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.

1 files changed, 0 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         ||