jellyfin/jellyfin-ffmpeg
1
0
Fork 0
mirror of https://github.com/jellyfin/jellyfin-ffmpeg.git synced 2026-09-03 05:09:58 +03:00
Code Issues Packages Projects Releases Wiki Activity

New upstream version 5.0.1

Browse Source
This commit is contained in:
nyanmisaka
2022-04-27 20:24:39 +08:00
parent 0dc42ed11e
commit b2af5b16fa
2502 changed files with 116308 additions and 53581 deletions
Download Patch File Download Diff File Expand all files Collapse all files
libavcodec/aacenc.c
+9 -7
View File
@@ -30,10 +30,12 @@
***********************************/
#include <float.h>
#include "libavutil/channel_layout.h"
#include "libavutil/libm.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "encode.h"
#include "put_bits.h"
#include "internal.h"
#include "mpeg4audio.h"
@@ -116,7 +118,7 @@ static int put_audio_specific_config(AVCodecContext *avctx)
put_bits(&pb, 5, AOT_SBR);
put_bits(&pb, 1, 0);
flush_put_bits(&pb);
avctx->extradata_size = put_bits_count(&pb) >> 3;
avctx->extradata_size = put_bytes_output(&pb);
return 0;
}
@@ -676,7 +678,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
start_ch += chans;
}
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
if ((ret = ff_alloc_packet(avctx, avpkt, 8192 * s->channels)) < 0)
return ret;
frame_bits = its = 0;
do {
@@ -882,6 +884,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
flush_put_bits(&s->pb);
s->last_frame_pb_count = put_bits_count(&s->pb);
avpkt->size = put_bytes_output(&s->pb);
s->lambda_sum += s->lambda;
s->lambda_count++;
@@ -889,7 +892,6 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = put_bits_count(&s->pb) >> 3;
*got_packet_ptr = 1;
return 0;
}
@@ -996,7 +998,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
/* Samplerate */
for (i = 0; i < 16; i++)
if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
break;
s->samplerate_index = i;
ERROR_IF(s->samplerate_index == 16 ||
@@ -1104,7 +1106,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
#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_FAST}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
{"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 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", "Default fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
@@ -1131,7 +1133,7 @@ static const AVCodecDefault aac_encode_defaults[] = {
{ NULL }
};
AVCodec ff_aac_encoder = {
const AVCodec ff_aac_encoder = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
@@ -1141,7 +1143,7 @@ AVCodec ff_aac_encoder = {
.encode2 = aac_encode_frame,
.close = aac_encode_end,
.defaults = aac_encode_defaults,
.supported_samplerates = mpeg4audio_sample_rates,
.supported_samplerates = ff_mpeg4audio_sample_rates,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,