eden-emu/FFmpeg
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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  FATE: use updated reference for aac-latm_stereo_to_51
  avconv: use libavresample
  Add libavresample
  FATE: avoid channel mixing in lavf-dv_fmt

Conflicts:
	Changelog
	Makefile
	cmdutils.c
	configure
	doc/APIchanges
	ffmpeg.c
	tests/lavf-regression.sh
	tests/ref/lavf/dv_fmt
	tests/ref/seek/lavf_dv

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-04-25 22:01:59 +02:00
commit 3ead79eaa3
33 changed files with 3875 additions and 27 deletions
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55
ffmpeg.c
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@ -36,7 +36,6 @@
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
#include "libavutil/opt.h"
#include "libavcodec/audioconvert.h"
#include "libavutil/audioconvert.h"
#include "libavutil/parseutils.h"
#include "libavutil/samplefmt.h"
@ -300,6 +299,7 @@ typedef struct OutputStream {
int audio_channels_mapped; ///< number of channels in audio_channels_map
int resample_sample_fmt;
int resample_channels;
uint64_t resample_channel_layout;
int resample_sample_rate;
float rematrix_volume;
AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
@ -1525,7 +1525,7 @@ static int encode_audio_frame(AVFormatContext *s, OutputStream *ost,
}
static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
int nb_samples)
int nb_samples, int *buf_linesize)
{
int64_t audio_buf_samples;
int audio_buf_size;
@ -1538,7 +1538,7 @@ static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
if (audio_buf_samples > INT_MAX)
return AVERROR(EINVAL);
audio_buf_size = av_samples_get_buffer_size(NULL, enc->channels,
audio_buf_size = av_samples_get_buffer_size(buf_linesize, enc->channels,
audio_buf_samples,
enc->sample_fmt, 0);
if (audio_buf_size < 0)
@ -1557,7 +1557,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
uint8_t *buftmp;
int64_t size_out;
int frame_bytes, resample_changed;
int frame_bytes, resample_changed, ret;
AVCodecContext *enc = ost->st->codec;
AVCodecContext *dec = ist->st->codec;
int osize = av_get_bytes_per_sample(enc->sample_fmt);
@ -1566,37 +1566,46 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
int size = decoded_frame->nb_samples * dec->channels * isize;
int planes = av_sample_fmt_is_planar(dec->sample_fmt) ? dec->channels : 1;
int i;
int out_linesize = 0;
int buf_linesize = decoded_frame->linesize[0];
av_assert0(planes <= AV_NUM_DATA_POINTERS);
for(i=0; i<planes; i++)
buf[i]= decoded_frame->data[i];
get_default_channel_layouts(ost, ist);
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples) < 0) {
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples, &out_linesize) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
exit_program(1);
}
if (enc->channels != dec->channels
|| enc->sample_fmt != dec->sample_fmt
|| enc->sample_rate!= dec->sample_rate
)
if (audio_sync_method > 1 ||
enc->channels != dec->channels ||
enc->channel_layout != dec->channel_layout ||
enc->sample_rate != dec->sample_rate ||
dec->sample_fmt != enc->sample_fmt)
ost->audio_resample = 1;
resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
ost->resample_channels != dec->channels ||
ost->resample_channel_layout != dec->channel_layout ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->swr) || resample_changed || ost->audio_channels_mapped) {
if (resample_changed) {
av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d chl:0x%"PRIx64" to rate:%d fmt:%s ch:%d chl:0x%"PRIx64"\n",
ist->file_index, ist->st->index,
ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), ost->resample_channels,
dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels);
ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt),
ost->resample_channels, ost->resample_channel_layout,
dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt),
dec->channels, dec->channel_layout);
ost->resample_sample_fmt = dec->sample_fmt;
ost->resample_channels = dec->channels;
ost->resample_channel_layout = dec->channel_layout;
ost->resample_sample_rate = dec->sample_rate;
swr_free(&ost->swr);
}
@ -1604,6 +1613,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
if (audio_sync_method <= 1 && !ost->audio_channels_mapped &&
ost->resample_sample_fmt == enc->sample_fmt &&
ost->resample_channels == enc->channels &&
ost->resample_channel_layout == enc->channel_layout &&
ost->resample_sample_rate == enc->sample_rate) {
//ost->swr = NULL;
ost->audio_resample = 0;
@ -1673,7 +1683,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
exit_program(1);
}
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta) < 0) {
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta, &out_linesize) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
exit_program(1);
}
@ -1686,11 +1696,11 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
buf[i] = t;
}
size += byte_delta;
buf_linesize = allocated_async_buf_size;
av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", idelta);
}
} else if (audio_sync_method > 1) {
int comp = av_clip(delta, -audio_sync_method, audio_sync_method);
av_assert0(ost->audio_resample);
av_log(NULL, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n",
delta, comp, enc->sample_rate);
// fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(ost->fifo)/(ost->st->codec->channels * 2));
@ -1703,8 +1713,10 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
if (ost->audio_resample || ost->audio_channels_mapped) {
buftmp = audio_buf;
size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp}, allocated_audio_buf_size / (enc->channels * osize),
buf, size / (dec->channels * isize));
size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp},
allocated_audio_buf_size / (enc->channels * osize),
buf,
size / (dec->channels * isize));
if (size_out < 0) {
av_log(NULL, AV_LOG_FATAL, "swr_convert failed\n");
exit_program(1);
@ -3078,6 +3090,7 @@ static int transcode_init(void)
if (!ost->fifo) {
return AVERROR(ENOMEM);
}
if (!codec->sample_rate)
codec->sample_rate = icodec->sample_rate;
choose_sample_rate(ost->st, ost->enc);
@ -3110,13 +3123,15 @@ static int transcode_init(void)
if (av_get_channel_layout_nb_channels(codec->channel_layout) != codec->channels)
codec->channel_layout = 0;
ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
ost->audio_resample |= codec->sample_fmt != icodec->sample_fmt
|| codec->channel_layout != icodec->channel_layout;
icodec->request_channels = codec->channels;
// ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
// ost->audio_resample |= codec->sample_fmt != icodec->sample_fmt
// || codec->channel_layout != icodec->channel_layout;
icodec->request_channels = codec-> channels;
ost->resample_sample_fmt = icodec->sample_fmt;
ost->resample_sample_rate = icodec->sample_rate;
ost->resample_channels = icodec->channels;
ost->resample_channel_layout = icodec->channel_layout;
break;
case AVMEDIA_TYPE_VIDEO:
if (!ost->filter) {