ffmpeg: add -map_channel option.

Based on an initial work by Baptiste Coudurier.
This commit is contained in:
Clément Bœsch 2011-11-04 18:54:01 +01:00
parent 682e0eaf14
commit f5cd136f2b
15 changed files with 228 additions and 22 deletions

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@ -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:

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@ -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.

123
ffmpeg.c
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@ -99,6 +99,11 @@ typedef struct StreamMap {
int sync_stream_index;
} StreamMap;
typedef struct {
int file_idx, stream_idx, channel_idx; // input
int ofile_idx, ostream_idx; // output
} AudioChannelMap;
/**
* select an input file for an output file
*/
@ -231,6 +236,8 @@ typedef struct OutputStream {
/* audio only */
int audio_resample;
int audio_channels_map[SWR_CH_MAX]; ///< list of the channels id to pick from the source stream
int audio_channels_mapped; ///< number of channels in audio_channels_map
int resample_sample_fmt;
int resample_channels;
int resample_sample_rate;
@ -313,6 +320,8 @@ typedef struct OptionsContext {
/* output options */
StreamMap *stream_maps;
int nb_stream_maps;
AudioChannelMap *audio_channel_maps; ///< one info entry per -map_channel
int nb_audio_channel_maps; ///< number of (valid) -map_channel settings
/* first item specifies output metadata, second is input */
MetadataMap (*meta_data_maps)[2];
int nb_meta_data_maps;
@ -409,6 +418,7 @@ static void reset_options(OptionsContext *o, int is_input)
}
av_freep(&o->stream_maps);
av_freep(&o->audio_channel_maps);
av_freep(&o->meta_data_maps);
av_freep(&o->streamid_map);
@ -857,7 +867,7 @@ need_realloc:
ost->resample_channels != dec->channels ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->swr) || resample_changed) {
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",
ist->file_index, ist->st->index,
@ -869,7 +879,7 @@ need_realloc:
swr_free(&ost->swr);
}
/* if audio_sync_method is >1 the resampler is needed for audio drift compensation */
if (audio_sync_method <= 1 &&
if (audio_sync_method <= 1 && !ost->audio_channels_mapped &&
ost->resample_sample_fmt == enc->sample_fmt &&
ost->resample_channels == enc->channels &&
ost->resample_sample_rate == enc->sample_rate) {
@ -879,8 +889,13 @@ need_realloc:
ost->swr = swr_alloc2(ost->swr,
enc->channel_layout, enc->sample_fmt, enc->sample_rate,
dec->channel_layout, dec->sample_fmt, dec->sample_rate,
ost->audio_channels_mapped ? ost->audio_channels_map : NULL,
0, NULL);
av_set_double(ost->swr, "rmvol", ost->rematrix_volume);
if (ost->audio_channels_mapped) {
av_set_int(ost->swr, "icl", av_get_default_channel_layout(ost->audio_channels_mapped));
av_set_int(ost->swr, "uch", ost->audio_channels_mapped);
}
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);
@ -2176,7 +2191,23 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
if (codec->sample_fmt == AV_SAMPLE_FMT_NONE)
codec->sample_fmt = icodec->sample_fmt;
choose_sample_fmt(ost->st, ost->enc);
if (!codec->channels) {
if (ost->audio_channels_mapped) {
/* the requested output channel is set to the number of
* -map_channel only if no -ac are specified */
if (!codec->channels) {
codec->channels = ost->audio_channels_mapped;
codec->channel_layout = av_get_default_channel_layout(codec->channels);
if (!codec->channel_layout) {
av_log(NULL, AV_LOG_FATAL, "Unable to find an appropriate channel layout for requested number of channel\n");
exit_program(1);
}
}
/* fill unused channel mapping with -1 (which means a muted
* channel in case the number of output channels is bigger
* than the number of mapped channel) */
for (j = ost->audio_channels_mapped; j < FF_ARRAY_ELEMS(ost->audio_channels_map); j++)
ost->audio_channels_map[j] = -1;
} else if (!codec->channels) {
codec->channels = icodec->channels;
codec->channel_layout = icodec->channel_layout;
}
@ -2388,6 +2419,15 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
input_streams[ost->source_index].st->index,
ost->file_index,
ost->index);
if (ost->audio_channels_mapped) {
av_log(NULL, AV_LOG_INFO, " [ch:");
for (j = 0; j < ost->audio_channels_mapped; j++)
if (ost->audio_channels_map[j] == -1)
av_log(NULL, AV_LOG_INFO, " M");
else
av_log(NULL, AV_LOG_INFO, " %d", ost->audio_channels_map[j]);
av_log(NULL, AV_LOG_INFO, "]");
}
if (ost->sync_ist != &input_streams[ost->source_index])
av_log(NULL, AV_LOG_INFO, " [sync #%d.%d]",
ost->sync_ist->file_index,
@ -2899,6 +2939,66 @@ static int opt_attach(OptionsContext *o, const char *opt, const char *arg)
return 0;
}
static int opt_map_channel(OptionsContext *o, const char *opt, const char *arg)
{
int n;
AVStream *st;
AudioChannelMap *m;
o->audio_channel_maps =
grow_array(o->audio_channel_maps, sizeof(*o->audio_channel_maps),
&o->nb_audio_channel_maps, o->nb_audio_channel_maps + 1);
m = &o->audio_channel_maps[o->nb_audio_channel_maps - 1];
/* muted channel syntax */
n = sscanf(arg, "%d:%d.%d", &m->channel_idx, &m->ofile_idx, &m->ostream_idx);
if ((n == 1 || n == 3) && m->channel_idx == -1) {
m->file_idx = m->stream_idx = -1;
if (n == 1)
m->ofile_idx = m->ostream_idx = -1;
return 0;
}
/* normal syntax */
n = sscanf(arg, "%d.%d.%d:%d.%d",
&m->file_idx, &m->stream_idx, &m->channel_idx,
&m->ofile_idx, &m->ostream_idx);
if (n != 3 && n != 5) {
av_log(NULL, AV_LOG_FATAL, "Syntax error, mapchan usage: "
"[file.stream.channel|-1][:syncfile:syncstream]\n");
exit_program(1);
}
if (n != 5) // only file.stream.channel specified
m->ofile_idx = m->ostream_idx = -1;
/* check input */
if (m->file_idx < 0 || m->file_idx >= nb_input_files) {
av_log(NULL, AV_LOG_FATAL, "mapchan: invalid input file index: %d\n",
m->file_idx);
exit_program(1);
}
if (m->stream_idx < 0 ||
m->stream_idx >= input_files[m->file_idx].nb_streams) {
av_log(NULL, AV_LOG_FATAL, "mapchan: invalid input file stream index #%d.%d\n",
m->file_idx, m->stream_idx);
exit_program(1);
}
st = input_files[m->file_idx].ctx->streams[m->stream_idx];
if (st->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
av_log(NULL, AV_LOG_FATAL, "mapchan: stream #%d.%d is not an audio stream.\n",
m->file_idx, m->stream_idx);
exit_program(1);
}
if (m->channel_idx < 0 || m->channel_idx >= st->codec->channels) {
av_log(NULL, AV_LOG_FATAL, "mapchan: invalid audio channel #%d.%d.%d\n",
m->file_idx, m->stream_idx, m->channel_idx);
exit_program(1);
}
return 0;
}
static void parse_meta_type(char *arg, char *type, int *index)
{
if (*arg) {
@ -3588,6 +3688,7 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc)
static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc)
{
int n;
AVStream *st;
OutputStream *ost;
AVCodecContext *audio_enc;
@ -3616,6 +3717,21 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc)
MATCH_PER_STREAM_OPT(rematrix_volume, f, ost->rematrix_volume, oc, st);
}
/* check for channel mapping for this audio stream */
for (n = 0; n < o->nb_audio_channel_maps; n++) {
AudioChannelMap *map = &o->audio_channel_maps[n];
InputStream *ist = &input_streams[ost->source_index];
if ((map->channel_idx == -1 || (ist->file_index == map->file_idx && ist->st->index == map->stream_idx)) &&
(map->ofile_idx == -1 || ost->file_index == map->ofile_idx) &&
(map->ostream_idx == -1 || ost->st->index == map->ostream_idx)) {
if (ost->audio_channels_mapped < FF_ARRAY_ELEMS(ost->audio_channels_map))
ost->audio_channels_map[ost->audio_channels_mapped++] = map->channel_idx;
else
av_log(NULL, AV_LOG_FATAL, "Max channel mapping for output %d.%d reached\n",
ost->file_index, ost->st->index);
}
}
return ost;
}
@ -4430,6 +4546,7 @@ static const OptionDef options[] = {
{ "codec", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(codec_names)}, "codec name", "codec" },
{ "pre", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(presets)}, "preset name", "preset" },
{ "map", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map}, "set input stream mapping", "file.stream[:syncfile.syncstream]" },
{ "map_channel", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map_channel}, "map an audio channel from one stream to another", "file.stream.channel[:syncfile.syncstream]" },
{ "map_meta_data", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map_meta_data}, "DEPRECATED set meta data information of outfile from infile",
"outfile[,metadata]:infile[,metadata]" },
{ "map_metadata", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map_metadata}, "set metadata information of outfile from infile",

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@ -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,

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@ -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)

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@ -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.

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@ -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{

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@ -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.

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@ -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;

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@ -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)

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@ -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

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@ -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

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