ image rescaling and pixel format conversion.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libswscale.html,libswscale}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
ffmpeg(1), ffplay(1), ffprobe(1), libswscale(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg-utils.texi
+2 -2
View File
@@ -23,12 +23,12 @@ by the libavutil library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
ffmpeg(1), ffplay(1), ffprobe(1), libavutil(3)
@end ifnothtml
@include authors.texi
doc/ffmpeg.texi
+44 -21
View File
@@ -289,8 +289,8 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
-to and -t are mutually exclusive and -t has priority.
@item -to @var{position} (@emph{output})
Stop writing the output at @var{position}.
@item -to @var{position} (@emph{input/output})
Stop writing the output or reading the input at @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@@ -470,6 +470,7 @@ the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
consists of only alphanumeric characters. The last key of a sequence of
progress information is always "progress".
@anchor{stdin option}
@item -stdin
Enable interaction on standard input. On by default unless standard input is
used as an input. To explicitly disable interaction you need to specify
@@ -756,6 +757,43 @@ If not specified, @samp{auto_any} is used.
platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
QSV device from that.)
@item opencl
@var{device} selects the platform and device as @emph{platform_index.device_index}.
The set of devices can also be filtered using the key-value pairs to find only
devices matching particular platform or device strings.
The strings usable as filters are:
@table @option
@item platform_profile
@item platform_version
@item platform_name
@item platform_vendor
@item platform_extensions
@item device_name
@item device_vendor
@item driver_version
@item device_version
@item device_profile
@item device_extensions
@item device_type
@end table
The indices and filters must together uniquely select a device.
Examples:
@table @emph
@item -init_hw_device opencl:0.1
Choose the second device on the first platform.
@item -init_hw_device opencl:,device_name=Foo9000
Choose the device with a name containing the string @emph{Foo9000}.
@item -init_hw_device opencl:1,device_type=gpu,device_extensions=cl_khr_fp16
Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
extension.
@end table
@end table
@item -init_hw_device @var{type}[=@var{name}]@@@var{source}
@@ -786,9 +824,6 @@ Do not use any hardware acceleration (the default).
@item auto
Automatically select the hardware acceleration method.
@item vda
Use Apple VDA hardware acceleration.
@item vdpau
Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
@@ -1132,10 +1167,6 @@ loss).
By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
This option will slow down the reading of the input(s) to the native frame rate
of the input(s). It is useful for real-time output (e.g. live streaming).
@item -loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
This option is deprecated, use -loop 1.
@item -loop_output @var{number_of_times}
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
@@ -1253,6 +1284,8 @@ or as a floating point number (e.g. 0.04166, 2.0833e-5)
Default value is 0.
@item -bitexact (@emph{input/output})
Enable bitexact mode for (de)muxer and (de/en)coder
@item -shortest (@emph{output})
Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
@@ -1375,16 +1408,6 @@ file or device. With low latency / high rate live streams, packets may be
discarded if they are not read in a timely manner; raising this value can
avoid it.
@item -override_ffserver (@emph{global})
Overrides the input specifications from @command{ffserver}. Using this
option you can map any input stream to @command{ffserver} and control
many aspects of the encoding from @command{ffmpeg}. Without this
option @command{ffmpeg} will transmit to @command{ffserver} what is
requested by @command{ffserver}.
The option is intended for cases where features are needed that cannot be
specified to @command{ffserver} but can be to @command{ffmpeg}.
@item -sdp_file @var{file} (@emph{global})
Print sdp information for an output stream to @var{file}.
This allows dumping sdp information when at least one output isn't an
@@ -1739,7 +1762,7 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@ifset config-not-all
@url{ffmpeg-all.html,ffmpeg-all},
@end ifset
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -1758,7 +1781,7 @@ ffmpeg(1),
@ifset config-not-all
ffmpeg-all(1),
@end ifset
ffplay(1), ffprobe(1), ffserver(1),
ffplay(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
doc/ffplay.texi
+2 -2
View File
@@ -291,7 +291,7 @@ Toggle full screen.
@ifset config-not-all
@url{ffplay-all.html,ffmpeg-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -310,7 +310,7 @@ ffplay(1),
@ifset config-not-all
ffplay-all(1),
@end ifset
ffmpeg(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
doc/ffprobe.texi
+2 -2
View File
@@ -653,7 +653,7 @@ DV, GXF and AVI timecodes are available in format metadata
@ifset config-not-all
@url{ffprobe-all.html,ffprobe-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -672,7 +672,7 @@ ffprobe(1),
@ifset config-not-all
ffprobe-all(1),
@end ifset
ffmpeg(1), ffplay(1), ffserver(1),
ffmpeg(1), ffplay(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
doc/ffserver.conf
-372
View File
@@ -1,372 +0,0 @@
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
HTTPPort 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
HTTPBindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000
# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -
##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.
<Feed feed1.ffm>
# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
#
# ffmpeg http://localhost:8090/feed1.ffm
# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200K
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
ACL allow 127.0.0.1
</Feed>
##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.
<Stream test1.mpg>
# coming from live feed 'feed1'
Feed feed1.ffm
# Format of the stream : you can choose among:
# mpeg : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# mjpeg : Generate a M-JPEG stream.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
Format mpeg
# Bitrate for the audio stream. Codecs usually support only a few
# different bitrates.
AudioBitRate 32
# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 1
# Sampling frequency for audio. When using low bitrates, you should
# lower this frequency to 22050 or 11025. The supported frequencies
# depend on the selected audio codec.
AudioSampleRate 44100
# Bitrate for the video stream
VideoBitRate 64
# Ratecontrol buffer size
VideoBufferSize 40
# Number of frames per second
VideoFrameRate 3
# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize 160x128
# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly
# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12
# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector
# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video
# Suppress audio
#NoAudio
# Suppress video
#NoVideo
#VideoQMin 3
#VideoQMax 31
# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15
# ACL:
# You can allow ranges of addresses (or single addresses)
#ACL ALLOW <first address> <last address>
# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address> <last address>
# You can repeat the ACL allow/deny as often as you like. It is on a per
# stream basis. The first match defines the action. If there are no matches,
# then the default is the inverse of the last ACL statement.
#
# Thus 'ACL allow localhost' only allows access from localhost.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.
</Stream>
##################################################################
# Example streams
# Multipart JPEG
#<Stream test.mjpg>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</Stream>
# Single JPEG
#<Stream test.jpg>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</Stream>
# Flash
#<Stream test.swf>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</Stream>
# ASF compatible
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
# MP3 audio
#<Stream test.mp3>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Ogg Vorbis audio
#<Stream test.ogg>
#Feed feed1.ffm
#Metadata title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Real with audio only at 32 kbits
#<Stream test.ra>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</Stream>
# Real with audio and video at 64 kbits
#<Stream test.rm>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</Stream>
##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF
#<Stream file.rm>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</Stream>
#<Stream file.asf>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Metadata author "Me"
#Metadata copyright "Super MegaCorp"
#Metadata title "Test stream from disk"
#Metadata comment "Test comment"
#</Stream>
##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp
#<Stream test1-rtsp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</Stream>
# Transcode an incoming live feed to another live feed,
# using libx264 and video presets
#<Stream live.h264>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec aac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</Stream>
##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# 'sdp' extension to the stream name (here
# http://localhost:8090/test1-sdp.sdp). You should usually give this
# file to your player to play the stream.
#
# The 'NoLoop' option can be used to avoid looping when the stream is
# terminated.
#<Stream test1-sdp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</Stream>
##################################################################
# Special streams
# Server status
<Stream stat.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</Stream>
# Redirect index.html to the appropriate site
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>
doc/ffserver.texi
-923
View File
@@ -1,923 +0,0 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffserver Documentation
@titlepage
@center @titlefont{ffserver Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION
@command{ffserver} is a streaming server for both audio and video.
It supports several live feeds, streaming from files and time shifting
on live feeds. You can seek to positions in the past on each live
feed, provided you specify a big enough feed storage.
@command{ffserver} is configured through a configuration file, which
is read at startup. If not explicitly specified, it will read from
@file{/etc/ffserver.conf}.
@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
RTP/RTSP/HTTP.
An @command{ffserver} instance will listen on some port as specified
in the configuration file. You can launch one or more instances of
@command{ffmpeg} and send one or more FFM streams to the port where
ffserver is expecting to receive them. Alternately, you can make
@command{ffserver} launch such @command{ffmpeg} instances at startup.
Input streams are called feeds, and each one is specified by a
@code{<Feed>} section in the configuration file.
For each feed you can have different output streams in various
formats, each one specified by a @code{<Stream>} section in the
configuration file.
@chapter Detailed description
@command{ffserver} works by forwarding streams encoded by
@command{ffmpeg}, or pre-recorded streams which are read from disk.
Precisely, @command{ffserver} acts as an HTTP server, accepting POST
requests from @command{ffmpeg} to acquire the stream to publish, and
serving RTSP clients or HTTP clients GET requests with the stream
media content.
A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
a port where @command{ffserver} is listening.
Each feed is identified by a unique name, corresponding to the name
of the resource published on @command{ffserver}, and is configured by
a dedicated @code{Feed} section in the configuration file.
The feed publish URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
@end example
where @var{ffserver_ip_address} is the IP address of the machine where
@command{ffserver} is installed, @var{http_port} is the port number of
the HTTP server (configured through the @option{HTTPPort} option), and
@var{feed_name} is the name of the corresponding feed defined in the
configuration file.
Each feed is associated to a file which is stored on disk. This stored
file is used to send pre-recorded data to a player as fast as
possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by
@command{ffserver}, and made accessible through the HTTP protocol to
clients.
A stream can be connected to a feed, or to a file. In the first case,
the published stream is forwarded from the corresponding feed
generated by a running instance of @command{ffmpeg}, in the second
case the stream is read from a pre-recorded file.
Each stream is identified by a unique name, corresponding to the name
of the resource served by @command{ffserver}, and is configured by
a dedicated @code{Stream} section in the configuration file.
The stream access HTTP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
@end example
The stream access RTSP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
@end example
@var{stream_name} is the name of the corresponding stream defined in
the configuration file. @var{options} is a list of options specified
after the URL which affects how the stream is served by
@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
and RTSP ports configured with the options @var{HTTPPort} and
@var{RTSPPort} respectively.
In case the stream is associated to a feed, the encoding parameters
must be configured in the stream configuration. They are sent to
@command{ffmpeg} when setting up the encoding. This allows
@command{ffserver} to define the encoding parameters used by
the @command{ffmpeg} encoders.
The @command{ffmpeg} @option{override_ffserver} commandline option
allows one to override the encoding parameters set by the server.
Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following
graph:
@verbatim
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
\ |_________|\ |__________|
\ \
\ \ __________
\ \ | |
\ \| stream 2 |
\ |__________|
\
\ _________ __________
\ | | | |
\| feed 2 |-----| stream 3 |
|_________| |__________|
_________ __________
| | | |
ffmpeg 2 -----| feed 3 |-----| stream 4 |
|_________| |__________|
_________ __________
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
@end verbatim
@anchor{FFM}
@section FFM, FFM2 formats
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
@section Status stream
@command{ffserver} supports an HTTP interface which exposes the
current status of the server.
Simply point your browser to the address of the special status stream
specified in the configuration file.
For example if you have:
@example
<Stream status.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</Stream>
@end example
then the server will post a page with the status information when
the special stream @file{status.html} is requested.
@section How do I make it work?
As a simple test, just run the following two command lines where INPUTFILE
is some file which you can decode with ffmpeg:
@example
ffserver -f doc/ffserver.conf &
ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
@end example
At this point you should be able to go to your Windows machine and fire up
Windows Media Player (WMP). Go to Open URL and enter
@example
http://<linuxbox>:8090/test.asf
@end example
You should (after a short delay) see video and hear audio.
WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
transfer the entire file before starting to play.
The same is true of AVI files.
You should edit the @file{ffserver.conf} file to suit your needs (in
terms of frame rates etc). Then install @command{ffserver} and
@command{ffmpeg}, write a script to start them up, and off you go.
@section What else can it do?
You can replay video from .ffm files that was recorded earlier.
However, there are a number of caveats, including the fact that the
ffserver parameters must match the original parameters used to record the
file. If they do not, then ffserver deletes the file before recording into it.
(Now that I write this, it seems broken).
You can fiddle with many of the codec choices and encoding parameters, and
there are a bunch more parameters that you cannot control. Post a message
to the mailing list if there are some 'must have' parameters. Look in
ffserver.conf for a list of the currently available controls.
It will automatically generate the ASX or RAM files that are often used
in browsers. These files are actually redirections to the underlying ASF
or RM file. The reason for this is that the browser often fetches the
entire file before starting up the external viewer. The redirection files
are very small and can be transferred quickly. [The stream itself is
often 'infinite' and thus the browser tries to download it and never
finishes.]
@section Tips
* When you connect to a live stream, most players (WMP, RA, etc) want to
buffer a certain number of seconds of material so that they can display the
signal continuously. However, ffserver (by default) starts sending data
in realtime. This means that there is a pause of a few seconds while the
buffering is being done by the player. The good news is that this can be
cured by adding a '?buffer=5' to the end of the URL. This means that the
stream should start 5 seconds in the past -- and so the first 5 seconds
of the stream are sent as fast as the network will allow. It will then
slow down to real time. This noticeably improves the startup experience.
You can also add a 'Preroll 15' statement into the ffserver.conf that will
add the 15 second prebuffering on all requests that do not otherwise
specify a time. In addition, ffserver will skip frames until a key_frame
is found. This further reduces the startup delay by not transferring data
that will be discarded.
@section Why does the ?buffer / Preroll stop working after a time?
It turns out that (on my machine at least) the number of frames successfully
grabbed is marginally less than the number that ought to be grabbed. This
means that the timestamp in the encoded data stream gets behind realtime.
This means that if you say 'Preroll 10', then when the stream gets 10
or more seconds behind, there is no Preroll left.
Fixing this requires a change in the internals of how timestamps are
handled.
@section Does the @code{?date=} stuff work.
Yes (subject to the limitation outlined above). Also note that whenever you
start ffserver, it deletes the ffm file (if any parameters have changed),
thus wiping out what you had recorded before.
The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
of the following formats (the 'T' is literal):
@example
* YYYY-MM-DDTHH:MM:SS (localtime)
* YYYY-MM-DDTHH:MM:SSZ (UTC)
@end example
You can omit the YYYY-MM-DD, and then it refers to the current day. However
note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
may be in the future and so is unlikely to be useful.
You use this by adding the ?date= to the end of the URL for the stream.
For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -f @var{configfile}
Read configuration file @file{configfile}. If not specified it will
read by default from @file{/etc/ffserver.conf}.
@item -n
Enable no-launch mode. This option disables all the @code{Launch}
directives within the various @code{<Feed>} sections. Since
@command{ffserver} will not launch any @command{ffmpeg} instances, you
will have to launch them manually.
@item -d
Enable debug mode. This option increases log verbosity, and directs
log messages to stdout. When specified, the @option{CustomLog} option
is ignored.
@end table
@chapter Configuration file syntax
@command{ffserver} reads a configuration file containing global
options and settings for each stream and feed.
The configuration file consists of global options and dedicated
sections, which must be introduced by "<@var{SECTION_NAME}
@var{ARGS}>" on a separate line and must be terminated by a line in
the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
Currently the following sections are recognized: @samp{Feed},
@samp{Stream}, @samp{Redirect}.
A line starting with @code{#} is ignored and treated as a comment.
Name of options and sections are case-insensitive.
@section ACL syntax
An ACL (Access Control List) specifies the address which are allowed
to access a given stream, or to write a given feed.
It accepts the following forms
@itemize
@item
Allow/deny access to @var{address}.
@example
ACL ALLOW <address>
ACL DENY <address>
@end example
@item
Allow/deny access to ranges of addresses from @var{first_address} to
@var{last_address}.
@example
ACL ALLOW <first_address> <last_address>
ACL DENY <first_address> <last_address>
@end example
@end itemize
You can repeat the ACL allow/deny as often as you like. It is on a per
stream basis. The first match defines the action. If there are no matches,
then the default is the inverse of the last ACL statement.
Thus 'ACL allow localhost' only allows access from localhost.
'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
allow everybody else.
@section Global options
@table @option
@item HTTPPort @var{port_number}
@item Port @var{port_number}
@item RTSPPort @var{port_number}
@var{HTTPPort} sets the HTTP server listening TCP port number,
@var{RTSPPort} sets the RTSP server listening TCP port number.
@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
You must select a different port from your standard HTTP web server if
it is running on the same computer.
If not specified, no corresponding server will be created.
@item HTTPBindAddress @var{ip_address}
@item BindAddress @var{ip_address}
@item RTSPBindAddress @var{ip_address}
Set address on which the HTTP/RTSP server is bound. Only useful if you
have several network interfaces.
@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
deprecated.
@item MaxHTTPConnections @var{n}
Set number of simultaneous HTTP connections that can be handled. It
has to be defined @emph{before} the @option{MaxClients} parameter,
since it defines the @option{MaxClients} maximum limit.
Default value is 2000.
@item MaxClients @var{n}
Set number of simultaneous requests that can be handled. Since
@command{ffserver} is very fast, it is more likely that you will want
to leave this high and use @option{MaxBandwidth}.
Default value is 5.
@item MaxBandwidth @var{kbps}
Set the maximum amount of kbit/sec that you are prepared to consume
when streaming to clients.
Default value is 1000.
@item CustomLog @var{filename}
Set access log file (uses standard Apache log file format). '-' is the
standard output.
If not specified @command{ffserver} will produce no log.
In case the commandline option @option{-d} is specified this option is
ignored, and the log is written to standard output.
@item NoDaemon
Set no-daemon mode. This option is currently ignored since now
@command{ffserver} will always work in no-daemon mode, and is
deprecated.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the all streams or not.
Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
The last occurrence overrides the previous if multiple definitions exist.
@end table
@section Feed section
A Feed section defines a feed provided to @command{ffserver}.
Each live feed contains one video and/or audio sequence coming from an
@command{ffmpeg} encoder or another @command{ffserver}. This sequence
may be encoded simultaneously with several codecs at several
resolutions.
A feed instance specification is introduced by a line in the form:
@example
<Feed FEED_FILENAME>
@end example
where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
The following options are recognized within a Feed section.
@table @option
@item File @var{filename}
@item ReadOnlyFile @var{filename}
Set the path where the feed file is stored on disk.
If not specified, the @file{/tmp/FEED.ffm} is assumed, where
@var{FEED} is the feed name.
If @option{ReadOnlyFile} is used the file is marked as read-only and
it will not be deleted or updated.
@item Truncate
Truncate the feed file, rather than appending to it. By default
@command{ffserver} will append data to the file, until the maximum
file size value is reached (see @option{FileMaxSize} option).
@item FileMaxSize @var{size}
Set maximum size of the feed file in bytes. 0 means unlimited. The
postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
recognized.
Default value is 5M.
@item Launch @var{args}
Launch an @command{ffmpeg} command when creating @command{ffserver}.
@var{args} must be a sequence of arguments to be provided to an
@command{ffmpeg} instance. The first provided argument is ignored, and
it is replaced by a path with the same dirname of the @command{ffserver}
instance, followed by the remaining argument and terminated with a
path corresponding to the feed.
When the launched process exits, @command{ffserver} will launch
another program instance.
In case you need a more complex @command{ffmpeg} configuration,
e.g. if you need to generate multiple FFM feeds with a single
@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
This option is ignored in case the commandline option @option{-n} is
specified.
@item ACL @var{spec}
Specify the list of IP address which are allowed or denied to write
the feed. Multiple ACL options can be specified.
@end table
@section Stream section
A Stream section defines a stream provided by @command{ffserver}, and
identified by a single name.
The stream is sent when answering a request containing the stream
name.
A stream section must be introduced by the line:
@example
<Stream STREAM_NAME>
@end example
where @var{STREAM_NAME} specifies the unique name of the stream.
The following options are recognized within a Stream section.
Encoding options are marked with the @emph{encoding} tag, and they are
used to set the encoding parameters, and are mapped to libavcodec
encoding options. Not all encoding options are supported, in
particular it is not possible to set encoder private options. In order
to override the encoding options specified by @command{ffserver}, you
can use the @command{ffmpeg} @option{override_ffserver} commandline
option.
Only one of the @option{Feed} and @option{File} options should be set.
@table @option
@item Feed @var{feed_name}
Set the input feed. @var{feed_name} must correspond to an existing
feed defined in a @code{Feed} section.
When this option is set, encoding options are used to setup the
encoding operated by the remote @command{ffmpeg} process.
@item File @var{filename}
Set the filename of the pre-recorded input file to stream.
When this option is set, encoding options are ignored and the input
file content is re-streamed as is.
@item Format @var{format_name}
Set the format of the output stream.
Must be the name of a format recognized by FFmpeg. If set to
@samp{status}, it is treated as a status stream.
@item InputFormat @var{format_name}
Set input format. If not specified, it is automatically guessed.
@item Preroll @var{n}
Set this to the number of seconds backwards in time to start. Note that
most players will buffer 5-10 seconds of video, and also you need to allow
for a keyframe to appear in the data stream.
Default value is 0.
@item StartSendOnKey
Do not send stream until it gets the first key frame. By default
@command{ffserver} will send data immediately.
@item MaxTime @var{n}
Set the number of seconds to run. This value set the maximum duration
of the stream a client will be able to receive.
A value of 0 means that no limit is set on the stream duration.
@item ACL @var{spec}
Set ACL for the stream.
@item DynamicACL @var{spec}
@item RTSPOption @var{option}
@item MulticastAddress @var{address}
@item MulticastPort @var{port}
@item MulticastTTL @var{integer}
@item NoLoop
@item FaviconURL @var{url}
Set favicon (favourite icon) for the server status page. It is ignored
for regular streams.
@item Author @var{value}
@item Comment @var{value}
@item Copyright @var{value}
@item Title @var{value}
Set metadata corresponding to the option. All these options are
deprecated in favor of @option{Metadata}.
@item Metadata @var{key} @var{value}
Set metadata value on the output stream.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the stream or not.
Default is @var{UseDefaults} unless disabled globally.
@item NoAudio
@item NoVideo
Suppress audio/video.
@item AudioCodec @var{codec_name} (@emph{encoding,audio})
Set audio codec.
@item AudioBitRate @var{rate} (@emph{encoding,audio})
Set bitrate for the audio stream in kbits per second.
@item AudioChannels @var{n} (@emph{encoding,audio})
Set number of audio channels.
@item AudioSampleRate @var{n} (@emph{encoding,audio})
Set sampling frequency for audio. When using low bitrates, you should
lower this frequency to 22050 or 11025. The supported frequencies
depend on the selected audio codec.
@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
Set generic or private option for audio stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVPresetAudio @var{preset} (@emph{encoding,audio})
Set preset for audio stream.
@item VideoCodec @var{codec_name} (@emph{encoding,video})
Set video codec.
@item VideoBitRate @var{n} (@emph{encoding,video})
Set bitrate for the video stream in kbits per second.
@item VideoBitRateRange @var{range} (@emph{encoding,video})
Set video bitrate range.
A range must be specified in the form @var{minrate}-@var{maxrate}, and
specifies the @option{minrate} and @option{maxrate} encoding options
expressed in kbits per second.
@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
Set video bitrate tolerance in kbits per second.
@item PixelFormat @var{pixel_format} (@emph{encoding,video})
Set video pixel format.
@item Debug @var{integer} (@emph{encoding,video})
Set video @option{debug} encoding option.
@item Strict @var{integer} (@emph{encoding,video})
Set video @option{strict} encoding option.
@item VideoBufferSize @var{n} (@emph{encoding,video})
Set ratecontrol buffer size, expressed in KB.
@item VideoFrameRate @var{n} (@emph{encoding,video})
Set number of video frames per second.
@item VideoSize (@emph{encoding,video})
Set size of the video frame, must be an abbreviation or in the form
@var{W}x@var{H}. See @ref{video size syntax,,the Video size section
in the ffmpeg-utils(1) manual,ffmpeg-utils}.
Default value is @code{160x128}.
@item VideoIntraOnly (@emph{encoding,video})
Transmit only intra frames (useful for low bitrates, but kills frame rate).
@item VideoGopSize @var{n} (@emph{encoding,video})
If non-intra only, an intra frame is transmitted every VideoGopSize
frames. Video synchronization can only begin at an intra frame.
@item VideoTag @var{tag} (@emph{encoding,video})
Set video tag.
@item VideoHighQuality (@emph{encoding,video})
@item Video4MotionVector (@emph{encoding,video})
@item BitExact (@emph{encoding,video})
Set bitexact encoding flag.
@item IdctSimple (@emph{encoding,video})
Set simple IDCT algorithm.
@item Qscale @var{n} (@emph{encoding,video})
Enable constant quality encoding, and set video qscale (quantization
scale) value, expressed in @var{n} QP units.
@item VideoQMin @var{n} (@emph{encoding,video})
@item VideoQMax @var{n} (@emph{encoding,video})
Set video qmin/qmax.
@item VideoQDiff @var{integer} (@emph{encoding,video})
Set video @option{qdiff} encoding option.
@item LumiMask @var{float} (@emph{encoding,video})
@item DarkMask @var{float} (@emph{encoding,video})
Set @option{lumi_mask}/@option{dark_mask} encoding options.
@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
Set generic or private option for video stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVPresetVideo @var{preset} (@emph{encoding,video})
Set preset for video stream.
@var{preset} must be the path of a preset file.
@end table
@subsection Server status stream
A server status stream is a special stream which is used to show
statistics about the @command{ffserver} operations.
It must be specified setting the option @option{Format} to
@samp{status}.
@section Redirect section
A redirect section specifies where to redirect the requested URL to
another page.
A redirect section must be introduced by the line:
@example
<Redirect NAME>
@end example
where @var{NAME} is the name of the page which should be redirected.
It only accepts the option @option{URL}, which specify the redirection
URL.
@chapter Stream examples
@itemize
@item
Multipart JPEG
@example
<Stream test.mjpg>
Feed feed1.ffm
Format mpjpeg
VideoFrameRate 2
VideoIntraOnly
NoAudio
Strict -1
</Stream>
@end example
@item
Single JPEG
@example
<Stream test.jpg>
Feed feed1.ffm
Format jpeg
VideoFrameRate 2
VideoIntraOnly
VideoSize 352x240
NoAudio
Strict -1
</Stream>
@end example
@item
Flash
@example
<Stream test.swf>
Feed feed1.ffm
Format swf
VideoFrameRate 2
VideoIntraOnly
NoAudio
</Stream>
@end example
@item
ASF compatible
@example
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
@end example
@item
MP3 audio
@example
<Stream test.mp3>
Feed feed1.ffm
Format mp2
AudioCodec mp3
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Ogg Vorbis audio
@example
<Stream test.ogg>
Feed feed1.ffm
Metadata title "Stream title"
AudioBitRate 64
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Real with audio only at 32 kbits
@example
<Stream test.ra>
Feed feed1.ffm
Format rm
AudioBitRate 32
NoVideo
</Stream>
@end example
@item
Real with audio and video at 64 kbits
@example
<Stream test.rm>
Feed feed1.ffm
Format rm
AudioBitRate 32
VideoBitRate 128
VideoFrameRate 25
VideoGopSize 25
</Stream>
@end example
@item
For stream coming from a file: you only need to set the input filename
and optionally a new format.
@example
<Stream file.rm>
File "/usr/local/httpd/htdocs/tlive.rm"
NoAudio
</Stream>
@end example
@example
<Stream file.asf>
File "/usr/local/httpd/htdocs/test.asf"
NoAudio
Metadata author "Me"
Metadata copyright "Super MegaCorp"
Metadata title "Test stream from disk"
Metadata comment "Test comment"
</Stream>
@end example
@end itemize
@c man end
@include config.texi
@ifset config-all
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffserver.html,ffserver},
@end ifset
@ifset config-not-all
@url{ffserver-all.html,ffserver-all},
@end ifset
the @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffserver(1),
@end ifset
@ifset config-not-all
ffserver-all(1),
@end ifset
the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffserver
@settitle ffserver video server
@end ignore
@bye
doc/fftools-common-opts.texi
-45
View File
@@ -315,51 +315,6 @@ Possible flags for this option are:
@item k8
@end table
@end table
@item -opencl_bench
This option is used to benchmark all available OpenCL devices and print the
results. This option is only available when FFmpeg has been compiled with
@code{--enable-opencl}.
When FFmpeg is configured with @code{--enable-opencl}, the options for the
global OpenCL context are set via @option{-opencl_options}. See the
"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
supported options. Amongst others, these options include the ability to select
a specific platform and device to run the OpenCL code on. By default, FFmpeg
will run on the first device of the first platform. While the options for the
global OpenCL context provide flexibility to the user in selecting the OpenCL
device of their choice, most users would probably want to select the fastest
OpenCL device for their system.
This option assists the selection of the most efficient configuration by
identifying the appropriate device for the user's system. The built-in
benchmark is run on all the OpenCL devices and the performance is measured for
each device. The devices in the results list are sorted based on their
performance with the fastest device listed first. The user can subsequently
invoke @command{ffmpeg} using the device deemed most appropriate via
@option{-opencl_options} to obtain the best performance for the OpenCL
accelerated code.
Typical usage to use the fastest OpenCL device involve the following steps.
Run the command:
@example
ffmpeg -opencl_bench
@end example
Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
i.e. fastest device in the list.
Select the platform and device using the command:
@example
ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
@end example
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with @code{--enable-opencl}.
@var{options} must be a list of @var{key}=@var{value} option pairs
separated by ':'. See the ``OpenCL Options'' section in the
ffmpeg-utils manual for the list of supported options.
@end table
@section AVOptions
doc/filters.texi
+902 -59
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doc/general.texi
+8 -1
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@@ -365,6 +365,7 @@ library:
@item Interplay MVE @tab @tab X
@tab Format used in various Interplay computer games.
@item Iterated Systems ClearVideo @tab @tab X
@tab I-frames only
@item IV8 @tab @tab X
@tab A format generated by IndigoVision 8000 video server.
@item IVF (On2) @tab X @tab X
@@ -424,6 +425,7 @@ library:
@item NC camera feed @tab @tab X
@tab NC (AVIP NC4600) camera streams
@item NIST SPeech HEader REsources @tab @tab X
@item Computerized Speech Lab NSP @tab @tab X
@item NTT TwinVQ (VQF) @tab @tab X
@tab Nippon Telegraph and Telephone Corporation TwinVQ.
@item Nullsoft Streaming Video @tab @tab X
@@ -438,6 +440,9 @@ library:
@item QCP @tab @tab X
@item raw ADTS (AAC) @tab X @tab X
@item raw AC-3 @tab X @tab X
@item raw AMR-NB @tab @tab X
@item raw AMR-WB @tab @tab X
@item raw aptX @tab X @tab X
@item raw Chinese AVS video @tab X @tab X
@item raw CRI ADX @tab X @tab X
@item raw Dirac @tab X @tab X
@@ -792,7 +797,7 @@ following image formats are supported:
@item LucasArts SANM/Smush @tab @tab X
@tab Used in LucasArts games / SMUSH animations.
@item lossless MJPEG @tab X @tab X
@item MagicYUV Video @tab @tab X
@item MagicYUV Video @tab X @tab X
@item Mandsoft Screen Capture Codec @tab @tab X
@item Microsoft ATC Screen @tab @tab X
@tab Also known as Microsoft Screen 3.
@@ -990,6 +995,8 @@ following image formats are supported:
@item Amazing Studio PAF Audio @tab @tab X
@item Apple lossless audio @tab X @tab X
@tab QuickTime fourcc 'alac'
@item aptX @tab X @tab X
@tab Used in Bluetooth A2DP
@item ATRAC1 @tab @tab X
@item ATRAC3 @tab @tab X
@item ATRAC3+ @tab @tab X
doc/indevs.texi
+6
View File
@@ -238,6 +238,8 @@ This sets the input video format to the format given by the FourCC. To see
the supported values of your device(s) use @option{list_formats}.
Note that there is a FourCC @option{'pal '} that can also be used
as @option{pal} (3 letters).
Default behavior is autodetection of the input video format, if the hardware
supports it.
@item bm_v210
This is a deprecated option, you can use @option{raw_format} instead.
@@ -311,6 +313,10 @@ Sets maximum input buffer size in bytes. If the buffering reaches this value,
incoming frames will be dropped.
Defaults to @samp{1073741824}.
@item audio_depth
Sets the audio sample bit depth. Must be @samp{16} or @samp{32}.
Defaults to @samp{16}.
@end table
@subsection Examples
doc/issue_tracker.txt
-3
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@@ -193,9 +193,6 @@ ffplay
ffprobe
issues in or related to ffprobe.c
ffserver
issues in or related to ffserver.c
postproc
issues in libpostproc/*
doc/libav-merge.txt
+3 -3
View File
@@ -94,13 +94,13 @@ Stuff that didn't reach the codebase:
- a853388d2 hevc: change the stride of the MC buffer to be in bytes instead of elements
- 0cef06df0 checkasm: add HEVC MC tests
- e7078e842 hevcdsp: add x86 SIMD for MC
- VAAPI VP8 decode hwaccel (currently under review: http://ffmpeg.org/pipermail/ffmpeg-devel/2017-February/thread.html#207348)
- Removal of the custom atomic API (5cc0057f49, see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-March/209003.html)
- 7993ec19a hevc: Add hevc_get_pixel_4/8/12/16/24/32/48/64
- new bitstream reader (see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-April/209609.html)
- use of the bsf instead of our parser for vp9 superframes (see fa1749dd34)
- use av_cpu_max_align() instead of hardcoding alignment requirements (see https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/215834.html)
- f44ec22e0 lavc: use av_cpu_max_align() instead of hardcoding alignment requirements
- 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
- Support recovery from an already present HLS playlist (see 16cb06bb30)
- Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
Collateral damage that needs work locally:
------------------------------------------
doc/libavcodec.texi
+2 -2
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@@ -26,13 +26,13 @@ implementing robust and fast codecs as well as for experimentation.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
libavutil(3)
@end ifnothtml
doc/libavdevice.texi
+2 -2
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@@ -23,13 +23,13 @@ VfW, DShow, and ALSA.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-devices(1),
libavutil(3), libavcodec(3), libavformat(3)
@end ifnothtml
doc/libavfilter.texi
+2 -2
View File
@@ -21,14 +21,14 @@ framework containing several filters, sources and sinks.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-filters.html,ffmpeg-filters},
@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-filters(1),
libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
@end ifnothtml
doc/libavformat.texi
+2 -2
View File
@@ -26,13 +26,13 @@ resource.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-formats(1), ffmpeg-protocols(1),
libavutil(3), libavcodec(3)
@end ifnothtml
doc/libavutil.texi
+2 -2
View File
@@ -42,12 +42,12 @@ It should avoid useless features that almost no one needs.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1)
@end ifnothtml
doc/libswresample.texi
+2 -2
View File
@@ -48,13 +48,13 @@ enabled through dedicated options.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-resampler(1),
libavutil(3)
@end ifnothtml
doc/libswscale.texi
+2 -2
View File
@@ -41,13 +41,13 @@ colorspaces differ.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-scaler(1),
libavutil(3)
@end ifnothtml
doc/mailing-list-faq.texi
+1 -2
View File
@@ -27,8 +27,7 @@ for examples.
@item
@url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user}:
For questions involving unscripted usage or compilation of the FFmpeg
command-line tools (@command{ffmpeg}, @command{ffprobe}, @command{ffplay},
@command{ffserver}).
command-line tools (@command{ffmpeg}, @command{ffprobe}, @command{ffplay}).
@item
@url{https://lists.ffmpeg.org/mailman/listinfo/libav-user/, libav-user}:
doc/muxers.texi
+144 -2
View File
@@ -247,6 +247,13 @@ DASH-templated name to used for the initialization segment. Default is "init-str
DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.m4s"
@item -utc_timing_url @var{utc_url}
URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
@item -http_user_agent @var{user_agent}
Override User-Agent field in HTTP header. Applicable only for HTTP output.
@item -http_persistent @var{http_persistent}
Use persistent HTTP connections. Applicable only for HTTP output.
@item -hls_playlist @var{hls_playlist}
Generate HLS playlist files as well. The master playlist is generated with the filename master.m3u8.
One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.
@item -adaptation_sets @var{adaptation_sets}
Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being the IDs
of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.
@@ -468,9 +475,12 @@ By default, the muxer creates a file for each segment produced. These files
have the same name as the playlist, followed by a sequential number and a
.ts extension.
Make sure to require a closed GOP when encoding and to set the GOP
size to fit your segment time constraint.
For example, to convert an input file with @command{ffmpeg}:
@example
ffmpeg -i in.nut out.m3u8
ffmpeg -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 out.m3u8
@end example
This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{out0.ts}, @file{out1.ts}, @file{out2.ts}, etc.
@@ -567,6 +577,31 @@ Should a relative path be specified, the path of the created segment
files will be relative to the current working directory.
When use_localtime_mkdir is set, the whole expanded value of @var{filename} will be written into the m3u8 segment list.
When @code{var_stream_map} is set with two or more variant streams, the
@var{filename} pattern must contain the string "%v", this string specifies
the position of variant stream index in the generated segment file names.
@example
ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
-hls_segment_filename 'file_%v_%03d.ts' out_%v.m3u8
@end example
This example will produce the playlists segment file sets:
@file{file_0_000.ts}, @file{file_0_001.ts}, @file{file_0_002.ts}, etc. and
@file{file_1_000.ts}, @file{file_1_001.ts}, @file{file_1_002.ts}, etc.
The string "%v" may be present in the filename or in the last directory name
containing the file. If the string is present in the directory name, then
sub-directories are created after expanding the directory name pattern. This
enables creation of segments corresponding to different variant streams in
subdirectories.
@example
ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
-hls_segment_filename 'vs%v/file_%03d.ts' vs%v/out.m3u8
@end example
This example will produce the playlists segment file sets:
@file{vs0/file_000.ts}, @file{vs0/file_001.ts}, @file{vs0/file_002.ts}, etc. and
@file{vs1/file_000.ts}, @file{vs1/file_001.ts}, @file{vs1/file_002.ts}, etc.
@item use_localtime
Use strftime() on @var{filename} to expand the segment filename with localtime.
@@ -693,6 +728,15 @@ the fmp4 files is used in hls after version 7.
@item hls_fmp4_init_filename @var{filename}
set filename to the fragment files header file, default filename is @file{init.mp4}.
When @code{var_stream_map} is set with two or more variant streams, the
@var{filename} pattern must contain the string "%v", this string specifies
the position of variant stream index in the generated init file names.
The string "%v" may be present in the filename or in the last directory name
containing the file. If the string is present in the directory name, then
sub-directories are created after expanding the directory name pattern. This
enables creation of init files corresponding to different variant streams in
subdirectories.
@item hls_flags @var{flags}
Possible values:
@@ -732,6 +776,10 @@ The file specified by @code{hls_key_info_file} will be checked periodically and
detect updates to the encryption info. Be sure to replace this file atomically,
including the file containing the AES encryption key.
@item independent_segments
Add the @code{#EXT-X-INDEPENDENT-SEGMENTS} to playlists that has video segments
and when all the segments of that playlist are guaranteed to start with a Key frame.
@item split_by_time
Allow segments to start on frames other than keyframes. This improves
behavior on some players when the time between keyframes is inconsistent,
@@ -794,6 +842,88 @@ files.
@item http_user_agent
Override User-Agent field in HTTP header. Applicable only for HTTP output.
@item var_stream_map
Map string which specifies how to group the audio, video and subtitle streams
into different variant streams. The variant stream groups are separated
by space.
Expected string format is like this "a:0,v:0 a:1,v:1 ....". Here a:, v:, s: are
the keys to specify audio, video and subtitle streams respectively.
Allowed values are 0 to 9 (limited just based on practical usage).
When there are two or more variant streams, the output filename pattern must
contain the string "%v", this string specifies the position of variant stream
index in the output media playlist filenames. The string "%v" may be present in
the filename or in the last directory name containing the file. If the string is
present in the directory name, then sub-directories are created after expanding
the directory name pattern. This enables creation of variant streams in
subdirectories.
@example
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
http://example.com/live/out_%v.m3u8
@end example
This example creates two hls variant streams. The first variant stream will
contain video stream of bitrate 1000k and audio stream of bitrate 64k and the
second variant stream will contain video stream of bitrate 256k and audio
stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and
out_1.m3u8 will be created.
@example
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \
-map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \
http://example.com/live/out_%v.m3u8
@end example
This example creates three hls variant streams. The first variant stream will
be a video only stream with video bitrate 1000k, the second variant stream will
be an audio only stream with bitrate 64k and the third variant stream will be a
video only stream with bitrate 256k. Here, three media playlist with file names
out_0.m3u8, out_1.m3u8 and out_2.m3u8 will be created.
@example
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
-map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
http://example.com/live/vs_%v/out.m3u8
@end example
This example creates the variant streams in subdirectories. Here, the first
media playlist is created at @file{http://example.com/live/vs_0/out.m3u8} and
the second one at @file{http://example.com/live/vs_1/out.m3u8}.
@example
ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k -b:v:1 3000k \
-map 0:a -map 0:a -map 0:v -map 0:v -f hls \
-var_stream_map "a:0,agroup:aud_low a:1,agroup:aud_high v:0,agroup:aud_low v:1,agroup:aud_high" \
-master_pl_name master.m3u8 \
http://example.com/live/out_%v.m3u8
@end example
This example creates two audio only and two video only variant streams. In
addition to the #EXT-X-STREAM-INF tag for each variant stream in the master
playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams
and they are mapped to the two video only variant streams with audio group names
'aud_low' and 'aud_high'.
By default, a single hls variant containing all the encoded streams is created.
@item master_pl_name
Create HLS master playlist with the given name.
@example
ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8
@end example
This example creates HLS master playlist with name master.m3u8 and it is
published at http://example.com/live/
@item master_pl_publish_rate
Publish master play list repeatedly every after specified number of segment intervals.
@example
ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 \
-hls_time 2 -master_pl_publish_rate 30 http://example.com/live/out.m3u8
@end example
This example creates HLS master playlist with name master.m3u8 and keep
publishing it repeatedly every after 30 segments i.e. every after 60s.
@item http_persistent
Use persistent HTTP connections. Applicable only for HTTP output.
@end table
@anchor{ico}
@@ -894,9 +1024,18 @@ can be used:
ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
@end example
You can set the file name with current frame's PTS:
@example
ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg"
@end example
@subsection Options
@table @option
@item frame_pts
If set to 1, expand the filename with pts from pkt->pts.
Default value is 0.
@item start_number
Start the sequence from the specified number. Default value is 1.
@@ -1649,6 +1788,9 @@ segment would usually span. Otherwise, the segment will be filled with the next
packet written. Defaults to @code{0}.
@end table
Make sure to require a closed GOP when encoding and to set the GOP
size to fit your segment time constraint.
@subsection Examples
@itemize
@@ -1657,7 +1799,7 @@ Remux the content of file @file{in.mkv} to a list of segments
@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
generated segments to @file{out.list}:
@example
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut
ffmpeg -i in.mkv -codec hevc -flags +cgop -g 60 -map 0 -f segment -segment_list out.list out%03d.nut
@end example
@item
doc/protocols.texi
+14 -2
View File
@@ -321,6 +321,9 @@ Sets the maximum delay in seconds after which to give up reconnecting
@item mime_type
Export the MIME type.
@item http_version
Exports the HTTP response version number. Usually "1.0" or "1.1".
@item icy
If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
supports this, the metadata has to be retrieved by the application by reading
@@ -363,7 +366,7 @@ If set to 1 enables experimental HTTP server. This can be used to send data when
used as an output option, or read data from a client with HTTP POST when used as
an input option.
If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
in ffmpeg.c and thus must not be used as a command line option.
@example
# Server side (sending):
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
@@ -1186,6 +1189,7 @@ Accepted options:
Start offset of the extracted segment, in bytes.
@item end
End offset of the extracted segment, in bytes.
If set to 0, extract till end of file.
@end table
Examples:
@@ -1201,6 +1205,11 @@ Play an AVI file directly from a TAR archive:
subfile,,start,183241728,end,366490624,,:archive.tar
@end example
Play a MPEG-TS file from start offset till end:
@example
subfile,,start,32815239,end,0,,:video.ts
@end example
@section tee
Writes the output to multiple protocols. The individual outputs are separated
@@ -1242,6 +1251,9 @@ Set receive buffer size, expressed bytes.
@item send_buffer_size=@var{bytes}
Set send buffer size, expressed bytes.
@item tcp_nodelay=@var{1|0}
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
@end table
The following example shows how to setup a listening TCP connection
@@ -1277,7 +1289,7 @@ If enabled, try to verify the peer that we are communicating with.
Note, if using OpenSSL, this currently only makes sure that the
peer certificate is signed by one of the root certificates in the CA
database, but it does not validate that the certificate actually
matches the host name we are trying to connect to. (With GnuTLS,
matches the host name we are trying to connect to. (With other backends,
the host name is validated as well.)
This is disabled by default since it requires a CA database to be
doc/utils.texi
-30
View File
@@ -1057,33 +1057,3 @@ indication of the corresponding powers of 10 and of 2.
@end table
@c man end EXPRESSION EVALUATION
@chapter OpenCL Options
@c man begin OPENCL OPTIONS
When FFmpeg is configured with @code{--enable-opencl}, it is possible
to set the options for the global OpenCL context.
The list of supported options follows:
@table @option
@item build_options
Set build options used to compile the registered kernels.
See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
@item platform_idx
Select the index of the platform to run OpenCL code.
The specified index must be one of the indexes in the device list
which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
@item device_idx
Select the index of the device used to run OpenCL code.
The specified index must be one of the indexes in the device list which
can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
@end table
@c man end OPENCL OPTIONS
ffbuild/.gitignore
+5
View File
@@ -0,0 +1,5 @@
/.config
/config.fate
/config.log
/config.mak
/config.sh
ffbuild/common.mak
+2 -3
View File
@@ -119,7 +119,7 @@ FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
LDLIBS = $(FFLIBS:%=%$(BUILDSUF))
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=EXTRALIBS-%),$($(lib))) $(EXTRALIBS)
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
@@ -163,8 +163,7 @@ $(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.o *~ *.h.c *.gcda *.gcno *.map *.ver *.version *.ho *$(DEFAULT_X86ASMD).asm *.ptx *.ptx.c
DISTCLEANSUFFIXES = *.pc
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
define RULES
ffbuild/library.mak
+9 -9
View File
@@ -1,6 +1,6 @@
include $(SRC_PATH)/ffbuild/common.mak
ifeq (,$(filter %clean,$(MAKECMDGOALS)))
ifeq (,$(filter %clean config,$(MAKECMDGOALS)))
-include $(SUBDIR)lib$(NAME).version
endif
@@ -31,13 +31,15 @@ define RULES
$(TOOLS): THISLIB = $(FULLNAME:%=$(LD_LIB))
$(TESTPROGS): THISLIB = $(SUBDIR)$(LIBNAME)
$(LIBOBJS): CPPFLAGS += -DBUILDING_$(NAME)
$(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
$$(LD) $(LDFLAGS) $(LDEXEFLAGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $(FFEXTRALIBS) $$(ELIBS)
$$(LD) $(LDFLAGS) $(LDEXEFLAGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $(FFEXTRALIBS) $$(EXTRALIBS-$$(*F)) $$(ELIBS)
$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h | $(SUBDIR)
$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$< > $$@
$(SUBDIR)lib$(FULLNAME).pc: $(SUBDIR)version.h | $(SUBDIR)
$(SUBDIR)lib$(FULLNAME).pc: $(SUBDIR)version.h ffbuild/config.sh | $(SUBDIR)
$$(M) $$(SRC_PATH)/ffbuild/pkgconfig_generate.sh $(NAME) "$(DESC)"
$(SUBDIR)lib$(NAME).ver: $(SUBDIR)lib$(NAME).v $(OBJS)
@@ -48,7 +50,7 @@ $(SUBDIR)$(SLIBNAME): $(SUBDIR)$(SLIBNAME_WITH_MAJOR)
$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
$(SLIB_CREATE_DEF_CMD)
$$(LD) $(SHFLAGS) $(LDFLAGS) $(LDLIBFLAGS) $$(LD_O) $$(filter %.o,$$^) $(FFEXTRALIBS)
$$(LD) $(SHFLAGS) $(LDFLAGS) $(LDSOFLAGS) $$(LD_O) $$(filter %.o,$$^) $(FFEXTRALIBS)
$(SLIB_EXTRA_CMD)
ifdef SUBDIR
@@ -59,10 +61,6 @@ clean::
$(RM) $(addprefix $(SUBDIR),$(CLEANFILES) $(CLEANSUFFIXES) $(LIBSUFFIXES)) \
$(CLEANSUFFIXES:%=$(SUBDIR)$(ARCH)/%) $(CLEANSUFFIXES:%=$(SUBDIR)tests/%)
distclean:: clean
$(RM) $(DISTCLEANSUFFIXES:%=$(SUBDIR)%) $(DISTCLEANSUFFIXES:%=$(SUBDIR)$(ARCH)/%) \
$(DISTCLEANSUFFIXES:%=$(SUBDIR)tests/%)
install-lib$(NAME)-shared: $(SUBDIR)$(SLIBNAME)
$(Q)mkdir -p "$(SHLIBDIR)"
$$(INSTALL) -m 755 $$< "$(SHLIBDIR)/$(SLIB_INSTALL_NAME)"
@@ -95,8 +93,10 @@ uninstall-libs::
uninstall-headers::
$(RM) $(addprefix "$(INCINSTDIR)/",$(HEADERS) $(BUILT_HEADERS))
$(RM) "$(PKGCONFIGDIR)/lib$(FULLNAME).pc"
-rmdir "$(INCINSTDIR)"
uninstall-pkgconfig::
$(RM) "$(PKGCONFIGDIR)/lib$(FULLNAME).pc"
endef
$(eval $(RULES))
fftools/Makefile
+1 -4
View File
@@ -1,12 +1,11 @@
AVPROGS-$(CONFIG_FFMPEG) += ffmpeg
AVPROGS-$(CONFIG_FFPLAY) += ffplay
AVPROGS-$(CONFIG_FFPROBE) += ffprobe
AVPROGS-$(CONFIG_FFSERVER) += ffserver
AVPROGS := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
PROGS += $(AVPROGS)
AVBASENAMES = ffmpeg ffplay ffprobe ffserver
AVBASENAMES = ffmpeg ffplay ffprobe
ALLAVPROGS = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
@@ -17,7 +16,6 @@ ifndef CONFIG_VIDEOTOOLBOX
OBJS-ffmpeg-$(CONFIG_VDA) += fftools/ffmpeg_videotoolbox.o
endif
OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += fftools/ffmpeg_videotoolbox.o
OBJS-ffserver += fftools/ffserver_config.o
define DOFFTOOL
OBJS-$(1) += fftools/cmdutils.o fftools/$(1).o $(OBJS-$(1)-yes)
@@ -29,7 +27,6 @@ $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(EXTRALIBS-$(1))
-include $$(OBJS-$(1):.o=.d)
endef
$(foreach P,$(AVPROGS-yes),$(eval OBJS-$(P)-$(CONFIG_OPENCL) += fftools/cmdutils_opencl.o))
$(foreach P,$(AVPROGS-yes),$(eval $(call DOFFTOOL,$(P))))
all: $(AVPROGS)
fftools/cmdutils.c
+2 -1
View File
@@ -38,6 +38,7 @@
#include "libswscale/swscale.h"
#include "libswresample/swresample.h"
#include "libpostproc/postprocess.h"
#include "libavutil/attributes.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
@@ -1702,7 +1703,7 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg)
#endif
while ((pix_desc = av_pix_fmt_desc_next(pix_desc))) {
enum AVPixelFormat pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
enum AVPixelFormat av_unused pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
printf("%c%c%c%c%c %-16s %d %2d\n",
sws_isSupportedInput (pix_fmt) ? 'I' : '.',
sws_isSupportedOutput(pix_fmt) ? 'O' : '.',
fftools/cmdutils.h
+1 -18
View File
@@ -105,12 +105,6 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg);
int opt_codec_debug(void *optctx, const char *opt, const char *arg);
#if CONFIG_OPENCL
int opt_opencl(void *optctx, const char *opt, const char *arg);
int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
#endif
/**
* Limit the execution time.
*/
@@ -155,6 +149,7 @@ typedef struct SpecifierOpt {
uint8_t *str;
int i;
int64_t i64;
uint64_t ui64;
float f;
double dbl;
} u;
@@ -206,17 +201,6 @@ typedef struct OptionDef {
void show_help_options(const OptionDef *options, const char *msg, int req_flags,
int rej_flags, int alt_flags);
#if CONFIG_OPENCL
#define CMDUTILS_COMMON_OPTIONS_OPENCL \
{ "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, \
"run benchmark on all OpenCL devices and show results" }, \
{ "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, \
"set OpenCL environment options" }, \
#else
#define CMDUTILS_COMMON_OPTIONS_OPENCL
#endif
#if CONFIG_AVDEVICE
#define CMDUTILS_COMMON_OPTIONS_AVDEVICE \
{ "sources" , OPT_EXIT | HAS_ARG, { .func_arg = show_sources }, \
@@ -256,7 +240,6 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
{ "max_alloc", HAS_ARG, { .func_arg = opt_max_alloc }, "set maximum size of a single allocated block", "bytes" }, \
{ "cpuflags", HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" }, \
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" }, \
CMDUTILS_COMMON_OPTIONS_OPENCL \
CMDUTILS_COMMON_OPTIONS_AVDEVICE \
/**
fftools/cmdutils_opencl.c
-283
View File
@@ -1,283 +0,0 @@
/*
* Copyright (C) 2013 Lenny Wang
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavutil/log.h"
#include "libavutil/opencl.h"
#include "libavutil/avstring.h"
#include "cmdutils.h"
typedef struct {
int platform_idx;
int device_idx;
char device_name[64];
int64_t runtime;
} OpenCLDeviceBenchmark;
const char *ocl_bench_source = AV_OPENCL_KERNEL(
inline unsigned char clip_uint8(int a)
{
if (a & (~0xFF))
return (-a)>>31;
else
return a;
}
kernel void unsharp_bench(
global unsigned char *src,
global unsigned char *dst,
global int *mask,
int width,
int height)
{
int i, j, local_idx, lc_idx, sum = 0;
int2 thread_idx, block_idx, global_idx, lm_idx;
thread_idx.x = get_local_id(0);
thread_idx.y = get_local_id(1);
block_idx.x = get_group_id(0);
block_idx.y = get_group_id(1);
global_idx.x = get_global_id(0);
global_idx.y = get_global_id(1);
local uchar data[32][32];
local int lc[128];
for (i = 0; i <= 1; i++) {
lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
for (j = 0; j <= 1; j++) {
lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
}
}
local_idx = thread_idx.y*16 + thread_idx.x;
if (local_idx < 128)
lc[local_idx] = mask[local_idx];
barrier(CLK_LOCAL_MEM_FENCE);
\n#pragma unroll\n
for (i = -4; i <= 4; i++) {
lm_idx.y = 8 + i + thread_idx.y;
\n#pragma unroll\n
for (j = -4; j <= 4; j++) {
lm_idx.x = 8 + j + thread_idx.x;
lc_idx = (i + 4)*8 + j + 4;
sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
}
}
int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
if (global_idx.x < width && global_idx.y < height)
dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
}
);
#define OCLCHECK(method, ... ) \
do { \
status = method(__VA_ARGS__); \
if (status != CL_SUCCESS) { \
av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n", \
av_opencl_errstr(status)); \
ret = AVERROR_EXTERNAL; \
goto end; \
} \
} while (0)
#define CREATEBUF(out, flags, size) \
do { \
out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
if (status != CL_SUCCESS) { \
av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n"); \
ret = AVERROR_EXTERNAL; \
goto end; \
} \
} while (0)
static void fill_rand_int(int *data, int n)
{
int i;
srand(av_gettime());
for (i = 0; i < n; i++)
data[i] = rand();
}
#define OPENCL_NB_ITER 5
static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
{
int i, arg = 0, width = 1920, height = 1088;
int64_t start, ret = 0;
cl_int status;
size_t kernel_len;
char *inbuf;
int *mask = NULL;
int buf_size = width * height * sizeof(char);
int mask_size = sizeof(uint32_t) * 128;
cl_mem cl_mask = NULL, cl_inbuf = NULL, cl_outbuf = NULL;
cl_kernel kernel = NULL;
cl_program program = NULL;
size_t local_work_size_2d[2] = {16, 16};
size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
ret = AVERROR(ENOMEM);
goto end;
}
fill_rand_int((int*)inbuf, buf_size/4);
fill_rand_int(mask, mask_size/4);
CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
kernel_len = strlen(ocl_bench_source);
program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
&kernel_len, &status);
if (status != CL_SUCCESS || !program) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
ret = AVERROR_EXTERNAL;
goto end;
}
status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
if (status != CL_SUCCESS) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
ret = AVERROR_EXTERNAL;
goto end;
}
kernel = clCreateKernel(program, "unsharp_bench", &status);
if (status != CL_SUCCESS) {
av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
ret = AVERROR_EXTERNAL;
goto end;
}
OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
buf_size, inbuf, 0, NULL, NULL);
OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
mask_size, mask, 0, NULL, NULL);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
start = av_gettime_relative();
for (i = 0; i < OPENCL_NB_ITER; i++)
OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
clFinish(ext_opencl_env->command_queue);
ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
end:
if (kernel)
clReleaseKernel(kernel);
if (program)
clReleaseProgram(program);
if (cl_inbuf)
clReleaseMemObject(cl_inbuf);
if (cl_outbuf)
clReleaseMemObject(cl_outbuf);
if (cl_mask)
clReleaseMemObject(cl_mask);
av_free(inbuf);
av_free(mask);
return ret;
}
static int compare_ocl_device_desc(const void *a, const void *b)
{
const OpenCLDeviceBenchmark* va = (const OpenCLDeviceBenchmark*)a;
const OpenCLDeviceBenchmark* vb = (const OpenCLDeviceBenchmark*)b;
return FFDIFFSIGN(va->runtime , vb->runtime);
}
int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
{
int i, j, nb_devices = 0, count = 0, ret = 0;
int64_t score = 0;
AVOpenCLDeviceList *device_list;
AVOpenCLDeviceNode *device_node = NULL;
OpenCLDeviceBenchmark *devices = NULL;
cl_platform_id platform;
ret = av_opencl_get_device_list(&device_list);
if (ret < 0) {
return ret;
}
for (i = 0; i < device_list->platform_num; i++)
nb_devices += device_list->platform_node[i]->device_num;
if (!nb_devices) {
av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
av_opencl_free_device_list(&device_list);
return AVERROR(EINVAL);
}
if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
av_opencl_free_device_list(&device_list);
return AVERROR(ENOMEM);
}
for (i = 0; i < device_list->platform_num; i++) {
for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
device_node = device_list->platform_node[i]->device_node[j];
platform = device_list->platform_node[i]->platform_id;
score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
if (score > 0) {
devices[count].platform_idx = i;
devices[count].device_idx = j;
devices[count].runtime = score;
av_strlcpy(devices[count].device_name, device_node->device_name,
sizeof(devices[count].device_name));
count++;
}
}
}
qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
for (i = 0; i < count; i++)
fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
devices[i].platform_idx, devices[i].device_idx,
devices[i].device_name, devices[i].runtime);
av_opencl_free_device_list(&device_list);
av_free(devices);
return 0;
}
int opt_opencl(void *optctx, const char *opt, const char *arg)
{
char *key, *value;
const char *opts = arg;
int ret = 0;
while (*opts) {
ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
if (ret < 0)
return ret;
ret = av_opencl_set_option(key, value);
if (ret < 0)
return ret;
if (*opts)
opts++;
}
return ret;
}
fftools/ffmpeg.c
+106 -87
View File
@@ -61,6 +61,7 @@
#include "libavutil/timestamp.h"
#include "libavutil/bprint.h"
#include "libavutil/time.h"
#include "libavutil/thread.h"
#include "libavutil/threadmessage.h"
#include "libavcodec/mathops.h"
#include "libavformat/os_support.h"
@@ -98,10 +99,6 @@
#include <conio.h>
#endif
#if HAVE_PTHREADS
#include <pthread.h>
#endif
#include <time.h>
#include "ffmpeg.h"
@@ -161,7 +158,7 @@ static struct termios oldtty;
static int restore_tty;
#endif
#if HAVE_PTHREADS
#if HAVE_THREADS
static void free_input_threads(void);
#endif
@@ -220,13 +217,18 @@ static void sub2video_push_ref(InputStream *ist, int64_t pts)
{
AVFrame *frame = ist->sub2video.frame;
int i;
int ret;
av_assert1(frame->data[0]);
ist->sub2video.last_pts = frame->pts = pts;
for (i = 0; i < ist->nb_filters; i++)
av_buffersrc_add_frame_flags(ist->filters[i]->filter, frame,
for (i = 0; i < ist->nb_filters; i++) {
ret = av_buffersrc_add_frame_flags(ist->filters[i]->filter, frame,
AV_BUFFERSRC_FLAG_KEEP_REF |
AV_BUFFERSRC_FLAG_PUSH);
if (ret != AVERROR_EOF && ret < 0)
av_log(NULL, AV_LOG_WARNING, "Error while add the frame to buffer source(%s).\n",
av_err2str(ret));
}
}
void sub2video_update(InputStream *ist, AVSubtitle *sub)
@@ -295,11 +297,15 @@ static void sub2video_heartbeat(InputStream *ist, int64_t pts)
static void sub2video_flush(InputStream *ist)
{
int i;
int ret;
if (ist->sub2video.end_pts < INT64_MAX)
sub2video_update(ist, NULL);
for (i = 0; i < ist->nb_filters; i++)
av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
for (i = 0; i < ist->nb_filters; i++) {
ret = av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
if (ret != AVERROR_EOF && ret < 0)
av_log(NULL, AV_LOG_WARNING, "Flush the frame error.\n");
}
}
/* end of sub2video hack */
@@ -327,13 +333,14 @@ static int main_return_code = 0;
static void
sigterm_handler(int sig)
{
int ret;
received_sigterm = sig;
received_nb_signals++;
term_exit_sigsafe();
if(received_nb_signals > 3) {
write(2/*STDERR_FILENO*/, "Received > 3 system signals, hard exiting\n",
ret = write(2/*STDERR_FILENO*/, "Received > 3 system signals, hard exiting\n",
strlen("Received > 3 system signals, hard exiting\n"));
if (ret < 0) { /* Do nothing */ };
exit(123);
}
}
@@ -568,7 +575,7 @@ static void ffmpeg_cleanup(int ret)
av_freep(&output_streams[i]);
}
#if HAVE_PTHREADS
#if HAVE_THREADS
free_input_threads();
#endif
for (i = 0; i < nb_input_files; i++) {
@@ -1199,24 +1206,6 @@ static void do_video_out(OutputFile *of,
#endif
return;
#if FF_API_LAVF_FMT_RAWPICTURE
if (of->ctx->oformat->flags & AVFMT_RAWPICTURE &&
enc->codec->id == AV_CODEC_ID_RAWVIDEO) {
/* raw pictures are written as AVPicture structure to
avoid any copies. We support temporarily the older
method. */
if (in_picture->interlaced_frame)
mux_par->field_order = in_picture->top_field_first ? AV_FIELD_TB:AV_FIELD_BT;
else
mux_par->field_order = AV_FIELD_PROGRESSIVE;
pkt.data = (uint8_t *)in_picture;
pkt.size = sizeof(AVPicture);
pkt.pts = av_rescale_q(in_picture->pts, enc->time_base, ost->mux_timebase);
pkt.flags |= AV_PKT_FLAG_KEY;
output_packet(of, &pkt, ost, 0);
} else
#endif
{
int forced_keyframe = 0;
double pts_time;
@@ -1897,10 +1886,6 @@ static void flush_encoders(void)
if (enc->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <= 1)
continue;
#if FF_API_LAVF_FMT_RAWPICTURE
if (enc->codec_type == AVMEDIA_TYPE_VIDEO && (of->ctx->oformat->flags & AVFMT_RAWPICTURE) && enc->codec->id == AV_CODEC_ID_RAWVIDEO)
continue;
#endif
if (enc->codec_type != AVMEDIA_TYPE_VIDEO && enc->codec_type != AVMEDIA_TYPE_AUDIO)
continue;
@@ -1991,11 +1976,16 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
InputFile *f = input_files [ist->file_index];
int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->mux_timebase);
AVPicture pict;
AVPacket opkt;
AVPacket opkt = { 0 };
av_init_packet(&opkt);
// EOF: flush output bitstream filters.
if (!pkt) {
output_packet(of, &opkt, ost, 1);
return;
}
if ((!ost->frame_number && !(pkt->flags & AV_PKT_FLAG_KEY)) &&
!ost->copy_initial_nonkeyframes)
return;
@@ -2079,23 +2069,6 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
}
av_copy_packet_side_data(&opkt, pkt);
#if FF_API_LAVF_FMT_RAWPICTURE
if (ost->st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO &&
ost->st->codecpar->codec_id == AV_CODEC_ID_RAWVIDEO &&
(of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
/* store AVPicture in AVPacket, as expected by the output format */
int ret = avpicture_fill(&pict, opkt.data, ost->st->codecpar->format, ost->st->codecpar->width, ost->st->codecpar->height);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "avpicture_fill failed: %s\n",
av_err2str(ret));
exit_program(1);
}
opkt.data = (uint8_t *)&pict;
opkt.size = sizeof(AVPicture);
opkt.flags |= AV_PKT_FLAG_KEY;
}
#endif
output_packet(of, &opkt, ost, 0);
}
@@ -2733,7 +2706,7 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt, int no_eo
}
/* handle stream copy */
if (!ist->decoding_needed) {
if (!ist->decoding_needed && pkt) {
ist->dts = ist->next_dts;
switch (ist->dec_ctx->codec_type) {
case AVMEDIA_TYPE_AUDIO:
@@ -2759,7 +2732,7 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt, int no_eo
ist->pts = ist->dts;
ist->next_pts = ist->next_dts;
}
for (i = 0; pkt && i < nb_output_streams; i++) {
for (i = 0; i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
if (!check_output_constraints(ist, ost) || ost->encoding_needed)
@@ -2816,44 +2789,77 @@ fail:
av_freep(&avc);
}
static const HWAccel *get_hwaccel(enum AVPixelFormat pix_fmt)
{
int i;
for (i = 0; hwaccels[i].name; i++)
if (hwaccels[i].pix_fmt == pix_fmt)
return &hwaccels[i];
return NULL;
}
static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat *pix_fmts)
{
InputStream *ist = s->opaque;
const enum AVPixelFormat *p;
int ret;
for (p = pix_fmts; *p != -1; p++) {
for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(*p);
const HWAccel *hwaccel;
const AVCodecHWConfig *config = NULL;
int i;
if (!(desc->flags & AV_PIX_FMT_FLAG_HWACCEL))
break;
hwaccel = get_hwaccel(*p);
if (!hwaccel ||
(ist->active_hwaccel_id && ist->active_hwaccel_id != hwaccel->id) ||
(ist->hwaccel_id != HWACCEL_AUTO && ist->hwaccel_id != hwaccel->id))
if (ist->hwaccel_id == HWACCEL_GENERIC ||
ist->hwaccel_id == HWACCEL_AUTO) {
for (i = 0;; i++) {
config = avcodec_get_hw_config(s->codec, i);
if (!config)
break;
if (!(config->methods &
AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX))
continue;
if (config->pix_fmt == *p)
break;
}
}
if (config) {
if (config->device_type != ist->hwaccel_device_type) {
// Different hwaccel offered, ignore.
continue;
}
ret = hwaccel_decode_init(s);
if (ret < 0) {
if (ist->hwaccel_id == HWACCEL_GENERIC) {
av_log(NULL, AV_LOG_FATAL,
"%s hwaccel requested for input stream #%d:%d, "
"but cannot be initialized.\n",
av_hwdevice_get_type_name(config->device_type),
ist->file_index, ist->st->index);
return AV_PIX_FMT_NONE;
}
continue;
}
} else {
const HWAccel *hwaccel = NULL;
int i;
for (i = 0; hwaccels[i].name; i++) {
if (hwaccels[i].pix_fmt == *p) {
hwaccel = &hwaccels[i];
break;
}
}
if (!hwaccel) {
// No hwaccel supporting this pixfmt.
continue;
}
if (hwaccel->id != ist->hwaccel_id) {
// Does not match requested hwaccel.
continue;
}
ret = hwaccel->init(s);
if (ret < 0) {
if (ist->hwaccel_id == hwaccel->id) {
av_log(NULL, AV_LOG_FATAL,
"%s hwaccel requested for input stream #%d:%d, "
"but cannot be initialized.\n", hwaccel->name,
ist->file_index, ist->st->index);
return AV_PIX_FMT_NONE;
}
continue;
}
if (ist->hw_frames_ctx) {
@@ -2862,7 +2868,6 @@ static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat
return AV_PIX_FMT_NONE;
}
ist->active_hwaccel_id = hwaccel->id;
ist->hwaccel_pix_fmt = *p;
break;
}
@@ -2910,7 +2915,7 @@ static int init_input_stream(int ist_index, char *error, int error_len)
/* Useful for subtitles retiming by lavf (FIXME), skipping samples in
* audio, and video decoders such as cuvid or mediacodec */
av_codec_set_pkt_timebase(ist->dec_ctx, ist->st->time_base);
ist->dec_ctx->pkt_timebase = ist->st->time_base;
if (!av_dict_get(ist->decoder_opts, "threads", NULL, 0))
av_dict_set(&ist->decoder_opts, "threads", "auto", 0);
@@ -3160,7 +3165,7 @@ static void set_encoder_id(OutputFile *of, OutputStream *ost)
uint8_t *encoder_string;
int encoder_string_len;
int format_flags = 0;
int codec_flags = 0;
int codec_flags = ost->enc_ctx->flags;
if (av_dict_get(ost->st->metadata, "encoder", NULL, 0))
return;
@@ -3988,7 +3993,7 @@ static int check_keyboard_interaction(int64_t cur_time)
return 0;
}
#if HAVE_PTHREADS
#if HAVE_THREADS
static void *input_thread(void *arg)
{
InputFile *f = arg;
@@ -4098,7 +4103,7 @@ static int get_input_packet(InputFile *f, AVPacket *pkt)
}
}
#if HAVE_PTHREADS
#if HAVE_THREADS
if (nb_input_files > 1)
return get_input_packet_mt(f, pkt);
#endif
@@ -4180,14 +4185,17 @@ static int seek_to_start(InputFile *ifile, AVFormatContext *is)
AVRational sample_rate = {1, avctx->sample_rate};
duration = av_rescale_q(ist->nb_samples, sample_rate, ist->st->time_base);
} else
} else {
continue;
}
} else {
if (ist->framerate.num) {
duration = av_rescale_q(1, ist->framerate, ist->st->time_base);
duration = av_rescale_q(1, av_inv_q(ist->framerate), ist->st->time_base);
} else if (ist->st->avg_frame_rate.num) {
duration = av_rescale_q(1, ist->st->avg_frame_rate, ist->st->time_base);
} else duration = 1;
duration = av_rescale_q(1, av_inv_q(ist->st->avg_frame_rate), ist->st->time_base);
} else {
duration = 1;
}
}
if (!ifile->duration)
ifile->time_base = ist->st->time_base;
@@ -4229,8 +4237,10 @@ static int process_input(int file_index)
return ret;
}
if (ret < 0 && ifile->loop) {
if ((ret = seek_to_start(ifile, is)) < 0)
return ret;
ret = seek_to_start(ifile, is);
if (ret < 0)
av_log(NULL, AV_LOG_WARNING, "Seek to start failed.\n");
else
ret = get_input_packet(ifile, &pkt);
if (ret == AVERROR(EAGAIN)) {
ifile->eagain = 1;
@@ -4536,6 +4546,15 @@ static int transcode_step(void)
}
if (ost->filter && ost->filter->graph->graph) {
if (!ost->initialized) {
char error[1024] = {0};
ret = init_output_stream(ost, error, sizeof(error));
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error initializing output stream %d:%d -- %s\n",
ost->file_index, ost->index, error);
exit_program(1);
}
}
if ((ret = transcode_from_filter(ost->filter->graph, &ist)) < 0)
return ret;
if (!ist)
@@ -4593,7 +4612,7 @@ static int transcode(void)
timer_start = av_gettime_relative();
#if HAVE_PTHREADS
#if HAVE_THREADS
if ((ret = init_input_threads()) < 0)
goto fail;
#endif
@@ -4624,14 +4643,14 @@ static int transcode(void)
/* dump report by using the output first video and audio streams */
print_report(0, timer_start, cur_time);
}
#if HAVE_PTHREADS
#if HAVE_THREADS
free_input_threads();
#endif
/* at the end of stream, we must flush the decoder buffers */
for (i = 0; i < nb_input_streams; i++) {
ist = input_streams[i];
if (!input_files[ist->file_index]->eof_reached && ist->decoding_needed) {
if (!input_files[ist->file_index]->eof_reached) {
process_input_packet(ist, NULL, 0);
}
}
@@ -4690,7 +4709,7 @@ static int transcode(void)
ret = 0;
fail:
#if HAVE_PTHREADS
#if HAVE_THREADS
free_input_threads();
#endif
fftools/ffmpeg.h
+5 -14
View File
@@ -25,10 +25,6 @@
#include <stdio.h>
#include <signal.h>
#if HAVE_PTHREADS
#include <pthread.h>
#endif
#include "cmdutils.h"
#include "libavformat/avformat.h"
@@ -45,6 +41,7 @@
#include "libavutil/hwcontext.h"
#include "libavutil/pixfmt.h"
#include "libavutil/rational.h"
#include "libavutil/thread.h"
#include "libavutil/threadmessage.h"
#include "libswresample/swresample.h"
@@ -61,14 +58,10 @@
enum HWAccelID {
HWACCEL_NONE = 0,
HWACCEL_AUTO,
HWACCEL_VDPAU,
HWACCEL_DXVA2,
HWACCEL_VDA,
HWACCEL_GENERIC,
HWACCEL_VIDEOTOOLBOX,
HWACCEL_QSV,
HWACCEL_VAAPI,
HWACCEL_CUVID,
HWACCEL_D3D11VA,
};
typedef struct HWAccel {
@@ -76,7 +69,6 @@ typedef struct HWAccel {
int (*init)(AVCodecContext *s);
enum HWAccelID id;
enum AVPixelFormat pix_fmt;
enum AVHWDeviceType device_type;
} HWAccel;
typedef struct HWDevice {
@@ -161,6 +153,7 @@ typedef struct OptionsContext {
float mux_preload;
float mux_max_delay;
int shortest;
int bitexact;
int video_disable;
int audio_disable;
@@ -369,11 +362,11 @@ typedef struct InputStream {
/* hwaccel options */
enum HWAccelID hwaccel_id;
enum AVHWDeviceType hwaccel_device_type;
char *hwaccel_device;
enum AVPixelFormat hwaccel_output_format;
/* hwaccel context */
enum HWAccelID active_hwaccel_id;
void *hwaccel_ctx;
void (*hwaccel_uninit)(AVCodecContext *s);
int (*hwaccel_get_buffer)(AVCodecContext *s, AVFrame *frame, int flags);
@@ -419,7 +412,7 @@ typedef struct InputFile {
int rate_emu;
int accurate_seek;
#if HAVE_PTHREADS
#if HAVE_THREADS
AVThreadMessageQueue *in_thread_queue;
pthread_t thread; /* thread reading from this file */
int non_blocking; /* reading packets from the thread should not block */
@@ -624,7 +617,6 @@ extern const AVIOInterruptCB int_cb;
extern const OptionDef options[];
extern const HWAccel hwaccels[];
extern int hwaccel_lax_profile_check;
extern AVBufferRef *hw_device_ctx;
#if CONFIG_QSV
extern char *qsv_device;
@@ -662,7 +654,6 @@ int ifilter_parameters_from_frame(InputFilter *ifilter, const AVFrame *frame);
int ffmpeg_parse_options(int argc, char **argv);
int vda_init(AVCodecContext *s);
int videotoolbox_init(AVCodecContext *s);
int qsv_init(AVCodecContext *s);
int cuvid_init(AVCodecContext *s);
fftools/ffmpeg_filter.c
+1
View File
@@ -340,6 +340,7 @@ int init_complex_filtergraph(FilterGraph *fg)
graph = avfilter_graph_alloc();
if (!graph)
return AVERROR(ENOMEM);
graph->nb_threads = 1;
ret = avfilter_graph_parse2(graph, fg->graph_desc, &inputs, &outputs);
if (ret < 0)
fftools/ffmpeg_hw.c
+172 -78
View File
@@ -64,6 +64,31 @@ static HWDevice *hw_device_add(void)
return hw_devices[nb_hw_devices++];
}
static char *hw_device_default_name(enum AVHWDeviceType type)
{
// Make an automatic name of the form "type%d". We arbitrarily
// limit at 1000 anonymous devices of the same type - there is
// probably something else very wrong if you get to this limit.
const char *type_name = av_hwdevice_get_type_name(type);
char *name;
size_t index_pos;
int index, index_limit = 1000;
index_pos = strlen(type_name);
name = av_malloc(index_pos + 4);
if (!name)
return NULL;
for (index = 0; index < index_limit; index++) {
snprintf(name, index_pos + 4, "%s%d", type_name, index);
if (!hw_device_get_by_name(name))
break;
}
if (index >= index_limit) {
av_freep(&name);
return NULL;
}
return name;
}
int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
{
// "type=name:device,key=value,key2=value2"
@@ -111,27 +136,11 @@ int hw_device_init_from_string(const char *arg, HWDevice **dev_out)
p += 1 + k;
} else {
// Give the device an automatic name of the form "type%d".
// We arbitrarily limit at 1000 anonymous devices of the same
// type - there is probably something else very wrong if you
// get to this limit.
size_t index_pos;
int index, index_limit = 1000;
index_pos = strlen(type_name);
name = av_malloc(index_pos + 4);
name = hw_device_default_name(type);
if (!name) {
err = AVERROR(ENOMEM);
goto fail;
}
for (index = 0; index < index_limit; index++) {
snprintf(name, index_pos + 4, "%s%d", type_name, index);
if (!hw_device_get_by_name(name))
break;
}
if (index >= index_limit) {
errmsg = "too many devices";
goto invalid;
}
}
if (!*p) {
@@ -214,6 +223,49 @@ fail:
goto done;
}
static int hw_device_init_from_type(enum AVHWDeviceType type,
const char *device,
HWDevice **dev_out)
{
AVBufferRef *device_ref = NULL;
HWDevice *dev;
char *name;
int err;
name = hw_device_default_name(type);
if (!name) {
err = AVERROR(ENOMEM);
goto fail;
}
err = av_hwdevice_ctx_create(&device_ref, type, device, NULL, 0);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR,
"Device creation failed: %d.\n", err);
goto fail;
}
dev = hw_device_add();
if (!dev) {
err = AVERROR(ENOMEM);
goto fail;
}
dev->name = name;
dev->type = type;
dev->device_ref = device_ref;
if (dev_out)
*dev_out = dev;
return 0;
fail:
av_freep(&name);
av_buffer_unref(&device_ref);
return err;
}
void hw_device_free_all(void)
{
int i;
@@ -226,80 +278,130 @@ void hw_device_free_all(void)
nb_hw_devices = 0;
}
static enum AVHWDeviceType hw_device_match_type_by_hwaccel(enum HWAccelID hwaccel_id)
static HWDevice *hw_device_match_by_codec(const AVCodec *codec)
{
const AVCodecHWConfig *config;
HWDevice *dev;
int i;
if (hwaccel_id == HWACCEL_NONE)
return AV_HWDEVICE_TYPE_NONE;
for (i = 0; hwaccels[i].name; i++) {
if (hwaccels[i].id == hwaccel_id)
return hwaccels[i].device_type;
for (i = 0;; i++) {
config = avcodec_get_hw_config(codec, i);
if (!config)
return NULL;
if (!(config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX))
continue;
dev = hw_device_get_by_type(config->device_type);
if (dev)
return dev;
}
return AV_HWDEVICE_TYPE_NONE;
}
static enum AVHWDeviceType hw_device_match_type_in_name(const char *codec_name)
{
const char *type_name;
enum AVHWDeviceType type;
for (type = av_hwdevice_iterate_types(AV_HWDEVICE_TYPE_NONE);
type != AV_HWDEVICE_TYPE_NONE;
type = av_hwdevice_iterate_types(type)) {
type_name = av_hwdevice_get_type_name(type);
if (strstr(codec_name, type_name))
return type;
}
return AV_HWDEVICE_TYPE_NONE;
}
int hw_device_setup_for_decode(InputStream *ist)
{
const AVCodecHWConfig *config;
enum AVHWDeviceType type;
HWDevice *dev;
int err;
HWDevice *dev = NULL;
int err, auto_device = 0;
if (ist->hwaccel_device) {
dev = hw_device_get_by_name(ist->hwaccel_device);
if (!dev) {
char *tmp;
type = hw_device_match_type_by_hwaccel(ist->hwaccel_id);
if (type == AV_HWDEVICE_TYPE_NONE) {
// No match - this isn't necessarily invalid, though,
// because an explicit device might not be needed or
// the hwaccel setup could be handled elsewhere.
if (ist->hwaccel_id == HWACCEL_AUTO) {
auto_device = 1;
} else if (ist->hwaccel_id == HWACCEL_GENERIC) {
type = ist->hwaccel_device_type;
err = hw_device_init_from_type(type, ist->hwaccel_device,
&dev);
} else {
// This will be dealt with by API-specific initialisation
// (using hwaccel_device), so nothing further needed here.
return 0;
}
tmp = av_asprintf("%s:%s", av_hwdevice_get_type_name(type),
ist->hwaccel_device);
if (!tmp)
return AVERROR(ENOMEM);
err = hw_device_init_from_string(tmp, &dev);
av_free(tmp);
if (err < 0)
return err;
} else {
if (ist->hwaccel_id == HWACCEL_AUTO) {
ist->hwaccel_device_type = dev->type;
} else if (ist->hwaccel_device_type != dev->type) {
av_log(ist->dec_ctx, AV_LOG_ERROR, "Invalid hwaccel device "
"specified for decoder: device %s of type %s is not "
"usable with hwaccel %s.\n", dev->name,
av_hwdevice_get_type_name(dev->type),
av_hwdevice_get_type_name(ist->hwaccel_device_type));
return AVERROR(EINVAL);
}
}
} else {
if (ist->hwaccel_id != HWACCEL_NONE)
type = hw_device_match_type_by_hwaccel(ist->hwaccel_id);
else
type = hw_device_match_type_in_name(ist->dec->name);
if (type != AV_HWDEVICE_TYPE_NONE) {
if (ist->hwaccel_id == HWACCEL_AUTO) {
auto_device = 1;
} else if (ist->hwaccel_id == HWACCEL_GENERIC) {
type = ist->hwaccel_device_type;
dev = hw_device_get_by_type(type);
if (!dev) {
hw_device_init_from_string(av_hwdevice_get_type_name(type),
&dev);
}
if (!dev)
err = hw_device_init_from_type(type, NULL, &dev);
} else {
// No device required.
dev = hw_device_match_by_codec(ist->dec);
if (!dev) {
// No device for this codec, but not using generic hwaccel
// and therefore may well not need one - ignore.
return 0;
}
}
}
if (auto_device) {
int i;
if (!avcodec_get_hw_config(ist->dec, 0)) {
// Decoder does not support any hardware devices.
return 0;
}
for (i = 0; !dev; i++) {
config = avcodec_get_hw_config(ist->dec, i);
if (!config)
break;
type = config->device_type;
dev = hw_device_get_by_type(type);
if (dev) {
av_log(ist->dec_ctx, AV_LOG_INFO, "Using auto "
"hwaccel type %s with existing device %s.\n",
av_hwdevice_get_type_name(type), dev->name);
}
}
for (i = 0; !dev; i++) {
config = avcodec_get_hw_config(ist->dec, i);
if (!config)
break;
type = config->device_type;
// Try to make a new device of this type.
err = hw_device_init_from_type(type, ist->hwaccel_device,
&dev);
if (err < 0) {
// Can't make a device of this type.
continue;
}
if (ist->hwaccel_device) {
av_log(ist->dec_ctx, AV_LOG_INFO, "Using auto "
"hwaccel type %s with new device created "
"from %s.\n", av_hwdevice_get_type_name(type),
ist->hwaccel_device);
} else {
av_log(ist->dec_ctx, AV_LOG_INFO, "Using auto "
"hwaccel type %s with new default device.\n",
av_hwdevice_get_type_name(type));
}
}
if (dev) {
ist->hwaccel_device_type = type;
} else {
av_log(ist->dec_ctx, AV_LOG_INFO, "Auto hwaccel "
"disabled: no device found.\n");
ist->hwaccel_id = HWACCEL_NONE;
return 0;
}
}
if (!dev) {
av_log(ist->dec_ctx, AV_LOG_WARNING, "No device available "
"for decoder (device type %s for codec %s).\n",
av_log(ist->dec_ctx, AV_LOG_ERROR, "No device available "
"for decoder: device type %s needed for codec %s.\n",
av_hwdevice_get_type_name(type), ist->dec->name);
return 0;
return err;
}
ist->dec_ctx->hw_device_ctx = av_buffer_ref(dev->device_ref);
@@ -311,24 +413,16 @@ int hw_device_setup_for_decode(InputStream *ist)
int hw_device_setup_for_encode(OutputStream *ost)
{
enum AVHWDeviceType type;
HWDevice *dev;
type = hw_device_match_type_in_name(ost->enc->name);
if (type != AV_HWDEVICE_TYPE_NONE) {
dev = hw_device_get_by_type(type);
if (!dev) {
av_log(ost->enc_ctx, AV_LOG_WARNING, "No device available "
"for encoder (device type %s for codec %s).\n",
av_hwdevice_get_type_name(type), ost->enc->name);
return 0;
}
dev = hw_device_match_by_codec(ost->enc);
if (dev) {
ost->enc_ctx->hw_device_ctx = av_buffer_ref(dev->device_ref);
if (!ost->enc_ctx->hw_device_ctx)
return AVERROR(ENOMEM);
return 0;
} else {
// No device required.
// No device required, or no device available.
return 0;
}
}
fftools/ffmpeg_opt.c
+75 -152
View File
@@ -66,41 +66,17 @@
}
const HWAccel hwaccels[] = {
#if HAVE_VDPAU_X11
{ "vdpau", hwaccel_decode_init, HWACCEL_VDPAU, AV_PIX_FMT_VDPAU,
AV_HWDEVICE_TYPE_VDPAU },
#endif
#if CONFIG_D3D11VA
{ "d3d11va", hwaccel_decode_init, HWACCEL_D3D11VA, AV_PIX_FMT_D3D11,
AV_HWDEVICE_TYPE_D3D11VA },
#endif
#if CONFIG_DXVA2
{ "dxva2", hwaccel_decode_init, HWACCEL_DXVA2, AV_PIX_FMT_DXVA2_VLD,
AV_HWDEVICE_TYPE_DXVA2 },
#endif
#if CONFIG_VDA
{ "vda", videotoolbox_init, HWACCEL_VDA, AV_PIX_FMT_VDA,
AV_HWDEVICE_TYPE_NONE },
#endif
#if CONFIG_VIDEOTOOLBOX
{ "videotoolbox", videotoolbox_init, HWACCEL_VIDEOTOOLBOX, AV_PIX_FMT_VIDEOTOOLBOX,
AV_HWDEVICE_TYPE_NONE },
{ "videotoolbox", videotoolbox_init, HWACCEL_VIDEOTOOLBOX, AV_PIX_FMT_VIDEOTOOLBOX },
#endif
#if CONFIG_LIBMFX
{ "qsv", qsv_init, HWACCEL_QSV, AV_PIX_FMT_QSV,
AV_HWDEVICE_TYPE_NONE },
#endif
#if CONFIG_VAAPI
{ "vaapi", hwaccel_decode_init, HWACCEL_VAAPI, AV_PIX_FMT_VAAPI,
AV_HWDEVICE_TYPE_VAAPI },
{ "qsv", qsv_init, HWACCEL_QSV, AV_PIX_FMT_QSV },
#endif
#if CONFIG_CUVID
{ "cuvid", cuvid_init, HWACCEL_CUVID, AV_PIX_FMT_CUDA,
AV_HWDEVICE_TYPE_NONE },
{ "cuvid", cuvid_init, HWACCEL_CUVID, AV_PIX_FMT_CUDA },
#endif
{ 0 },
};
int hwaccel_lax_profile_check = 0;
AVBufferRef *hw_device_ctx;
HWDevice *filter_hw_device;
@@ -141,7 +117,6 @@ static int file_overwrite = 0;
static int no_file_overwrite = 0;
static int do_psnr = 0;
static int input_sync;
static int override_ffserver = 0;
static int input_stream_potentially_available = 0;
static int ignore_unknown_streams = 0;
static int copy_unknown_streams = 0;
@@ -195,12 +170,15 @@ static void init_options(OptionsContext *o)
static int show_hwaccels(void *optctx, const char *opt, const char *arg)
{
enum AVHWDeviceType type = AV_HWDEVICE_TYPE_NONE;
int i;
printf("Hardware acceleration methods:\n");
for (i = 0; hwaccels[i].name; i++) {
while ((type = av_hwdevice_iterate_types(type)) !=
AV_HWDEVICE_TYPE_NONE)
printf("%s\n", av_hwdevice_get_type_name(type));
for (i = 0; hwaccels[i].name; i++)
printf("%s\n", hwaccels[i].name);
}
printf("\n");
return 0;
}
@@ -723,7 +701,8 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
AVStream *st = ic->streams[i];
AVCodecParameters *par = st->codecpar;
InputStream *ist = av_mallocz(sizeof(*ist));
char *framerate = NULL, *hwaccel = NULL, *hwaccel_device = NULL;
char *framerate = NULL, *hwaccel_device = NULL;
const char *hwaccel = NULL;
char *hwaccel_output_format = NULL;
char *codec_tag = NULL;
char *next;
@@ -787,20 +766,20 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
exit_program(1);
}
if (o->bitexact)
ist->dec_ctx->flags |= AV_CODEC_FLAG_BITEXACT;
switch (par->codec_type) {
case AVMEDIA_TYPE_VIDEO:
if(!ist->dec)
ist->dec = avcodec_find_decoder(par->codec_id);
#if FF_API_LOWRES
if (av_codec_get_lowres(st->codec)) {
av_codec_set_lowres(ist->dec_ctx, av_codec_get_lowres(st->codec));
if (st->codec->lowres) {
ist->dec_ctx->lowres = st->codec->lowres;
ist->dec_ctx->width = st->codec->width;
ist->dec_ctx->height = st->codec->height;
ist->dec_ctx->coded_width = st->codec->coded_width;
ist->dec_ctx->coded_height = st->codec->coded_height;
#if FF_API_EMU_EDGE
ist->dec_ctx->flags |= CODEC_FLAG_EMU_EDGE;
#endif
}
#endif
@@ -820,11 +799,16 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
MATCH_PER_STREAM_OPT(hwaccels, str, hwaccel, ic, st);
if (hwaccel) {
// The NVDEC hwaccels use a CUDA device, so remap the name here.
if (!strcmp(hwaccel, "nvdec"))
hwaccel = "cuda";
if (!strcmp(hwaccel, "none"))
ist->hwaccel_id = HWACCEL_NONE;
else if (!strcmp(hwaccel, "auto"))
ist->hwaccel_id = HWACCEL_AUTO;
else {
enum AVHWDeviceType type;
int i;
for (i = 0; hwaccels[i].name; i++) {
if (!strcmp(hwaccels[i].name, hwaccel)) {
@@ -833,10 +817,23 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
}
}
if (!ist->hwaccel_id) {
type = av_hwdevice_find_type_by_name(hwaccel);
if (type != AV_HWDEVICE_TYPE_NONE) {
ist->hwaccel_id = HWACCEL_GENERIC;
ist->hwaccel_device_type = type;
}
}
if (!ist->hwaccel_id) {
av_log(NULL, AV_LOG_FATAL, "Unrecognized hwaccel: %s.\n",
hwaccel);
av_log(NULL, AV_LOG_FATAL, "Supported hwaccels: ");
type = AV_HWDEVICE_TYPE_NONE;
while ((type = av_hwdevice_iterate_types(type)) !=
AV_HWDEVICE_TYPE_NONE)
av_log(NULL, AV_LOG_FATAL, "%s ",
av_hwdevice_get_type_name(type));
for (i = 0; hwaccels[i].name; i++)
av_log(NULL, AV_LOG_FATAL, "%s ", hwaccels[i].name);
av_log(NULL, AV_LOG_FATAL, "\n");
@@ -977,6 +974,21 @@ static int open_input_file(OptionsContext *o, const char *filename)
char * data_codec_name = NULL;
int scan_all_pmts_set = 0;
if (o->stop_time != INT64_MAX && o->recording_time != INT64_MAX) {
o->stop_time = INT64_MAX;
av_log(NULL, AV_LOG_WARNING, "-t and -to cannot be used together; using -t.\n");
}
if (o->stop_time != INT64_MAX && o->recording_time == INT64_MAX) {
int64_t start_time = o->start_time == AV_NOPTS_VALUE ? 0 : o->start_time;
if (o->stop_time <= start_time) {
av_log(NULL, AV_LOG_ERROR, "-to value smaller than -ss; aborting.\n");
exit_program(1);
} else {
o->recording_time = o->stop_time - start_time;
}
}
if (o->format) {
if (!(file_iformat = av_find_input_format(o->format))) {
av_log(NULL, AV_LOG_FATAL, "Unknown input format: '%s'\n", o->format);
@@ -996,7 +1008,6 @@ static int open_input_file(OptionsContext *o, const char *filename)
print_error(filename, AVERROR(ENOMEM));
exit_program(1);
}
ic->flags |= AVFMT_FLAG_KEEP_SIDE_DATA;
if (o->nb_audio_sample_rate) {
av_dict_set_int(&o->g->format_opts, "sample_rate", o->audio_sample_rate[o->nb_audio_sample_rate - 1].u.i, 0);
}
@@ -1031,25 +1042,23 @@ static int open_input_file(OptionsContext *o, const char *filename)
MATCH_PER_TYPE_OPT(codec_names, str, subtitle_codec_name, ic, "s");
MATCH_PER_TYPE_OPT(codec_names, str, data_codec_name, ic, "d");
ic->video_codec_id = video_codec_name ?
find_codec_or_die(video_codec_name , AVMEDIA_TYPE_VIDEO , 0)->id : AV_CODEC_ID_NONE;
ic->audio_codec_id = audio_codec_name ?
find_codec_or_die(audio_codec_name , AVMEDIA_TYPE_AUDIO , 0)->id : AV_CODEC_ID_NONE;
ic->subtitle_codec_id= subtitle_codec_name ?
find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0)->id : AV_CODEC_ID_NONE;
ic->data_codec_id = data_codec_name ?
find_codec_or_die(data_codec_name, AVMEDIA_TYPE_DATA, 0)->id : AV_CODEC_ID_NONE;
if (video_codec_name)
av_format_set_video_codec (ic, find_codec_or_die(video_codec_name , AVMEDIA_TYPE_VIDEO , 0));
ic->video_codec = find_codec_or_die(video_codec_name , AVMEDIA_TYPE_VIDEO , 0);
if (audio_codec_name)
av_format_set_audio_codec (ic, find_codec_or_die(audio_codec_name , AVMEDIA_TYPE_AUDIO , 0));
ic->audio_codec = find_codec_or_die(audio_codec_name , AVMEDIA_TYPE_AUDIO , 0);
if (subtitle_codec_name)
av_format_set_subtitle_codec(ic, find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0));
ic->subtitle_codec = find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0);
if (data_codec_name)
av_format_set_data_codec(ic, find_codec_or_die(data_codec_name, AVMEDIA_TYPE_DATA, 0));
ic->data_codec = find_codec_or_die(data_codec_name , AVMEDIA_TYPE_DATA , 0);
ic->video_codec_id = video_codec_name ? ic->video_codec->id : AV_CODEC_ID_NONE;
ic->audio_codec_id = audio_codec_name ? ic->audio_codec->id : AV_CODEC_ID_NONE;
ic->subtitle_codec_id = subtitle_codec_name ? ic->subtitle_codec->id : AV_CODEC_ID_NONE;
ic->data_codec_id = data_codec_name ? ic->data_codec->id : AV_CODEC_ID_NONE;
ic->flags |= AVFMT_FLAG_NONBLOCK;
if (o->bitexact)
ic->flags |= AVFMT_FLAG_BITEXACT;
ic->interrupt_callback = int_cb;
if (!av_dict_get(o->g->format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE)) {
@@ -1151,7 +1160,7 @@ static int open_input_file(OptionsContext *o, const char *filename)
f->loop = o->loop;
f->duration = 0;
f->time_base = (AVRational){ 1, 1 };
#if HAVE_PTHREADS
#if HAVE_THREADS
f->thread_queue_size = o->thread_queue_size > 0 ? o->thread_queue_size : 8;
#endif
@@ -1371,6 +1380,10 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
ost->encoder_opts = filter_codec_opts(o->g->codec_opts, AV_CODEC_ID_NONE, oc, st, NULL);
}
if (o->bitexact)
ost->enc_ctx->flags |= AV_CODEC_FLAG_BITEXACT;
MATCH_PER_STREAM_OPT(time_bases, str, time_base, oc, st);
if (time_base) {
AVRational q;
@@ -1663,7 +1676,7 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
av_log(NULL, AV_LOG_FATAL, "Could not allocate memory for intra matrix.\n");
exit_program(1);
}
av_codec_set_chroma_intra_matrix(video_enc, p);
video_enc->chroma_intra_matrix = p;
parse_matrix_coeffs(p, chroma_intra_matrix);
}
MATCH_PER_STREAM_OPT(inter_matrices, str, inter_matrix, oc, st);
@@ -1983,57 +1996,6 @@ static int copy_chapters(InputFile *ifile, OutputFile *ofile, int copy_metadata)
return 0;
}
static int read_ffserver_streams(OptionsContext *o, AVFormatContext *s, const char *filename)
{
int i, err;
AVFormatContext *ic = avformat_alloc_context();
ic->flags |= AVFMT_FLAG_KEEP_SIDE_DATA;
ic->interrupt_callback = int_cb;
err = avformat_open_input(&ic, filename, NULL, NULL);
if (err < 0)
return err;
/* copy stream format */
for(i=0;i<ic->nb_streams;i++) {
AVStream *st;
OutputStream *ost;
AVCodec *codec;
const char *enc_config;
codec = avcodec_find_encoder(ic->streams[i]->codecpar->codec_id);
if (!codec) {
av_log(s, AV_LOG_ERROR, "no encoder found for codec id %i\n", ic->streams[i]->codecpar->codec_id);
return AVERROR(EINVAL);
}
if (codec->type == AVMEDIA_TYPE_AUDIO)
opt_audio_codec(o, "c:a", codec->name);
else if (codec->type == AVMEDIA_TYPE_VIDEO)
opt_video_codec(o, "c:v", codec->name);
ost = new_output_stream(o, s, codec->type, -1);
st = ost->st;
avcodec_get_context_defaults3(st->codec, codec);
enc_config = av_stream_get_recommended_encoder_configuration(ic->streams[i]);
if (enc_config) {
AVDictionary *opts = NULL;
av_dict_parse_string(&opts, enc_config, "=", ",", 0);
av_opt_set_dict2(st->codec, &opts, AV_OPT_SEARCH_CHILDREN);
av_dict_free(&opts);
}
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && !ost->stream_copy)
choose_sample_fmt(st, codec);
else if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO && !ost->stream_copy)
choose_pixel_fmt(st, st->codec, codec, st->codecpar->format);
avcodec_copy_context(ost->enc_ctx, st->codec);
if (enc_config)
av_dict_parse_string(&ost->encoder_opts, enc_config, "=", ",", 0);
}
avformat_close_input(&ic);
return err;
}
static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
AVFormatContext *oc)
{
@@ -2092,7 +2054,6 @@ static int open_output_file(OptionsContext *o, const char *filename)
{
AVFormatContext *oc;
int i, j, err;
AVOutputFormat *file_oformat;
OutputFile *of;
OutputStream *ost;
InputStream *ist;
@@ -2141,7 +2102,6 @@ static int open_output_file(OptionsContext *o, const char *filename)
if (o->recording_time != INT64_MAX)
oc->duration = o->recording_time;
file_oformat= oc->oformat;
oc->interrupt_callback = int_cb;
e = av_dict_get(o->g->format_opts, "fflags", NULL, 0);
@@ -2149,6 +2109,10 @@ static int open_output_file(OptionsContext *o, const char *filename)
const AVOption *o = av_opt_find(oc, "fflags", NULL, 0, 0);
av_opt_eval_flags(oc, o, e->value, &format_flags);
}
if (o->bitexact) {
format_flags |= AVFMT_FLAG_BITEXACT;
oc->flags |= AVFMT_FLAG_BITEXACT;
}
/* create streams for all unlabeled output pads */
for (i = 0; i < nb_filtergraphs; i++) {
@@ -2168,47 +2132,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
}
}
/* ffserver seeking with date=... needs a date reference */
if (!strcmp(file_oformat->name, "ffm") &&
!(format_flags & AVFMT_FLAG_BITEXACT) &&
av_strstart(filename, "http:", NULL)) {
int err = parse_option(o, "metadata", "creation_time=now", options);
if (err < 0) {
print_error(filename, err);
exit_program(1);
}
}
if (!strcmp(file_oformat->name, "ffm") && !override_ffserver &&
av_strstart(filename, "http:", NULL)) {
int j;
/* special case for files sent to ffserver: we get the stream
parameters from ffserver */
int err = read_ffserver_streams(o, oc, filename);
if (err < 0) {
print_error(filename, err);
exit_program(1);
}
for(j = nb_output_streams - oc->nb_streams; j < nb_output_streams; j++) {
ost = output_streams[j];
for (i = 0; i < nb_input_streams; i++) {
ist = input_streams[i];
if(ist->st->codecpar->codec_type == ost->st->codecpar->codec_type){
ost->sync_ist= ist;
ost->source_index= i;
if(ost->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) ost->avfilter = av_strdup("anull");
if(ost->st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) ost->avfilter = av_strdup("null");
ist->discard = 0;
ist->st->discard = ist->user_set_discard;
break;
}
}
if(!ost->sync_ist){
av_log(NULL, AV_LOG_FATAL, "Missing %s stream which is required by this ffm\n", av_get_media_type_string(ost->st->codecpar->codec_type));
exit_program(1);
}
}
} else if (!o->nb_stream_maps) {
if (!o->nb_stream_maps) {
char *subtitle_codec_name = NULL;
/* pick the "best" stream of each type */
@@ -3404,7 +3328,7 @@ const OptionDef options[] = {
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(recording_time) },
"record or transcode \"duration\" seconds of audio/video",
"duration" },
{ "to", HAS_ARG | OPT_TIME | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(stop_time) },
{ "to", HAS_ARG | OPT_TIME | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(stop_time) },
"record or transcode stop time", "time_stop" },
{ "fs", HAS_ARG | OPT_INT64 | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(limit_filesize) },
"set the limit file size in bytes", "limit_size" },
@@ -3472,6 +3396,9 @@ const OptionDef options[] = {
{ "shortest", OPT_BOOL | OPT_EXPERT | OPT_OFFSET |
OPT_OUTPUT, { .off = OFFSET(shortest) },
"finish encoding within shortest input" },
{ "bitexact", OPT_BOOL | OPT_EXPERT | OPT_OFFSET |
OPT_OUTPUT | OPT_INPUT, { .off = OFFSET(bitexact) },
"bitexact mode" },
{ "apad", OPT_STRING | HAS_ARG | OPT_SPEC |
OPT_OUTPUT, { .off = OFFSET(apad) },
"audio pad", "" },
@@ -3632,7 +3559,7 @@ const OptionDef options[] = {
{ "hwaccel_output_format", OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
OPT_SPEC | OPT_INPUT, { .off = OFFSET(hwaccel_output_formats) },
"select output format used with HW accelerated decoding", "format" },
#if CONFIG_VDA || CONFIG_VIDEOTOOLBOX
#if CONFIG_VIDEOTOOLBOX
{ "videotoolbox_pixfmt", HAS_ARG | OPT_STRING | OPT_EXPERT, { &videotoolbox_pixfmt}, "" },
#endif
{ "hwaccels", OPT_EXIT, { .func_arg = show_hwaccels },
@@ -3640,8 +3567,6 @@ const OptionDef options[] = {
{ "autorotate", HAS_ARG | OPT_BOOL | OPT_SPEC |
OPT_EXPERT | OPT_INPUT, { .off = OFFSET(autorotate) },
"automatically insert correct rotate filters" },
{ "hwaccel_lax_profile_check", OPT_BOOL | OPT_EXPERT, { &hwaccel_lax_profile_check},
"attempt to decode anyway if HW accelerated decoder's supported profiles do not exactly match the stream" },
/* audio options */
{ "aframes", OPT_AUDIO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_audio_frames },
@@ -3699,8 +3624,6 @@ const OptionDef options[] = {
"set the maximum demux-decode delay", "seconds" },
{ "muxpreload", OPT_FLOAT | HAS_ARG | OPT_EXPERT | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(mux_preload) },
"set the initial demux-decode delay", "seconds" },
{ "override_ffserver", OPT_BOOL | OPT_EXPERT | OPT_OUTPUT, { &override_ffserver },
"override the options from ffserver", "" },
{ "sdp_file", HAS_ARG | OPT_EXPERT | OPT_OUTPUT, { .func_arg = opt_sdp_file },
"specify a file in which to print sdp information", "file" },
fftools/ffmpeg_videotoolbox.c
+2 -39
View File
@@ -23,12 +23,7 @@
#endif
#include "libavcodec/avcodec.h"
#if CONFIG_VDA
# include "libavcodec/vda.h"
#endif
#if CONFIG_VIDEOTOOLBOX
#include "libavcodec/videotoolbox.h"
#endif
#include "libavutil/imgutils.h"
#include "ffmpeg.h"
@@ -114,15 +109,7 @@ static void videotoolbox_uninit(AVCodecContext *s)
av_frame_free(&vt->tmp_frame);
if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
#if CONFIG_VIDEOTOOLBOX
av_videotoolbox_default_free(s);
#endif
} else {
#if CONFIG_VDA
av_vda_default_free(s);
#endif
}
av_freep(&ist->hwaccel_ctx);
}
@@ -147,8 +134,7 @@ int videotoolbox_init(AVCodecContext *s)
goto fail;
}
if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
#if CONFIG_VIDEOTOOLBOX
// TODO: reindent
if (!videotoolbox_pixfmt) {
ret = av_videotoolbox_default_init(s);
} else {
@@ -166,31 +152,8 @@ int videotoolbox_init(AVCodecContext *s)
ret = av_videotoolbox_default_init2(s, vtctx);
CFRelease(pixfmt_str);
}
#endif
} else {
#if CONFIG_VDA
if (!videotoolbox_pixfmt) {
ret = av_vda_default_init(s);
} else {
AVVDAContext *vdactx = av_vda_alloc_context();
CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
videotoolbox_pixfmt,
kCFStringEncodingUTF8);
#if HAVE_UTGETOSTYPEFROMSTRING
vdactx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
#else
av_log(s, loglevel, "UTGetOSTypeFromString() is not available "
"on this platform, %s pixel format can not be honored from "
"the command line\n", videotoolbox_pixfmt);
#endif
ret = av_vda_default_init2(s, vdactx);
CFRelease(pixfmt_str);
}
#endif
}
if (ret < 0) {
av_log(NULL, loglevel,
"Error creating %s decoder.\n", ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX ? "Videotoolbox" : "VDA");
av_log(NULL, loglevel, "Error creating Videotoolbox decoder.\n");
goto fail;
}
fftools/ffplay.c
+55 -77
View File
@@ -361,6 +361,8 @@ static AVPacket flush_pkt;
static SDL_Window *window;
static SDL_Renderer *renderer;
static SDL_RendererInfo renderer_info = {0};
static SDL_AudioDeviceID audio_dev;
static const struct TextureFormatEntry {
enum AVPixelFormat format;
@@ -607,7 +609,7 @@ static int decoder_decode_frame(Decoder *d, AVFrame *frame, AVSubtitle *sub) {
if (ret >= 0) {
AVRational tb = (AVRational){1, frame->sample_rate};
if (frame->pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(frame->pts, av_codec_get_pkt_timebase(d->avctx), tb);
frame->pts = av_rescale_q(frame->pts, d->avctx->pkt_timebase, tb);
else if (d->next_pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(d->next_pts, d->next_pts_tb, tb);
if (frame->pts != AV_NOPTS_VALUE) {
@@ -1191,7 +1193,7 @@ static void stream_component_close(VideoState *is, int stream_index)
switch (codecpar->codec_type) {
case AVMEDIA_TYPE_AUDIO:
decoder_abort(&is->auddec, &is->sampq);
SDL_CloseAudio();
SDL_CloseAudioDevice(audio_dev);
decoder_destroy(&is->auddec);
swr_free(&is->swr_ctx);
av_freep(&is->audio_buf1);
@@ -1282,7 +1284,6 @@ static void do_exit(VideoState *is)
SDL_DestroyRenderer(renderer);
if (window)
SDL_DestroyWindow(window);
av_lockmgr_register(NULL);
uninit_opts();
#if CONFIG_AVFILTER
av_freep(&vfilters_list);
@@ -1320,38 +1321,15 @@ static int video_open(VideoState *is)
h = default_height;
}
if (!window) {
int flags = SDL_WINDOW_SHOWN;
if (!window_title)
window_title = input_filename;
if (is_full_screen)
flags |= SDL_WINDOW_FULLSCREEN_DESKTOP;
if (borderless)
flags |= SDL_WINDOW_BORDERLESS;
else
flags |= SDL_WINDOW_RESIZABLE;
window = SDL_CreateWindow(window_title, SDL_WINDOWPOS_UNDEFINED, SDL_WINDOWPOS_UNDEFINED, w, h, flags);
SDL_SetHint(SDL_HINT_RENDER_SCALE_QUALITY, "linear");
if (window) {
SDL_RendererInfo info;
renderer = SDL_CreateRenderer(window, -1, SDL_RENDERER_ACCELERATED | SDL_RENDERER_PRESENTVSYNC);
if (!renderer) {
av_log(NULL, AV_LOG_WARNING, "Failed to initialize a hardware accelerated renderer: %s\n", SDL_GetError());
renderer = SDL_CreateRenderer(window, -1, 0);
}
if (renderer) {
if (!SDL_GetRendererInfo(renderer, &info))
av_log(NULL, AV_LOG_VERBOSE, "Initialized %s renderer.\n", info.name);
}
}
} else {
SDL_SetWindowSize(window, w, h);
}
SDL_SetWindowTitle(window, window_title);
if (!window || !renderer) {
av_log(NULL, AV_LOG_FATAL, "SDL: could not set video mode - exiting\n");
do_exit(is);
}
SDL_SetWindowSize(window, w, h);
SDL_SetWindowPosition(window, SDL_WINDOWPOS_CENTERED, SDL_WINDOWPOS_CENTERED);
if (is_full_screen)
SDL_SetWindowFullscreen(window, SDL_WINDOW_FULLSCREEN_DESKTOP);
SDL_ShowWindow(window);
is->width = w;
is->height = h;
@@ -1362,7 +1340,7 @@ static int video_open(VideoState *is)
/* display the current picture, if any */
static void video_display(VideoState *is)
{
if (!window)
if (!is->width)
video_open(is);
SDL_SetRenderDrawColor(renderer, 0, 0, 0, 255);
@@ -1850,10 +1828,18 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
AVCodecParameters *codecpar = is->video_st->codecpar;
AVRational fr = av_guess_frame_rate(is->ic, is->video_st, NULL);
AVDictionaryEntry *e = NULL;
int i;
int nb_pix_fmts = 0;
int i, j;
for (i = 0; i < FF_ARRAY_ELEMS(pix_fmts); i++)
pix_fmts[i] = sdl_texture_format_map[i].format;
for (i = 0; i < renderer_info.num_texture_formats; i++) {
for (j = 0; j < FF_ARRAY_ELEMS(sdl_texture_format_map) - 1; j++) {
if (renderer_info.texture_formats[i] == sdl_texture_format_map[j].texture_fmt) {
pix_fmts[nb_pix_fmts++] = sdl_texture_format_map[j].format;
break;
}
}
}
pix_fmts[nb_pix_fmts] = AV_PIX_FMT_NONE;
while ((e = av_dict_get(sws_dict, "", e, AV_DICT_IGNORE_SUFFIX))) {
if (!strcmp(e->key, "sws_flags")) {
@@ -2465,7 +2451,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
else {
memset(stream, 0, len1);
if (!is->muted && is->audio_buf)
SDL_MixAudio(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1, is->audio_volume);
SDL_MixAudioFormat(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, AUDIO_S16SYS, len1, is->audio_volume);
}
len -= len1;
stream += len1;
@@ -2510,7 +2496,7 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
wanted_spec.samples = FFMAX(SDL_AUDIO_MIN_BUFFER_SIZE, 2 << av_log2(wanted_spec.freq / SDL_AUDIO_MAX_CALLBACKS_PER_SEC));
wanted_spec.callback = sdl_audio_callback;
wanted_spec.userdata = opaque;
while (SDL_OpenAudio(&wanted_spec, &spec) < 0) {
while (!(audio_dev = SDL_OpenAudioDevice(NULL, 0, &wanted_spec, &spec, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE | SDL_AUDIO_ALLOW_CHANNELS_CHANGE))) {
av_log(NULL, AV_LOG_WARNING, "SDL_OpenAudio (%d channels, %d Hz): %s\n",
wanted_spec.channels, wanted_spec.freq, SDL_GetError());
wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)];
@@ -2576,7 +2562,7 @@ static int stream_component_open(VideoState *is, int stream_index)
ret = avcodec_parameters_to_context(avctx, ic->streams[stream_index]->codecpar);
if (ret < 0)
goto fail;
av_codec_set_pkt_timebase(avctx, ic->streams[stream_index]->time_base);
avctx->pkt_timebase = ic->streams[stream_index]->time_base;
codec = avcodec_find_decoder(avctx->codec_id);
@@ -2597,22 +2583,15 @@ static int stream_component_open(VideoState *is, int stream_index)
}
avctx->codec_id = codec->id;
if(stream_lowres > av_codec_get_max_lowres(codec)){
if (stream_lowres > codec->max_lowres) {
av_log(avctx, AV_LOG_WARNING, "The maximum value for lowres supported by the decoder is %d\n",
av_codec_get_max_lowres(codec));
stream_lowres = av_codec_get_max_lowres(codec);
codec->max_lowres);
stream_lowres = codec->max_lowres;
}
av_codec_set_lowres(avctx, stream_lowres);
avctx->lowres = stream_lowres;
#if FF_API_EMU_EDGE
if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
#endif
if (fast)
avctx->flags2 |= AV_CODEC_FLAG2_FAST;
#if FF_API_EMU_EDGE
if(codec->capabilities & AV_CODEC_CAP_DR1)
avctx->flags |= CODEC_FLAG_EMU_EDGE;
#endif
opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index], codec);
if (!av_dict_get(opts, "threads", NULL, 0))
@@ -2680,7 +2659,7 @@ static int stream_component_open(VideoState *is, int stream_index)
}
if ((ret = decoder_start(&is->auddec, audio_thread, is)) < 0)
goto out;
SDL_PauseAudio(0);
SDL_PauseAudioDevice(audio_dev, 0);
break;
case AVMEDIA_TYPE_VIDEO:
is->video_stream = stream_index;
@@ -3661,27 +3640,6 @@ void show_help_default(const char *opt, const char *arg)
);
}
static int lockmgr(void **mtx, enum AVLockOp op)
{
switch(op) {
case AV_LOCK_CREATE:
*mtx = SDL_CreateMutex();
if(!*mtx) {
av_log(NULL, AV_LOG_FATAL, "SDL_CreateMutex(): %s\n", SDL_GetError());
return 1;
}
return 0;
case AV_LOCK_OBTAIN:
return !!SDL_LockMutex(*mtx);
case AV_LOCK_RELEASE:
return !!SDL_UnlockMutex(*mtx);
case AV_LOCK_DESTROY:
SDL_DestroyMutex(*mtx);
return 0;
}
return 1;
}
/* Called from the main */
int main(int argc, char **argv)
{
@@ -3743,14 +3701,34 @@ int main(int argc, char **argv)
SDL_EventState(SDL_SYSWMEVENT, SDL_IGNORE);
SDL_EventState(SDL_USEREVENT, SDL_IGNORE);
if (av_lockmgr_register(lockmgr)) {
av_log(NULL, AV_LOG_FATAL, "Could not initialize lock manager!\n");
do_exit(NULL);
}
av_init_packet(&flush_pkt);
flush_pkt.data = (uint8_t *)&flush_pkt;
if (!display_disable) {
int flags = SDL_WINDOW_HIDDEN;
if (borderless)
flags |= SDL_WINDOW_BORDERLESS;
else
flags |= SDL_WINDOW_RESIZABLE;
window = SDL_CreateWindow(program_name, SDL_WINDOWPOS_UNDEFINED, SDL_WINDOWPOS_UNDEFINED, default_width, default_height, flags);
SDL_SetHint(SDL_HINT_RENDER_SCALE_QUALITY, "linear");
if (window) {
renderer = SDL_CreateRenderer(window, -1, SDL_RENDERER_ACCELERATED | SDL_RENDERER_PRESENTVSYNC);
if (!renderer) {
av_log(NULL, AV_LOG_WARNING, "Failed to initialize a hardware accelerated renderer: %s\n", SDL_GetError());
renderer = SDL_CreateRenderer(window, -1, 0);
}
if (renderer) {
if (!SDL_GetRendererInfo(renderer, &renderer_info))
av_log(NULL, AV_LOG_VERBOSE, "Initialized %s renderer.\n", renderer_info.name);
}
}
if (!window || !renderer || !renderer_info.num_texture_formats) {
av_log(NULL, AV_LOG_FATAL, "Failed to create window or renderer: %s", SDL_GetError());
do_exit(NULL);
}
}
is = stream_open(input_filename, file_iformat);
if (!is) {
av_log(NULL, AV_LOG_FATAL, "Failed to initialize VideoState!\n");
fftools/ffprobe.c
+2 -4
View File
@@ -2792,7 +2792,7 @@ static int show_format(WriterContext *w, InputFile *ifile)
else print_str_opt("size", "N/A");
if (fmt_ctx->bit_rate > 0) print_val ("bit_rate", fmt_ctx->bit_rate, unit_bit_per_second_str);
else print_str_opt("bit_rate", "N/A");
print_int("probe_score", av_format_get_probe_score(fmt_ctx));
print_int("probe_score", fmt_ctx->probe_score);
if (do_show_format_tags)
ret = show_tags(w, fmt_ctx->metadata, SECTION_ID_FORMAT_TAGS);
@@ -2828,8 +2828,6 @@ static int open_input_file(InputFile *ifile, const char *filename)
exit_program(1);
}
fmt_ctx->flags |= AVFMT_FLAG_KEEP_SIDE_DATA;
if (!av_dict_get(format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE)) {
av_dict_set(&format_opts, "scan_all_pmts", "1", AV_DICT_DONT_OVERWRITE);
scan_all_pmts_set = 1;
@@ -2912,7 +2910,7 @@ static int open_input_file(InputFile *ifile, const char *filename)
av_dict_set(&codec_opts, "threads", "1", 0);
}
av_codec_set_pkt_timebase(ist->dec_ctx, stream->time_base);
ist->dec_ctx->pkt_timebase = stream->time_base;
ist->dec_ctx->framerate = stream->avg_frame_rate;
if (avcodec_open2(ist->dec_ctx, codec, &opts) < 0) {
fftools/ffserver.c
-4026
View File
File diff suppressed because it is too large Load Diff
fftools/ffserver_config.c
-1325
View File
File diff suppressed because it is too large Load Diff
fftools/ffserver_config.h
-155
View File
@@ -1,155 +0,0 @@
/*
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef FFTOOLS_FFSERVER_CONFIG_H
#define FFTOOLS_FFSERVER_CONFIG_H
#define FFM_PACKET_SIZE 4096
#include "libavutil/dict.h"
#include "libavformat/avformat.h"
#include "libavformat/network.h"
#define FFSERVER_MAX_STREAMS 20
/* each generated stream is described here */
enum FFServerStreamType {
STREAM_TYPE_LIVE,
STREAM_TYPE_STATUS,
STREAM_TYPE_REDIRECT,
};
enum FFServerIPAddressAction {
IP_ALLOW = 1,
IP_DENY,
};
typedef struct FFServerIPAddressACL {
struct FFServerIPAddressACL *next;
enum FFServerIPAddressAction action;
/* These are in host order */
struct in_addr first;
struct in_addr last;
} FFServerIPAddressACL;
/**
* This holds the stream parameters for an AVStream, it cannot be a AVStream
* because AVStreams cannot be instanciated without a AVFormatContext, especially
* not outside libavformat.
*
* The fields of this struct have the same semantics as the fields of an AVStream.
*/
typedef struct LayeredAVStream {
int index;
int id;
AVCodecParameters *codecpar;
AVCodecContext *codec;
AVRational time_base;
int pts_wrap_bits;
AVRational sample_aspect_ratio;
char *recommended_encoder_configuration;
} LayeredAVStream;
/* description of each stream of the ffserver.conf file */
typedef struct FFServerStream {
enum FFServerStreamType stream_type;
char filename[1024]; /* stream filename */
struct FFServerStream *feed; /* feed we are using (can be null if coming from file) */
AVDictionary *in_opts; /* input parameters */
AVDictionary *metadata; /* metadata to set on the stream */
AVInputFormat *ifmt; /* if non NULL, force input format */
AVOutputFormat *fmt;
FFServerIPAddressACL *acl;
char dynamic_acl[1024];
int nb_streams;
int prebuffer; /* Number of milliseconds early to start */
int64_t max_time; /* Number of milliseconds to run */
int send_on_key;
LayeredAVStream *streams[FFSERVER_MAX_STREAMS];
int feed_streams[FFSERVER_MAX_STREAMS]; /* index of streams in the feed */
char feed_filename[1024]; /* file name of the feed storage, or
input file name for a stream */
pid_t pid; /* Of ffmpeg process */
time_t pid_start; /* Of ffmpeg process */
char **child_argv;
struct FFServerStream *next;
unsigned bandwidth; /* bandwidth, in kbits/s */
/* RTSP options */
char *rtsp_option;
/* multicast specific */
int is_multicast;
struct in_addr multicast_ip;
int multicast_port; /* first port used for multicast */
int multicast_ttl;
int loop; /* if true, send the stream in loops (only meaningful if file) */
char single_frame; /* only single frame */
/* feed specific */
int feed_opened; /* true if someone is writing to the feed */
int is_feed; /* true if it is a feed */
int readonly; /* True if writing is prohibited to the file */
int truncate; /* True if feeder connection truncate the feed file */
int conns_served;
int64_t bytes_served;
int64_t feed_max_size; /* maximum storage size, zero means unlimited */
int64_t feed_write_index; /* current write position in feed (it wraps around) */
int64_t feed_size; /* current size of feed */
struct FFServerStream *next_feed;
} FFServerStream;
typedef struct FFServerConfig {
char *filename;
FFServerStream *first_feed; /* contains only feeds */
FFServerStream *first_stream; /* contains all streams, including feeds */
unsigned int nb_max_http_connections;
unsigned int nb_max_connections;
uint64_t max_bandwidth;
int debug;
int bitexact;
char logfilename[1024];
struct sockaddr_in http_addr;
struct sockaddr_in rtsp_addr;
int errors;
int warnings;
int use_defaults;
// Following variables MUST NOT be used outside configuration parsing code.
enum AVCodecID guessed_audio_codec_id;
enum AVCodecID guessed_video_codec_id;
AVDictionary *video_opts; /* AVOptions for video encoder */
AVDictionary *audio_opts; /* AVOptions for audio encoder */
AVCodecContext *dummy_actx; /* Used internally to test audio AVOptions. */
AVCodecContext *dummy_vctx; /* Used internally to test video AVOptions. */
int no_audio;
int no_video;
int line_num;
int stream_use_defaults;
} FFServerConfig;
void ffserver_get_arg(char *buf, int buf_size, const char **pp);
void ffserver_parse_acl_row(FFServerStream *stream, FFServerStream* feed,
FFServerIPAddressACL *ext_acl,
const char *p, const char *filename, int line_num);
int ffserver_parse_ffconfig(const char *filename, FFServerConfig *config);
void ffserver_free_child_args(void *argsp);
#endif /* FFTOOLS_FFSERVER_CONFIG_H */
libavcodec/.gitignore
+4
View File
@@ -0,0 +1,4 @@
/*_tablegen
/*_tables.c
/*_tables.h
/bsf_list.c
libavcodec/Makefile
+51 -29
View File
@@ -1,7 +1,9 @@
NAME = avcodec
DESC = FFmpeg codec library
HEADERS = avcodec.h \
HEADERS = ac3_parser.h \
adts_parser.h \
avcodec.h \
avdct.h \
avfft.h \
d3d11va.h \
@@ -12,15 +14,15 @@ HEADERS = avcodec.h \
mediacodec.h \
qsv.h \
vaapi.h \
vda.h \
vdpau.h \
version.h \
videotoolbox.h \
vorbis_parser.h \
xvmc.h \
OBJS = allcodecs.o \
audioconvert.o \
OBJS = ac3_parser.o \
adts_parser.o \
allcodecs.o \
avdct.o \
avpacket.o \
avpicture.o \
@@ -45,8 +47,6 @@ OBJS = allcodecs.o \
profiles.o \
qsv_api.o \
raw.o \
resample.o \
resample2.o \
utils.o \
vorbis_parser.o \
xiph.o \
@@ -54,11 +54,17 @@ OBJS = allcodecs.o \
# subsystems
OBJS-$(CONFIG_AANDCTTABLES) += aandcttab.o
OBJS-$(CONFIG_AC3DSP) += ac3dsp.o ac3.o ac3tab.o
OBJS-$(CONFIG_ADTS_HEADER) += adts_header.o mpeg4audio.o
OBJS-$(CONFIG_AMF) += amfenc.o
OBJS-$(CONFIG_AUDIO_FRAME_QUEUE) += audio_frame_queue.o
OBJS-$(CONFIG_AUDIODSP) += audiodsp.o
OBJS-$(CONFIG_BLOCKDSP) += blockdsp.o
OBJS-$(CONFIG_BSWAPDSP) += bswapdsp.o
OBJS-$(CONFIG_CABAC) += cabac.o
OBJS-$(CONFIG_CBS) += cbs.o
OBJS-$(CONFIG_CBS_H264) += cbs_h2645.o h2645_parse.o
OBJS-$(CONFIG_CBS_H265) += cbs_h2645.o h2645_parse.o
OBJS-$(CONFIG_CBS_MPEG2) += cbs_mpeg2.o
OBJS-$(CONFIG_CRYSTALHD) += crystalhd.o
OBJS-$(CONFIG_DCT) += dct.o dct32_fixed.o dct32_float.o
OBJS-$(CONFIG_ERROR_RESILIENCE) += error_resilience.o
@@ -91,7 +97,6 @@ OBJS-$(CONFIG_INTRAX8) += intrax8.o intrax8dsp.o msmpeg4data.o
OBJS-$(CONFIG_IVIDSP) += ivi_dsp.o
OBJS-$(CONFIG_JNI) += ffjni.o jni.o
OBJS-$(CONFIG_JPEGTABLES) += jpegtables.o
OBJS-$(CONFIG_LIBXVID) += libxvid_rc.o
OBJS-$(CONFIG_LLAUDDSP) += lossless_audiodsp.o
OBJS-$(CONFIG_LLVIDDSP) += lossless_videodsp.o
OBJS-$(CONFIG_LLVIDENCDSP) += lossless_videoencdsp.o
@@ -146,10 +151,10 @@ OBJS-$(CONFIG_ZERO12V_DECODER) += 012v.o
OBJS-$(CONFIG_A64MULTI_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps_float.o \
aacadtsdec.o mpeg4audio.o kbdwin.o \
mpeg4audio.o kbdwin.o \
sbrdsp.o aacpsdsp_float.o cbrt_data.o
OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
aacadtsdec.o mpeg4audio.o kbdwin.o \
mpeg4audio.o kbdwin.o \
sbrdsp_fixed.o aacpsdsp_fixed.o cbrt_data_fixed.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacenctab.o \
aacpsy.o aactab.o \
@@ -183,6 +188,8 @@ OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o \
OBJS-$(CONFIG_ANM_DECODER) += anm.o
OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_APTX_DECODER) += aptx.o
OBJS-$(CONFIG_APTX_ENCODER) += aptx.o
OBJS-$(CONFIG_APNG_DECODER) += png.o pngdec.o pngdsp.o
OBJS-$(CONFIG_APNG_ENCODER) += png.o pngenc.o
OBJS-$(CONFIG_SSA_DECODER) += assdec.o ass.o
@@ -331,18 +338,18 @@ OBJS-$(CONFIG_H264_DECODER) += h264dec.o h264_cabac.o h264_cavlc.o \
h264_mb.o h264_picture.o \
h264_refs.o h264_sei.o \
h264_slice.o h264data.o
OBJS-$(CONFIG_H264_CUVID_DECODER) += cuvid.o
OBJS-$(CONFIG_H264_AMF_ENCODER) += amfenc_h264.o
OBJS-$(CONFIG_H264_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_H264_MEDIACODEC_DECODER) += mediacodecdec.o
OBJS-$(CONFIG_H264_MMAL_DECODER) += mmaldec.o
OBJS-$(CONFIG_H264_NVENC_ENCODER) += nvenc_h264.o
OBJS-$(CONFIG_NVENC_ENCODER) += nvenc_h264.o
OBJS-$(CONFIG_NVENC_H264_ENCODER) += nvenc_h264.o
OBJS-$(CONFIG_H264_VDA_DECODER) += vda_h264_dec.o
OBJS-$(CONFIG_H264_OMX_ENCODER) += omx.o
OBJS-$(CONFIG_H264_QSV_DECODER) += qsvdec_h2645.o
OBJS-$(CONFIG_H264_QSV_ENCODER) += qsvenc_h264.o
OBJS-$(CONFIG_H264_RKMPP_DECODER) += rkmppdec.o
OBJS-$(CONFIG_H264_VAAPI_ENCODER) += vaapi_encode_h264.o vaapi_encode_h26x.o
OBJS-$(CONFIG_H264_VAAPI_ENCODER) += vaapi_encode_h264.o
OBJS-$(CONFIG_H264_VIDEOTOOLBOX_ENCODER) += videotoolboxenc.o
OBJS-$(CONFIG_H264_V4L2M2M_DECODER) += v4l2_m2m_dec.o
OBJS-$(CONFIG_H264_V4L2M2M_ENCODER) += v4l2_m2m_enc.o
@@ -351,7 +358,8 @@ OBJS-$(CONFIG_HAP_ENCODER) += hapenc.o hap.o
OBJS-$(CONFIG_HEVC_DECODER) += hevcdec.o hevc_mvs.o \
hevc_cabac.o hevc_refs.o hevcpred.o \
hevcdsp.o hevc_filter.o hevc_data.o
OBJS-$(CONFIG_HEVC_CUVID_DECODER) += cuvid.o
OBJS-$(CONFIG_HEVC_AMF_ENCODER) += amfenc_hevc.o
OBJS-$(CONFIG_HEVC_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_HEVC_MEDIACODEC_DECODER) += mediacodecdec.o
OBJS-$(CONFIG_HEVC_NVENC_ENCODER) += nvenc_hevc.o
OBJS-$(CONFIG_NVENC_HEVC_ENCODER) += nvenc_hevc.o
@@ -359,7 +367,7 @@ OBJS-$(CONFIG_HEVC_QSV_DECODER) += qsvdec_h2645.o
OBJS-$(CONFIG_HEVC_QSV_ENCODER) += qsvenc_hevc.o hevc_ps_enc.o \
hevc_data.o
OBJS-$(CONFIG_HEVC_RKMPP_DECODER) += rkmppdec.o
OBJS-$(CONFIG_HEVC_VAAPI_ENCODER) += vaapi_encode_h265.o vaapi_encode_h26x.o
OBJS-$(CONFIG_HEVC_VAAPI_ENCODER) += vaapi_encode_h265.o
OBJS-$(CONFIG_HEVC_V4L2M2M_DECODER) += v4l2_m2m_dec.o
OBJS-$(CONFIG_HEVC_V4L2M2M_ENCODER) += v4l2_m2m_enc.o
OBJS-$(CONFIG_HNM4_VIDEO_DECODER) += hnm4video.o
@@ -396,6 +404,7 @@ OBJS-$(CONFIG_M101_DECODER) += m101.o
OBJS-$(CONFIG_MACE3_DECODER) += mace.o
OBJS-$(CONFIG_MACE6_DECODER) += mace.o
OBJS-$(CONFIG_MAGICYUV_DECODER) += magicyuv.o
OBJS-$(CONFIG_MAGICYUV_ENCODER) += magicyuvenc.o
OBJS-$(CONFIG_MDEC_DECODER) += mdec.o mpeg12.o mpeg12data.o
OBJS-$(CONFIG_METASOUND_DECODER) += metasound.o metasound_data.o \
twinvq.o
@@ -405,6 +414,7 @@ OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpegenc_common.o \
mjpegenc_huffman.o
OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o
OBJS-$(CONFIG_MJPEG_QSV_ENCODER) += qsvenc_jpeg.o
OBJS-$(CONFIG_MJPEG_VAAPI_ENCODER) += vaapi_encode_mjpeg.o
OBJS-$(CONFIG_MLP_DECODER) += mlpdec.o mlpdsp.o
OBJS-$(CONFIG_MLP_ENCODER) += mlpenc.o mlp.o
@@ -616,7 +626,7 @@ OBJS-$(CONFIG_VC1_DECODER) += vc1dec.o vc1_block.o vc1_loopfilter.o
vc1_mc.o vc1_pred.o vc1.o vc1data.o \
msmpeg4dec.o msmpeg4.o msmpeg4data.o \
wmv2dsp.o wmv2data.o
OBJS-$(CONFIG_VC1_CUVID_DECODER) += cuvid.o
OBJS-$(CONFIG_VC1_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_VC1_MMAL_DECODER) += mmaldec.o
OBJS-$(CONFIG_VC1_QSV_DECODER) += qsvdec_other.o
OBJS-$(CONFIG_VC1_V4L2M2M_DECODER) += v4l2_m2m_dec.o
@@ -635,7 +645,7 @@ OBJS-$(CONFIG_VP6_DECODER) += vp6.o vp56.o vp56data.o \
vp6dsp.o vp56rac.o
OBJS-$(CONFIG_VP7_DECODER) += vp8.o vp56rac.o
OBJS-$(CONFIG_VP8_DECODER) += vp8.o vp56rac.o
OBJS-$(CONFIG_VP8_CUVID_DECODER) += cuvid.o
OBJS-$(CONFIG_VP8_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_VP8_MEDIACODEC_DECODER) += mediacodecdec.o
OBJS-$(CONFIG_VP8_QSV_DECODER) += qsvdec_other.o
OBJS-$(CONFIG_VP8_RKMPP_DECODER) += rkmppdec.o
@@ -645,7 +655,7 @@ OBJS-$(CONFIG_VP8_V4L2M2M_ENCODER) += v4l2_m2m_enc.o
OBJS-$(CONFIG_VP9_DECODER) += vp9.o vp9data.o vp9dsp.o vp9lpf.o vp9recon.o \
vp9block.o vp9prob.o vp9mvs.o vp56rac.o \
vp9dsp_8bpp.o vp9dsp_10bpp.o vp9dsp_12bpp.o
OBJS-$(CONFIG_VP9_CUVID_DECODER) += cuvid.o
OBJS-$(CONFIG_VP9_CUVID_DECODER) += cuviddec.o
OBJS-$(CONFIG_VP9_MEDIACODEC_DECODER) += mediacodecdec.o
OBJS-$(CONFIG_VP9_RKMPP_DECODER) += rkmppdec.o
OBJS-$(CONFIG_VP9_VAAPI_ENCODER) += vaapi_encode_vp9.o
@@ -822,8 +832,8 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcmenc.o adpcm_data.o
# hardware accelerators
OBJS-$(CONFIG_D3D11VA) += dxva2.o
OBJS-$(CONFIG_DXVA2) += dxva2.o
OBJS-$(CONFIG_NVDEC) += nvdec.o
OBJS-$(CONFIG_VAAPI) += vaapi_decode.o
OBJS-$(CONFIG_VDA) += vda.o videotoolbox.o
OBJS-$(CONFIG_VIDEOTOOLBOX) += videotoolbox.o
OBJS-$(CONFIG_VDPAU) += vdpau.o
@@ -831,36 +841,44 @@ OBJS-$(CONFIG_H263_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_H263_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
OBJS-$(CONFIG_H264_D3D11VA_HWACCEL) += dxva2_h264.o
OBJS-$(CONFIG_H264_DXVA2_HWACCEL) += dxva2_h264.o
OBJS-$(CONFIG_H264_NVDEC_HWACCEL) += nvdec_h264.o
OBJS-$(CONFIG_H264_QSV_HWACCEL) += qsvdec_h2645.o
OBJS-$(CONFIG_H264_VAAPI_HWACCEL) += vaapi_h264.o
OBJS-$(CONFIG_H264_VDA_HWACCEL) += vda_h264.o
OBJS-$(CONFIG_H264_VDPAU_HWACCEL) += vdpau_h264.o
OBJS-$(CONFIG_H264_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
OBJS-$(CONFIG_HEVC_D3D11VA_HWACCEL) += dxva2_hevc.o
OBJS-$(CONFIG_HEVC_DXVA2_HWACCEL) += dxva2_hevc.o
OBJS-$(CONFIG_HEVC_NVDEC_HWACCEL) += nvdec_hevc.o
OBJS-$(CONFIG_HEVC_QSV_HWACCEL) += qsvdec_h2645.o
OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL) += vaapi_hevc.o
OBJS-$(CONFIG_HEVC_VDPAU_HWACCEL) += vdpau_hevc.o
OBJS-$(CONFIG_MPEG1_NVDEC_HWACCEL) += nvdec_mpeg12.o
OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL) += vdpau_mpeg12.o
OBJS-$(CONFIG_MPEG1_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
OBJS-$(CONFIG_MPEG1_XVMC_HWACCEL) += mpegvideo_xvmc.o
OBJS-$(CONFIG_MPEG2_D3D11VA_HWACCEL) += dxva2_mpeg2.o
OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL) += dxva2_mpeg2.o
OBJS-$(CONFIG_MPEG2_NVDEC_HWACCEL) += nvdec_mpeg12.o
OBJS-$(CONFIG_MPEG2_QSV_HWACCEL) += qsvdec_other.o
OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL) += vaapi_mpeg2.o
OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL) += vdpau_mpeg12.o
OBJS-$(CONFIG_MPEG2_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
OBJS-$(CONFIG_MPEG2_XVMC_HWACCEL) += mpegvideo_xvmc.o
OBJS-$(CONFIG_MPEG4_NVDEC_HWACCEL) += nvdec_mpeg4.o
OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL) += vdpau_mpeg4.o
OBJS-$(CONFIG_MPEG4_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
OBJS-$(CONFIG_VC1_D3D11VA_HWACCEL) += dxva2_vc1.o
OBJS-$(CONFIG_VC1_DXVA2_HWACCEL) += dxva2_vc1.o
OBJS-$(CONFIG_VC1_NVDEC_HWACCEL) += nvdec_vc1.o
OBJS-$(CONFIG_VC1_QSV_HWACCEL) += qsvdec_other.o
OBJS-$(CONFIG_VC1_VAAPI_HWACCEL) += vaapi_vc1.o
OBJS-$(CONFIG_VC1_VDPAU_HWACCEL) += vdpau_vc1.o
OBJS-$(CONFIG_VP8_NVDEC_HWACCEL) += nvdec_vp8.o
OBJS-$(CONFIG_VP8_VAAPI_HWACCEL) += vaapi_vp8.o
OBJS-$(CONFIG_VP9_D3D11VA_HWACCEL) += dxva2_vp9.o
OBJS-$(CONFIG_VP9_DXVA2_HWACCEL) += dxva2_vp9.o
OBJS-$(CONFIG_VP9_NVDEC_HWACCEL) += nvdec_vp9.o
OBJS-$(CONFIG_VP9_VAAPI_HWACCEL) += vaapi_vp9.o
OBJS-$(CONFIG_VP8_QSV_HWACCEL) += qsvdec_other.o
@@ -880,7 +898,6 @@ OBJS-$(CONFIG_MXF_MUXER) += dnxhddata.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
OBJS-$(CONFIG_NUT_DEMUXER) += mpegaudiodata.o mpeg4audio.o
OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_SPDIF_MUXER) += dca.o
OBJS-$(CONFIG_TAK_DEMUXER) += tak.o
OBJS-$(CONFIG_WEBM_MUXER) += mpeg4audio.o
@@ -951,15 +968,14 @@ OBJS-$(CONFIG_LIBX262_ENCODER) += libx264.o
OBJS-$(CONFIG_LIBX264_ENCODER) += libx264.o
OBJS-$(CONFIG_LIBX265_ENCODER) += libx265.o
OBJS-$(CONFIG_LIBXAVS_ENCODER) += libxavs.o
OBJS-$(CONFIG_LIBXVID_ENCODER) += libxvid.o
OBJS-$(CONFIG_LIBXVID_ENCODER) += libxvid.o libxvid_rc.o
OBJS-$(CONFIG_LIBZVBI_TELETEXT_DECODER) += libzvbi-teletextdec.o ass.o
# parsers
OBJS-$(CONFIG_AAC_LATM_PARSER) += latm_parser.o
OBJS-$(CONFIG_AAC_PARSER) += aac_parser.o aac_ac3_parser.o \
aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o \
aac_ac3_parser.o
mpeg4audio.o
OBJS-$(CONFIG_AC3_PARSER) += ac3tab.o aac_ac3_parser.o
OBJS-$(CONFIG_ADX_PARSER) += adx_parser.o adx.o
OBJS-$(CONFIG_BMP_PARSER) += bmp_parser.o
OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs_parser.o
@@ -989,7 +1005,8 @@ OBJS-$(CONFIG_PNG_PARSER) += png_parser.o
OBJS-$(CONFIG_MPEGAUDIO_PARSER) += mpegaudio_parser.o
OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o \
mpeg12.o mpeg12data.o
OBJS-$(CONFIG_OPUS_PARSER) += opus_parser.o opus.o vorbis_data.o
OBJS-$(CONFIG_OPUS_PARSER) += opus_parser.o opus.o opustab.o \
opus_rc.o vorbis_data.o
OBJS-$(CONFIG_PNG_PARSER) += png_parser.o
OBJS-$(CONFIG_PNM_PARSER) += pnm_parser.o pnm.o
OBJS-$(CONFIG_RV30_PARSER) += rv34_parser.o
@@ -1004,14 +1021,16 @@ OBJS-$(CONFIG_VP9_PARSER) += vp9_parser.o
OBJS-$(CONFIG_XMA_PARSER) += xma_parser.o
# bitstream filters
OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o aacadtsdec.o \
mpeg4audio.o
OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o mpeg4audio.o
OBJS-$(CONFIG_CHOMP_BSF) += chomp_bsf.o
OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += dump_extradata_bsf.o
OBJS-$(CONFIG_DCA_CORE_BSF) += dca_core_bsf.o
OBJS-$(CONFIG_EXTRACT_EXTRADATA_BSF) += extract_extradata_bsf.o \
h2645_parse.o
OBJS-$(CONFIG_H264_METADATA_BSF) += h264_metadata_bsf.o
OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF) += h264_mp4toannexb_bsf.o
OBJS-$(CONFIG_H264_REDUNDANT_PPS_BSF) += h264_redundant_pps_bsf.o
OBJS-$(CONFIG_HEVC_METADATA_BSF) += h265_metadata_bsf.o
OBJS-$(CONFIG_HEVC_MP4TOANNEXB_BSF) += hevc_mp4toannexb_bsf.o
OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF) += imx_dump_header_bsf.o
OBJS-$(CONFIG_MJPEG2JPEG_BSF) += mjpeg2jpeg_bsf.o
@@ -1020,10 +1039,12 @@ OBJS-$(CONFIG_MPEG4_UNPACK_BFRAMES_BSF) += mpeg4_unpack_bframes_bsf.o
OBJS-$(CONFIG_MOV2TEXTSUB_BSF) += movsub_bsf.o
OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o \
mpegaudiodata.o
OBJS-$(CONFIG_MPEG2_METADATA_BSF) += mpeg2_metadata_bsf.o
OBJS-$(CONFIG_NOISE_BSF) += noise_bsf.o
OBJS-$(CONFIG_NULL_BSF) += null_bsf.o
OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF) += remove_extradata_bsf.o
OBJS-$(CONFIG_TEXT2MOVSUB_BSF) += movsub_bsf.o
OBJS-$(CONFIG_TRACE_HEADERS_BSF) += trace_headers_bsf.o
OBJS-$(CONFIG_VP9_RAW_REORDER_BSF) += vp9_raw_reorder_bsf.o
OBJS-$(CONFIG_VP9_SUPERFRAME_BSF) += vp9_superframe_bsf.o
OBJS-$(CONFIG_VP9_SUPERFRAME_SPLIT_BSF) += vp9_superframe_split_bsf.o
@@ -1048,6 +1069,7 @@ SKIPHEADERS += %_tablegen.h \
aacenc_quantization_misc.h \
$(ARCH)/vp56_arith.h \
SKIPHEADERS-$(CONFIG_AMF) += amfenc.h
SKIPHEADERS-$(CONFIG_D3D11VA) += d3d11va.h dxva2_internal.h
SKIPHEADERS-$(CONFIG_DXVA2) += dxva2.h dxva2_internal.h
SKIPHEADERS-$(CONFIG_JNI) += ffjni.h
@@ -1060,9 +1082,8 @@ SKIPHEADERS-$(CONFIG_QSVDEC) += qsvdec.h
SKIPHEADERS-$(CONFIG_QSVENC) += qsvenc.h
SKIPHEADERS-$(CONFIG_XVMC) += xvmc.h
SKIPHEADERS-$(CONFIG_VAAPI) += vaapi_decode.h vaapi_encode.h
SKIPHEADERS-$(CONFIG_VDA) += vda.h vda_vt_internal.h
SKIPHEADERS-$(CONFIG_VDPAU) += vdpau.h vdpau_internal.h
SKIPHEADERS-$(CONFIG_VIDEOTOOLBOX) += videotoolbox.h vda_vt_internal.h
SKIPHEADERS-$(CONFIG_VIDEOTOOLBOX) += videotoolbox.h vt_internal.h
SKIPHEADERS-$(CONFIG_V4L2_M2M) += v4l2_buffers.h v4l2_context.h v4l2_m2m.h
TESTPROGS = avpacket \
@@ -1082,6 +1103,7 @@ TESTPROGS-$(CONFIG_GOLOMB) += golomb
TESTPROGS-$(CONFIG_IDCTDSP) += dct
TESTPROGS-$(CONFIG_IIRFILTER) += iirfilter
TESTPROGS-$(HAVE_MMX) += motion
TESTPROGS-$(CONFIG_MPEGVIDEO) += mpeg12framerate
TESTPROGS-$(CONFIG_RANGECODER) += rangecoder
TESTPROGS-$(CONFIG_SNOW_ENCODER) += snowenc
libavcodec/aac_ac3_parser.c
-12
View File
@@ -86,20 +86,8 @@ get_next:
the frame). */
if (avctx->codec_id != AV_CODEC_ID_AAC) {
avctx->sample_rate = s->sample_rate;
/* (E-)AC-3: allow downmixing to stereo or mono */
if (s->channels > 1 &&
avctx->request_channel_layout == AV_CH_LAYOUT_MONO) {
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
} else if (s->channels > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
avctx->channels = 2;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
} else {
avctx->channels = s->channels;
avctx->channel_layout = s->channel_layout;
}
s1->duration = s->samples;
avctx->audio_service_type = s->service_type;
}
libavcodec/aac_adtstoasc_bsf.c
+8 -7
View File
@@ -19,8 +19,9 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "adts_header.h"
#include "adts_parser.h"
#include "avcodec.h"
#include "aacadtsdec.h"
#include "bsf.h"
#include "put_bits.h"
#include "get_bits.h"
@@ -52,12 +53,12 @@ static int aac_adtstoasc_filter(AVBSFContext *bsfc, AVPacket *out)
if (bsfc->par_in->extradata && in->size >= 2 && (AV_RB16(in->data) >> 4) != 0xfff)
goto finish;
if (in->size < AAC_ADTS_HEADER_SIZE)
if (in->size < AV_AAC_ADTS_HEADER_SIZE)
goto packet_too_small;
init_get_bits(&gb, in->data, AAC_ADTS_HEADER_SIZE * 8);
init_get_bits(&gb, in->data, AV_AAC_ADTS_HEADER_SIZE * 8);
if (avpriv_aac_parse_header(&gb, &hdr) < 0) {
if (ff_adts_header_parse(&gb, &hdr) < 0) {
av_log(bsfc, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
ret = AVERROR_INVALIDDATA;
goto fail;
@@ -70,10 +71,10 @@ static int aac_adtstoasc_filter(AVBSFContext *bsfc, AVPacket *out)
goto fail;
}
in->size -= AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
in->size -= AV_AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
if (in->size <= 0)
goto packet_too_small;
in->data += AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
in->data += AV_AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
if (!ctx->first_frame_done) {
int pce_size = 0;
@@ -91,7 +92,7 @@ static int aac_adtstoasc_filter(AVBSFContext *bsfc, AVPacket *out)
goto fail;
}
init_put_bits(&pb, pce_data, MAX_PCE_SIZE);
pce_size = avpriv_copy_pce_data(&pb, &gb)/8;
pce_size = ff_copy_pce_data(&pb, &gb) / 8;
flush_put_bits(&pb);
in->size -= get_bits_count(&gb)/8;
in->data += get_bits_count(&gb)/8;
libavcodec/aac_parser.c
+6 -4
View File
@@ -22,7 +22,8 @@
#include "parser.h"
#include "aac_ac3_parser.h"
#include "aacadtsdec.h"
#include "adts_header.h"
#include "adts_parser.h"
#include "get_bits.h"
#include "mpeg4audio.h"
@@ -38,9 +39,10 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
} tmp;
tmp.u64 = av_be2ne64(state);
init_get_bits(&bits, tmp.u8+8-AAC_ADTS_HEADER_SIZE, AAC_ADTS_HEADER_SIZE * 8);
init_get_bits(&bits, tmp.u8 + 8 - AV_AAC_ADTS_HEADER_SIZE,
AV_AAC_ADTS_HEADER_SIZE * 8);
if ((size = avpriv_aac_parse_header(&bits, &hdr)) < 0)
if ((size = ff_adts_header_parse(&bits, &hdr)) < 0)
return 0;
*need_next_header = 0;
*new_frame_start = 1;
@@ -54,7 +56,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
static av_cold int aac_parse_init(AVCodecParserContext *s1)
{
AACAC3ParseContext *s = s1->priv_data;
s->header_size = AAC_ADTS_HEADER_SIZE;
s->header_size = AV_AAC_ADTS_HEADER_SIZE;
s->sync = aac_sync;
return 0;
}
libavcodec/aacdec.c
+1 -1
View File
@@ -50,11 +50,11 @@
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
#include "adts_header.h"
#include "cbrt_data.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "aacadtsdec.h"
#include "profiles.h"
#include "libavutil/intfloat.h"
libavcodec/aacdec_fixed.c
+4 -4
View File
@@ -75,11 +75,11 @@
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
#include "adts_header.h"
#include "cbrt_data.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "aacadtsdec.h"
#include "profiles.h"
#include "libavutil/intfloat.h"
@@ -307,9 +307,9 @@ static av_always_inline void predict(PredictorState *ps, int *coef,
if (shift < 31) {
if (shift > 0) {
*coef += (pv.mant + (1 << (shift - 1))) >> shift;
*coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
} else
*coef += pv.mant << -shift;
*coef += (unsigned)pv.mant << -shift;
}
}
@@ -394,7 +394,7 @@ static void apply_dependent_coupling_fixed(AACContext *ac,
for (k = offsets[i]; k < offsets[i + 1]; k++) {
tmp = (int)(((int64_t)src[group * 128 + k] * c + \
(int64_t)0x1000000000) >> 37);
dest[group * 128 + k] += tmp * (1 << shift);
dest[group * 128 + k] += tmp * (1U << shift);
}
}
}
libavcodec/aacdec_template.c
+1 -1
View File
@@ -2955,7 +2955,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags, ret;
size = avpriv_aac_parse_header(gb, &hdr_info);
size = ff_adts_header_parse(gb, &hdr_info);
if (size > 0) {
if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
// This is 2 for "VLB " audio in NSV files.
libavcodec/aacenc.c
+82 -15
View File
@@ -50,17 +50,59 @@
static AVOnce aac_table_init = AV_ONCE_INIT;
static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
{
int i, j;
AACEncContext *s = avctx->priv_data;
AACPCEInfo *pce = &s->pce;
const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
put_bits(pb, 4, 0);
put_bits(pb, 2, avctx->profile);
put_bits(pb, 4, s->samplerate_index);
put_bits(pb, 4, pce->num_ele[0]); /* Front */
put_bits(pb, 4, pce->num_ele[1]); /* Side */
put_bits(pb, 4, pce->num_ele[2]); /* Back */
put_bits(pb, 2, pce->num_ele[3]); /* LFE */
put_bits(pb, 3, 0); /* Assoc data */
put_bits(pb, 4, 0); /* CCs */
put_bits(pb, 1, 0); /* Stereo mixdown */
put_bits(pb, 1, 0); /* Mono mixdown */
put_bits(pb, 1, 0); /* Something else */
for (i = 0; i < 4; i++) {
for (j = 0; j < pce->num_ele[i]; j++) {
if (i < 3)
put_bits(pb, 1, pce->pairing[i][j]);
put_bits(pb, 4, pce->index[i][j]);
}
}
avpriv_align_put_bits(pb);
put_bits(pb, 8, strlen(aux_data));
avpriv_put_string(pb, aux_data, 0);
}
/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
*/
static void put_audio_specific_config(AVCodecContext *avctx)
static int put_audio_specific_config(AVCodecContext *avctx)
{
PutBitContext pb;
AACEncContext *s = avctx->priv_data;
int channels = s->channels - (s->channels == 8 ? 1 : 0);
int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
const int max_size = 32;
init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
avctx->extradata = av_mallocz(max_size);
if (!avctx->extradata)
return AVERROR(ENOMEM);
init_put_bits(&pb, avctx->extradata, max_size);
put_bits(&pb, 5, s->profile+1); //profile
put_bits(&pb, 4, s->samplerate_index); //sample rate index
put_bits(&pb, 4, channels);
@@ -68,12 +110,17 @@ static void put_audio_specific_config(AVCodecContext *avctx)
put_bits(&pb, 1, 0); //frame length - 1024 samples
put_bits(&pb, 1, 0); //does not depend on core coder
put_bits(&pb, 1, 0); //is not extension
if (s->needs_pce)
put_pce(&pb, avctx);
//Explicitly Mark SBR absent
put_bits(&pb, 11, 0x2b7); //sync extension
put_bits(&pb, 5, AOT_SBR);
put_bits(&pb, 1, 0);
flush_put_bits(&pb);
avctx->extradata_size = put_bits_count(&pb) >> 3;
return 0;
}
void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
@@ -488,7 +535,7 @@ static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
{
int ch;
int end = 2048 + (frame ? frame->nb_samples : 0);
const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
const uint8_t *channel_map = s->reorder_map;
/* copy and remap input samples */
for (ch = 0; ch < s->channels; ch++) {
@@ -895,7 +942,6 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
int ch;
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
for(ch = 0; ch < s->channels; ch++)
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
@@ -920,16 +966,35 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
/* Constants */
s->last_frame_pb_count = 0;
avctx->extradata_size = 5;
avctx->frame_size = 1024;
avctx->initial_padding = 1024;
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
/* Channel map and unspecified bitrate guessing */
s->channels = avctx->channels;
ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
"Unsupported number of channels: %d\n", s->channels);
s->needs_pce = 1;
for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) {
if (avctx->channel_layout == aac_normal_chan_layouts[i]) {
s->needs_pce = s->options.pce;
break;
}
}
if (s->needs_pce) {
for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
if (avctx->channel_layout == aac_pce_configs[i].layout)
break;
ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout\n");
av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout\n");
s->pce = aac_pce_configs[i];
s->reorder_map = s->pce.reorder_map;
s->chan_map = s->pce.config_map;
} else {
s->reorder_map = aac_chan_maps[s->channels - 1];
s->chan_map = aac_chan_configs[s->channels - 1];
}
if (!avctx->bit_rate) {
for (i = 1; i <= s->chan_map[0]; i++) {
avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
@@ -1015,7 +1080,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
if ((ret = alloc_buffers(avctx, s)) < 0)
goto fail;
put_audio_specific_config(avctx);
if ((ret = put_audio_specific_config(avctx)))
goto fail;
sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
@@ -1052,24 +1118,25 @@ fail:
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption aacenc_options[] = {
{"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
{"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_FAST}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
{"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
{"fast", "Default fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
{"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
{"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
{"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
{"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
{"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
{"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
{NULL}
};
static const AVClass aacenc_class = {
"AAC encoder",
av_default_item_name,
aacenc_options,
LIBAVUTIL_VERSION_INT,
.class_name = "AAC encoder",
.item_name = av_default_item_name,
.option = aacenc_options,
.version = LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault aac_encode_defaults[] = {
libavcodec/aacenc.h
+285 -1
View File
@@ -45,6 +45,7 @@ typedef struct AACEncOptions {
int pns;
int tns;
int ltp;
int pce;
int pred;
int mid_side;
int intensity_stereo;
@@ -89,6 +90,286 @@ typedef struct AACQuantizeBandCostCacheEntry {
uint16_t generation;
} AACQuantizeBandCostCacheEntry;
typedef struct AACPCEInfo {
int64_t layout;
int num_ele[4]; ///< front, side, back, lfe
int pairing[3][8]; ///< front, side, back
int index[4][8]; ///< front, side, back, lfe
uint8_t config_map[16]; ///< configs the encoder's channel specific settings
uint8_t reorder_map[16]; ///< maps channels from lavc to aac order
} AACPCEInfo;
/**
* List of PCE (Program Configuration Element) for the channel layouts listed
* in channel_layout.h
*
* For those wishing in the future to add other layouts:
*
* - num_ele: number of elements in each group of front, side, back, lfe channels
* (an element is of type SCE (single channel), CPE (channel pair) for
* the first 3 groups; and is LFE for LFE group).
*
* - pairing: 0 for an SCE element or 1 for a CPE; does not apply to LFE group
*
* - index: there are three independent indices for SCE, CPE and LFE;
* they are incremented irrespective of the group to which the element belongs;
* they are not reset when going from one group to another
*
* Example: for 7.0 channel layout,
* .pairing = { { 1, 0 }, { 1 }, { 1 }, }, (3 CPE and 1 SCE in front group)
* .index = { { 0, 0 }, { 1 }, { 2 }, },
* (index is 0 for the single SCE but goes from 0 to 2 for the CPEs)
*
* The index order impacts the channel ordering. But is otherwise arbitrary
* (the sequence could have been 2, 0, 1 instead of 0, 1, 2).
*
* Spec allows for discontinuous indices, e.g. if one has a total of two SCE,
* SCE.0 SCE.15 is OK per spec; BUT it won't be decoded by our AAC decoder
* which at this time requires that indices fully cover some range starting
* from 0 (SCE.1 SCE.0 is OK but not SCE.0 SCE.15).
*
* - config_map: total number of elements and their types. Beware, the way the
* types are ordered impacts the final channel ordering.
*
* - reorder_map: reorders the channels.
*
*/
static const AACPCEInfo aac_pce_configs[] = {
{
.layout = AV_CH_LAYOUT_MONO,
.num_ele = { 1, 0, 0, 0 },
.pairing = { { 0 }, },
.index = { { 0 }, },
.config_map = { 1, TYPE_SCE, },
.reorder_map = { 0 },
},
{
.layout = AV_CH_LAYOUT_STEREO,
.num_ele = { 1, 0, 0, 0 },
.pairing = { { 1 }, },
.index = { { 0 }, },
.config_map = { 1, TYPE_CPE, },
.reorder_map = { 0, 1 },
},
{
.layout = AV_CH_LAYOUT_2POINT1,
.num_ele = { 1, 0, 0, 1 },
.pairing = { { 1 }, },
.index = { { 0 },{ 0 },{ 0 },{ 0 } },
.config_map = { 2, TYPE_CPE, TYPE_LFE },
.reorder_map = { 0, 1, 2 },
},
{
.layout = AV_CH_LAYOUT_2_1,
.num_ele = { 1, 0, 1, 0 },
.pairing = { { 1 },{ 0 },{ 0 } },
.index = { { 0 },{ 0 },{ 0 }, },
.config_map = { 2, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2 },
},
{
.layout = AV_CH_LAYOUT_SURROUND,
.num_ele = { 2, 0, 0, 0 },
.pairing = { { 1, 0 }, },
.index = { { 0, 0 }, },
.config_map = { 2, TYPE_CPE, TYPE_SCE, },
.reorder_map = { 0, 1, 2 },
},
{
.layout = AV_CH_LAYOUT_3POINT1,
.num_ele = { 2, 0, 0, 1 },
.pairing = { { 1, 0 }, },
.index = { { 0, 0 }, { 0 }, { 0 }, { 0 }, },
.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_LFE },
.reorder_map = { 0, 1, 2, 3 },
},
{
.layout = AV_CH_LAYOUT_4POINT0,
.num_ele = { 2, 0, 1, 0 },
.pairing = { { 1, 0 }, { 0 }, { 0 }, },
.index = { { 0, 0 }, { 0 }, { 1 } },
.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3 },
},
{
.layout = AV_CH_LAYOUT_4POINT1,
.num_ele = { 2, 1, 1, 0 },
.pairing = { { 1, 0 }, { 0 }, { 0 }, },
.index = { { 0, 0 }, { 1 }, { 2 }, { 0 } },
.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4 },
},
{
.layout = AV_CH_LAYOUT_2_2,
.num_ele = { 1, 1, 0, 0 },
.pairing = { { 1 }, { 1 }, },
.index = { { 0 }, { 1 }, },
.config_map = { 2, TYPE_CPE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3 },
},
{
.layout = AV_CH_LAYOUT_QUAD,
.num_ele = { 1, 0, 1, 0 },
.pairing = { { 1 }, { 0 }, { 1 }, },
.index = { { 0 }, { 0 }, { 1 } },
.config_map = { 2, TYPE_CPE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3 },
},
{
.layout = AV_CH_LAYOUT_5POINT0,
.num_ele = { 2, 1, 0, 0 },
.pairing = { { 1, 0 }, { 1 }, },
.index = { { 0, 0 }, { 1 } },
.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4 },
},
{
.layout = AV_CH_LAYOUT_5POINT1,
.num_ele = { 2, 1, 1, 0 },
.pairing = { { 1, 0 }, { 0 }, { 1 }, },
.index = { { 0, 0 }, { 1 }, { 1 } },
.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4, 5 },
},
{
.layout = AV_CH_LAYOUT_5POINT0_BACK,
.num_ele = { 2, 0, 1, 0 },
.pairing = { { 1, 0 }, { 0 }, { 1 } },
.index = { { 0, 0 }, { 0 }, { 1 } },
.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4 },
},
{
.layout = AV_CH_LAYOUT_5POINT1_BACK,
.num_ele = { 2, 1, 1, 0 },
.pairing = { { 1, 0 }, { 0 }, { 1 }, },
.index = { { 0, 0 }, { 1 }, { 1 } },
.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4, 5 },
},
{
.layout = AV_CH_LAYOUT_6POINT0,
.num_ele = { 2, 1, 1, 0 },
.pairing = { { 1, 0 }, { 1 }, { 0 }, },
.index = { { 0, 0 }, { 1 }, { 1 } },
.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5 },
},
{
.layout = AV_CH_LAYOUT_6POINT0_FRONT,
.num_ele = { 2, 1, 0, 0 },
.pairing = { { 1, 1 }, { 1 } },
.index = { { 1, 0 }, { 2 }, },
.config_map = { 3, TYPE_CPE, TYPE_CPE, TYPE_CPE, },
.reorder_map = { 0, 1, 2, 3, 4, 5 },
},
{
.layout = AV_CH_LAYOUT_HEXAGONAL,
.num_ele = { 2, 0, 2, 0 },
.pairing = { { 1, 0 },{ 0 },{ 1, 0 }, },
.index = { { 0, 0 },{ 0 },{ 1, 1 } },
.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, },
.reorder_map = { 0, 1, 2, 3, 4, 5 },
},
{
.layout = AV_CH_LAYOUT_6POINT1,
.num_ele = { 2, 1, 2, 0 },
.pairing = { { 1, 0 },{ 0 },{ 1, 0 }, },
.index = { { 0, 0 },{ 1 },{ 1, 2 } },
.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
},
{
.layout = AV_CH_LAYOUT_6POINT1_BACK,
.num_ele = { 2, 1, 2, 0 },
.pairing = { { 1, 0 }, { 0 }, { 1, 0 }, },
.index = { { 0, 0 }, { 1 }, { 1, 2 } },
.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
},
{
.layout = AV_CH_LAYOUT_6POINT1_FRONT,
.num_ele = { 2, 1, 2, 0 },
.pairing = { { 1, 0 }, { 0 }, { 1, 0 }, },
.index = { { 0, 0 }, { 1 }, { 1, 2 } },
.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
},
{
.layout = AV_CH_LAYOUT_7POINT0,
.num_ele = { 2, 1, 1, 0 },
.pairing = { { 1, 0 }, { 1 }, { 1 }, },
.index = { { 0, 0 }, { 1 }, { 2 }, },
.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
},
{
.layout = AV_CH_LAYOUT_7POINT0_FRONT,
.num_ele = { 2, 1, 1, 0 },
.pairing = { { 1, 0 }, { 1 }, { 1 }, },
.index = { { 0, 0 }, { 1 }, { 2 }, },
.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
},
{
.layout = AV_CH_LAYOUT_7POINT1,
.num_ele = { 2, 1, 2, 0 },
.pairing = { { 1, 0 }, { 0 }, { 1, 1 }, },
.index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
},
{
.layout = AV_CH_LAYOUT_7POINT1_WIDE,
.num_ele = { 2, 1, 2, 0 },
.pairing = { { 1, 0 }, { 0 },{ 1, 1 }, },
.index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
},
{
.layout = AV_CH_LAYOUT_7POINT1_WIDE_BACK,
.num_ele = { 2, 1, 2, 0 },
.pairing = { { 1, 0 }, { 0 }, { 1, 1 }, },
.index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
},
{
.layout = AV_CH_LAYOUT_OCTAGONAL,
.num_ele = { 2, 1, 2, 0 },
.pairing = { { 1, 0 }, { 1 }, { 1, 0 }, },
.index = { { 0, 0 }, { 1 }, { 2, 1 } },
.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
},
{ /* Meant for order 2/mixed ambisonics */
.layout = AV_CH_LAYOUT_OCTAGONAL | AV_CH_TOP_CENTER,
.num_ele = { 2, 2, 2, 0 },
.pairing = { { 1, 0 }, { 1, 0 }, { 1, 0 }, },
.index = { { 0, 0 }, { 1, 1 }, { 2, 2 } },
.config_map = { 6, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8 },
},
{ /* Meant for order 2/mixed ambisonics */
.layout = AV_CH_LAYOUT_6POINT0_FRONT | AV_CH_BACK_CENTER |
AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT | AV_CH_TOP_CENTER,
.num_ele = { 2, 2, 2, 0 },
.pairing = { { 1, 1 }, { 1, 0 }, { 1, 0 }, },
.index = { { 0, 1 }, { 2, 0 }, { 3, 1 } },
.config_map = { 6, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 },
},
{
.layout = AV_CH_LAYOUT_HEXADECAGONAL,
.num_ele = { 4, 2, 4, 0 },
.pairing = { { 1, 0, 1, 0 }, { 1, 1 }, { 1, 0, 1, 0 }, },
.index = { { 0, 0, 1, 1 }, { 2, 3 }, { 4, 2, 5, 3 } },
.config_map = { 10, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15 },
},
};
/**
* AAC encoder context
*/
@@ -99,12 +380,15 @@ typedef struct AACEncContext {
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
AVFloatDSPContext *fdsp;
float *planar_samples[8]; ///< saved preprocessed input
AACPCEInfo pce; ///< PCE data, if needed
float *planar_samples[16]; ///< saved preprocessed input
int profile; ///< copied from avctx
int needs_pce; ///< flag for non-standard layout
LPCContext lpc; ///< used by TNS
int samplerate_index; ///< MPEG-4 samplerate index
int channels; ///< channel count
const uint8_t *reorder_map; ///< lavc to aac reorder map
const uint8_t *chan_map; ///< channel configuration map
ChannelElement *cpe; ///< channel elements
libavcodec/aacenctab.h
+12 -1
View File
@@ -36,13 +36,24 @@
/** Total number of codebooks, including special ones **/
#define CB_TOT_ALL 15
#define AAC_MAX_CHANNELS 8
#define AAC_MAX_CHANNELS 16
extern const uint8_t *ff_aac_swb_size_1024[];
extern const int ff_aac_swb_size_1024_len;
extern const uint8_t *ff_aac_swb_size_128[];
extern const int ff_aac_swb_size_128_len;
/* Supported layouts without using a PCE */
static const int64_t aac_normal_chan_layouts[7] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
AV_CH_LAYOUT_7POINT1,
};
/** default channel configurations */
static const uint8_t aac_chan_configs[AAC_MAX_CHANNELS][6] = {
{1, TYPE_SCE}, // 1 channel - single channel element
libavcodec/aacpsdsp_template.c
+4 -4
View File
@@ -130,12 +130,12 @@ static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
INTFLOAT apd_im = in_im;
in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
link_delay_im, fractional_delay_im);
in_re -= a_re;
in_re -= (UINTFLOAT)a_re;
in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
link_delay_im, fractional_delay_re);
in_im -= a_im;
ap_delay[m][n+5][0] = apd_re + AAC_MUL31(ag[m], in_re);
ap_delay[m][n+5][1] = apd_im + AAC_MUL31(ag[m], in_im);
in_im -= (UINTFLOAT)a_im;
ap_delay[m][n+5][0] = apd_re + (UINTFLOAT)AAC_MUL31(ag[m], in_re);
ap_delay[m][n+5][1] = apd_im + (UINTFLOAT)AAC_MUL31(ag[m], in_im);
}
out[n][0] = AAC_MUL16(transient_gain[n], in_re);
out[n][1] = AAC_MUL16(transient_gain[n], in_im);

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