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authorJustin Ruggles <justin.ruggles@gmail.com>2012-10-02 16:08:20 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2012-12-05 11:23:37 -0500
commit9a71d362a6b77ec215045e5a9d6bdfc427e37ce4 (patch)
treed5f7aaa3856f071b51f7cb2fefcb00881d6c69e5
parentb30a363331ac79331c1d002992689b5ff35bf813 (diff)
downloadffmpeg-9a71d362a6b77ec215045e5a9d6bdfc427e37ce4.tar.gz
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.
-rw-r--r--avconv.c59
-rw-r--r--avconv_filter.c23
2 files changed, 23 insertions, 59 deletions
diff --git a/avconv.c b/avconv.c
index b2e1f045dc..444e74d8d9 100644
--- a/avconv.c
+++ b/avconv.c
@@ -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 ||
diff --git a/avconv_filter.c b/avconv_filter.c
index 8f430b0be4..e9412abc96 100644
--- a/avconv_filter.c
+++ b/avconv_filter.c
@@ -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;