generic-library/ffmpeg
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Add libswresample.

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Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2011-09-19 06:13:30 +02:00
parent 53e37840bf
commit b5875b9111
18 changed files with 1634 additions and 67 deletions
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83
ffmpeg.c
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@@ -45,6 +45,7 @@
#include "libavutil/avstring.h"
#include "libavutil/libm.h"
#include "libavformat/os_support.h"
#include "libswresample/swresample.h"
#include "libavformat/ffm.h" // not public API
@@ -229,15 +230,14 @@ typedef struct OutputStream {
/* audio only */
int audio_resample;
ReSampleContext *resample; /* for audio resampling */
int resample_sample_fmt;
int resample_channels;
int resample_sample_rate;
int reformat_pair;
AVAudioConvert *reformat_ctx;
AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
FILE *logfile;
struct SwrContext *swr;
#if CONFIG_AVFILTER
AVFilterContext *output_video_filter;
AVFilterContext *input_video_filter;
@@ -843,14 +843,15 @@ need_realloc:
exit_program(1);
}
if (enc->channels != dec->channels)
if (enc->channels != dec->channels
|| enc->sample_fmt != dec->sample_fmt)
ost->audio_resample = 1;
resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
ost->resample_channels != dec->channels ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->resample) || resample_changed) {
if ((ost->audio_resample && !ost->swr) || resample_changed) {
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",
ist->file_index, ist->st->index,
@@ -859,24 +860,29 @@ need_realloc:
ost->resample_sample_fmt = dec->sample_fmt;
ost->resample_channels = dec->channels;
ost->resample_sample_rate = dec->sample_rate;
if (ost->resample)
audio_resample_close(ost->resample);
swr_free(&ost->swr);
}
/* if audio_sync_method is >1 the resampler is needed for audio drift compensation */
if (audio_sync_method <= 1 &&
ost->resample_sample_fmt == enc->sample_fmt &&
ost->resample_channels == enc->channels &&
ost->resample_sample_rate == enc->sample_rate) {
ost->resample = NULL;
//ost->swr = NULL;
ost->audio_resample = 0;
} else {
if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
ost->resample = av_audio_resample_init(enc->channels, dec->channels,
enc->sample_rate, dec->sample_rate,
enc->sample_fmt, dec->sample_fmt,
16, 10, 0, 0.8);
if (!ost->resample) {
ost->swr = swr_alloc2(ost->swr,
enc->channel_layout, enc->sample_fmt, enc->sample_rate,
dec->channel_layout, dec->sample_fmt, dec->sample_rate,
0, NULL);
av_set_int(ost->swr, "ich", dec->channels);
av_set_int(ost->swr, "och", enc->channels);
if(audio_sync_method>1) av_set_int(ost->swr, "flags", SWR_FLAG_RESAMPLE);
if(ost->swr && swr_init(ost->swr) < 0){
fprintf(stderr, "swr_init() failed\n");
swr_free(&ost->swr);
}
if (!ost->swr) {
fprintf(stderr, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
dec->channels, dec->sample_rate,
enc->channels, enc->sample_rate);
@@ -885,21 +891,7 @@ need_realloc:
}
}
#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt &&
MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) {
if (ost->reformat_ctx)
av_audio_convert_free(ost->reformat_ctx);
ost->reformat_ctx = av_audio_convert_alloc(enc->sample_fmt, 1,
dec->sample_fmt, 1, NULL, 0);
if (!ost->reformat_ctx) {
fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
av_get_sample_fmt_name(dec->sample_fmt),
av_get_sample_fmt_name(enc->sample_fmt));
exit_program(1);
}
ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt);
}
av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt);
if(audio_sync_method){
double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts
@@ -941,7 +933,7 @@ need_realloc:
if(verbose > 2)
fprintf(stderr, "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));
av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate);
swr_compensate(ost->swr, comp, enc->sample_rate);
}
}
}else
@@ -950,30 +942,15 @@ need_realloc:
if (ost->audio_resample) {
buftmp = audio_buf;
size_out = audio_resample(ost->resample,
(short *)buftmp, (short *)buf,
size / (dec->channels * isize));
size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp}, audio_buf_size / (enc->channels * osize),
(const uint8_t*[]){buf }, size / (dec->channels * isize));
size_out = size_out * enc->channels * osize;
} else {
buftmp = buf;
size_out = size;
}
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt) {
const void *ibuf[6]= {buftmp};
void *obuf[6]= {audio_buf};
int istride[6]= {isize};
int ostride[6]= {osize};
int len= size_out/istride[0];
if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
printf("av_audio_convert() failed\n");
if (exit_on_error)
exit_program(1);
return;
}
buftmp = audio_buf;
size_out = len*osize;
}
av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt);
/* now encode as many frames as possible */
if (enc->frame_size > 1) {
@@ -2133,7 +2110,6 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
if (!ost->fifo) {
return AVERROR(ENOMEM);
}
ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE);
if (!codec->sample_rate) {
codec->sample_rate = icodec->sample_rate;
}
@@ -2149,6 +2125,8 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
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;
ist->decoding_needed = 1;
ost->encoding_needed = 1;
@@ -2679,10 +2657,7 @@ static int transcode(OutputFile *output_files, int nb_output_files,
av_free(ost->forced_kf_pts);
if (ost->video_resample)
sws_freeContext(ost->img_resample_ctx);
if (ost->resample)
audio_resample_close(ost->resample);
if (ost->reformat_ctx)
av_audio_convert_free(ost->reformat_ctx);
swr_free(&ost->swr);
av_dict_free(&ost->opts);
}
}