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ffmpeg: automatically insert volume filter when -vol is used.

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Deprecate -vol.

Inspired by asyncts auto-insert patch from Anton Khirnov.
This commit is contained in:
Clément Bœsch 2012-05-17 17:27:20 +02:00 committed by Michael Niedermayer
parent 22a3a5ee0c
commit a99a3b1bb3
1 changed files with 21 additions and 59 deletions
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ffmpeg.c
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@ -870,6 +870,27 @@ static int configure_audio_filters(FilterGraph *fg, AVFilterContext **in_filter,
*out_filter = format;
}
if (audio_volume != 256) {
AVFilterContext *volume;
char args[256];
snprintf(args, sizeof(args), "%lf", audio_volume / 256.);
av_log(NULL, AV_LOG_WARNING, "-vol has been deprecated. Used the "
"volume audio filter instead (-af volume=%s).\n", args);
ret = avfilter_graph_create_filter(&volume,
avfilter_get_by_name("volume"),
"volume", args, NULL, fg->graph);
if (ret < 0)
return ret;
ret = avfilter_link(*in_filter, 0, volume, 0);
if (ret < 0)
return ret;
*in_filter = volume;
}
return 0;
}
@ -2357,7 +2378,6 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
{
AVFrame *decoded_frame;
AVCodecContext *avctx = ist->st->codec;
int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
int i, ret, resample_changed;
if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
@ -2409,64 +2429,6 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
avctx->sample_rate;
#endif
// preprocess audio (volume)
if (audio_volume != 256) {
int decoded_data_size = decoded_frame->nb_samples * avctx->channels * bps;
void *samples = decoded_frame->data[0];
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_U8:
{
uint8_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;
*volp++ = av_clip_uint8(v);
}
break;
}
case AV_SAMPLE_FMT_S16:
{
int16_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int v = ((*volp) * audio_volume + 128) >> 8;
*volp++ = av_clip_int16(v);
}
break;
}
case AV_SAMPLE_FMT_S32:
{
int32_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);
*volp++ = av_clipl_int32(v);
}
break;
}
case AV_SAMPLE_FMT_FLT:
{
float *volp = samples;
float scale = audio_volume / 256.f;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
*volp++ *= scale;
}
break;
}
case AV_SAMPLE_FMT_DBL:
{
double *volp = samples;
double scale = audio_volume / 256.;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
*volp++ *= scale;
}
break;
}
default:
av_log(NULL, AV_LOG_FATAL,
"Audio volume adjustment on sample format %s is not supported.\n",
av_get_sample_fmt_name(ist->st->codec->sample_fmt));
exit_program(1);
}
}
rate_emu_sleep(ist);
resample_changed = ist->resample_sample_fmt != decoded_frame->format ||