Commit 9a71d362 authored by Justin Ruggles's avatar Justin Ruggles

avconv: deprecate the -vol option

Remove the code for volume scaling in avconv.c and instead auto-insert a
volume filter into the beginning of the filter chain.
parent b30a3633
......@@ -1081,7 +1081,6 @@ static int decode_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()))
......@@ -1106,64 +1105,6 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
pkt->pts = AV_NOPTS_VALUE;
}
// 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(1);
}
}
rate_emu_sleep(ist);
resample_changed = ist->resample_sample_fmt != decoded_frame->format ||
......
......@@ -452,6 +452,29 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
first_filter = async;
pad_idx = 0;
}
if (audio_volume != 256) {
AVFilterContext *volume;
av_log(NULL, AV_LOG_WARNING, "-vol has been deprecated. Use the volume "
"audio filter instead.\n");
snprintf(args, sizeof(args), "volume=%f", audio_volume / 256.0);
snprintf(name, sizeof(name), "graph %d volume for input stream %d:%d",
fg->index, ist->file_index, ist->st->index);
ret = avfilter_graph_create_filter(&volume,
avfilter_get_by_name("volume"),
name, args, NULL, fg->graph);
if (ret < 0)
return ret;
ret = avfilter_link(volume, 0, first_filter, pad_idx);
if (ret < 0)
return ret;
first_filter = volume;
pad_idx = 0;
}
if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
return ret;
......
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