Commit f5cd136f authored by Clément Bœsch's avatar Clément Bœsch

ffmpeg: add -map_channel option.

Based on an initial work by Baptiste Coudurier.
parent 682e0eaf
......@@ -72,6 +72,7 @@ easier to use. The changes are:
- Prores encoder
- Video Decoder Acceleration (VDA) HWAccel module.
- replacement Indeo 3 decoder
- new ffmpeg option: -map_channel
version 0.8:
......
......@@ -721,6 +721,44 @@ ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
Note that using this option disables the default mappings for this output file.
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
Map an audio channel from a given input to an output. If
@var{output_file_id}.@var{stream_specifier} are not set, the audio channel will
be mapped on all the audio streams.
Using "-1" instead of
@var{input_file_id}.@var{stream_specifier}.@var{channel_id} will map a muted
channel.
For example, assuming @var{INPUT} is a stereo audio file, you can switch the
two audio channels with the following command:
@example
ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT
@end example
If you want to mute the first channel and keep the second:
@example
ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT
@end example
The order of the "-map_channel" option specifies the order of the channels in
the output stream. The output channel layout is guessed from the number of
channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac"
in combination of "-map_channel" makes the channel gain levels to be updated if
channel layouts don't match (for instance two "-map_channel" options and "-ac
6").
You can also extract each channel of an @var{INPUT} to specific outputs; the
following command extract each channel of the audio stream (file 0, stream 0)
to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1}:
@example
ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1
@end example
Note that "-map_channel" is currently limited to the scope of one input for
each output; you can't for example use it to pick multiple input audio files
and mix them into one single output.
@item -map_metadata[:@var{metadata_type}][:@var{index}] @var{infile}[:@var{metadata_type}][:@var{index}] (@emph{output,per-metadata})
Set metadata information of the next output file from @var{infile}. Note that
those are file indices (zero-based), not filenames.
......
This diff is collapsed.
......@@ -2077,7 +2077,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc2(NULL, is->audio_tgt_channel_layout, is->audio_tgt_fmt, is->audio_tgt_freq,
dec_channel_layout, dec->sample_fmt, dec->sample_rate,
0, NULL);
NULL, 0, NULL);
if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
fprintf(stderr, "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
dec->sample_rate,
......
......@@ -35,11 +35,13 @@
struct AVAudioConvert {
int channels;
int fmt_pair;
const int *ch_map;
};
AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int flags)
int channels, const int *ch_map,
int flags)
{
AVAudioConvert *ctx;
ctx = av_malloc(sizeof(AVAudioConvert));
......@@ -47,6 +49,7 @@ AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
return NULL;
ctx->channels = channels;
ctx->fmt_pair = out_fmt + AV_SAMPLE_FMT_NB*in_fmt;
ctx->ch_map = ch_map;
return ctx;
}
......@@ -58,15 +61,17 @@ void swr_audio_convert_free(AVAudioConvert **ctx)
int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData*in, int len)
{
int ch;
const uint8_t null_input[8] = {0};
av_assert0(ctx->channels == out->ch_count);
//FIXME optimize common cases
for(ch=0; ch<ctx->channels; ch++){
const int is= (in ->planar ? 1 : in->ch_count) * in->bps;
const int ich= ctx->ch_map ? ctx->ch_map[ch] : ch;
const int is= ich < 0 ? 0 : (in->planar ? 1 : in->ch_count) * in->bps;
const int os= (out->planar ? 1 :out->ch_count) *out->bps;
const uint8_t *pi= in ->ch[ch];
const uint8_t *pi= ich < 0 ? null_input : in->ch[ich];
uint8_t *po= out->ch[ch];
uint8_t *end= po + os*len;
if(!po)
......
......@@ -42,11 +42,14 @@ typedef struct AVAudioConvert AVAudioConvert;
* @param in_fmt Input sample format
* @param channels Number of channels
* @param flags See AV_CPU_FLAG_xx
* @param ch_map list of the channels id to pick from the source stream, NULL
* if all channels must be selected
* @return NULL on error
*/
AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int flags);
int channels, const int *ch_map,
int flags);
/**
* Free audio sample format converter context.
......
......@@ -38,6 +38,7 @@
static const AVOption options[]={
{"ich", "input channel count", OFFSET( in.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
{"och", "output channel count", OFFSET(out.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
{"uch", "used channel count", OFFSET(used_ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
{"isr", "input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
{"osr", "output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
//{"ip" , "input planar" , OFFSET( in.planar ), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
......@@ -76,7 +77,7 @@ SwrContext *swr_alloc(void){
SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx){
const int *channel_map, int log_offset, void *log_ctx){
if(!s) s= swr_alloc();
if(!s) return NULL;
......@@ -90,9 +91,11 @@ SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampl
av_set_int(s, "isf", in_sample_fmt);
av_set_int(s, "isr", in_sample_rate);
s->channel_map = channel_map;
s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
s->int_sample_fmt = AV_SAMPLE_FMT_S16;
s->used_ch_count= s-> in.ch_count;
return s;
}
......@@ -167,13 +170,16 @@ int swr_init(SwrContext *s){
return -1;
}
if(s-> in.ch_count && s-> in_ch_layout && s->in.ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than there actually is, ignoring layout\n");
if(!s->used_ch_count)
s->used_ch_count= s->in.ch_count;
if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
s-> in_ch_layout= 0;
}
if(!s-> in_ch_layout)
s-> in_ch_layout= av_get_default_channel_layout(s->in.ch_count);
s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
if(!s->out_ch_layout)
s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
......@@ -182,10 +188,13 @@ int swr_init(SwrContext *s){
#define RSC 1 //FIXME finetune
if(!s-> in.ch_count)
s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
if(!s->used_ch_count)
s->used_ch_count= s->in.ch_count;
if(!s->out.ch_count)
s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
av_assert0(s-> in.ch_count);
av_assert0(s->used_ch_count);
av_assert0(s->out.ch_count);
s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
......@@ -193,22 +202,27 @@ av_assert0(s->out.ch_count);
s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
if(!s->resample && !s->rematrix){
if(!s->resample && !s->rematrix && !s->channel_map){
s->full_convert = swr_audio_convert_alloc(s->out_sample_fmt,
s-> in_sample_fmt, s-> in.ch_count, 0);
s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
return 0;
}
s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
s-> in_sample_fmt, s-> in.ch_count, 0);
s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
s->int_sample_fmt, s->out.ch_count, 0);
s->int_sample_fmt, s->out.ch_count, NULL, 0);
s->postin= s->in;
s->preout= s->out;
s->midbuf= s->in;
s->in_buffer= s->in;
if(s->channel_map){
s->postin.ch_count=
s->midbuf.ch_count=
s->in_buffer.ch_count= s->used_ch_count;
}
if(!s->resample_first){
s->midbuf.ch_count= s->out.ch_count;
s->in_buffer.ch_count = s->out.ch_count;
......@@ -325,7 +339,7 @@ int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_coun
if((ret=realloc_audio(&s->postin, in_count))<0)
return ret;
if(s->resample_first){
av_assert0(s->midbuf.ch_count == s-> in.ch_count);
av_assert0(s->midbuf.ch_count == s->used_ch_count);
if((ret=realloc_audio(&s->midbuf, out_count))<0)
return ret;
}else{
......
......@@ -25,7 +25,7 @@
#include "libavutil/samplefmt.h"
#define LIBSWRESAMPLE_VERSION_MAJOR 0
#define LIBSWRESAMPLE_VERSION_MINOR 1
#define LIBSWRESAMPLE_VERSION_MINOR 2
#define LIBSWRESAMPLE_VERSION_MICRO 0
#define SWR_CH_MAX 16
......@@ -57,7 +57,7 @@ int swr_init(struct SwrContext *s);
*/
struct SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx);
const int *channel_map, int log_offset, void *log_ctx);
/**
* Free the given SwrContext.
......
......@@ -45,6 +45,8 @@ typedef struct SwrContext { //FIXME find unused fields
int out_sample_rate;
int flags;
float slev, clev, rematrix_volume;
const int *channel_map; ///< channel index (or -1 if muted channel) map
int used_ch_count; ///< number of used channels (mapped channel count if channel_map, otherwise in.ch_count)
//below are private
int int_bps;
......
......@@ -131,9 +131,11 @@ int main(int argc, char **argv){
in_sample_rate, out_sample_rate,
av_get_sample_fmt_name(in_sample_fmt), av_get_sample_fmt_name(out_sample_fmt));
forw_ctx = swr_alloc2(forw_ctx, out_ch_layout, out_sample_fmt+planar_out, out_sample_rate,
in_ch_layout, in_sample_fmt+planar_in , in_sample_rate, 0, 0);
backw_ctx = swr_alloc2(backw_ctx,in_ch_layout, in_sample_fmt, in_sample_rate,
out_ch_layout, out_sample_fmt+planar_out, out_sample_rate, 0, 0);
in_ch_layout, in_sample_fmt+planar_in , in_sample_rate,
NULL, 0, 0);
backw_ctx = swr_alloc2(backw_ctx,in_ch_layout, in_sample_fmt, in_sample_rate,
out_ch_layout, out_sample_fmt+planar_out, out_sample_rate,
NULL, 0, 0);
if(swr_init( forw_ctx) < 0)
fprintf(stderr, "swr_init(->) failed\n");
if(swr_init(backw_ctx) < 0)
......
......@@ -30,6 +30,14 @@ tests/data/asynth-16000-1.sw: tests/audiogen$(HOSTEXESUF)
@mkdir -p tests/data
$(M)./$< $@ 16000 1
tests/data/mapchan-6ch.sw: tests/audiogen$(HOSTEXESUF)
@mkdir -p tests/data
$(M)./$< $@ 22050 6
tests/data/mapchan-mono.sw: tests/audiogen$(HOSTEXESUF)
@mkdir -p tests/data
$(M)./$< $@ 22050 1
tests/data/asynth%.sw tests/vsynth%/00.pgm: TAG = GEN
include $(SRC_PATH)/tests/fate.mak
......
......@@ -137,6 +137,18 @@ FATE_TESTS += fate-g722enc
fate-g722enc: tests/data/asynth-16000-1.sw
fate-g722enc: CMD = md5 -ar 16000 -ac 1 -f s16le -i $(TARGET_PATH)/tests/data/asynth-16000-1.sw -acodec g722 -ac 1 -f g722
FATE_TESTS += fate-mapchan-6ch-extract-2
fate-mapchan-6ch-extract-2: tests/data/mapchan-6ch.sw
fate-mapchan-6ch-extract-2: CMD = avconv -ar 22050 -ac 6 -i $(TARGET_PATH)/tests/data/mapchan-6ch.sw -map_channel 0.0.0 -f wav md5: -map_channel 0.0.1 -f wav md5:
FATE_TESTS += fate-mapchan-6ch-extract-2-downmix-mono
fate-mapchan-6ch-extract-2-downmix-mono: tests/data/mapchan-6ch.sw
fate-mapchan-6ch-extract-2-downmix-mono: CMD = md5 -ar 22050 -ac 6 -i $(TARGET_PATH)/tests/data/mapchan-6ch.sw -map_channel 0.0.1 -map_channel 0.0.0 -ac 1 -f wav
FATE_TESTS += fate-mapchan-silent-mono
fate-mapchan-silent-mono: tests/data/mapchan-mono.sw
fate-mapchan-silent-mono: CMD = md5 -ar 22050 -ac 1 -i $(TARGET_PATH)/tests/data/mapchan-mono.sw -map_channel -1 -map_channel 0.0.0 -f wav
FATE_TESTS += fate-msmpeg4v1
fate-msmpeg4v1: CMD = framecrc -flags +bitexact -dct fastint -idct simple -i $(SAMPLES)/msmpeg4v1/mpg4.avi -an
......
6f091fe8c0be88c75921731dc9f74314
5c2d162b9024329eb367295d37b8ca0a
4f5148f08587a4b9794aa52aec7852ac
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