diff --git a/avconv.c b/avconv.c index a9cd9bf481..fa39e6b052 100644 --- a/avconv.c +++ b/avconv.c @@ -1664,12 +1664,59 @@ static int output_packet(InputStream *ist, int ist_index, // preprocess audio (volume) if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { if (audio_volume != 256) { + switch (ist->st->codec->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: + { short *volp; volp = samples; for(i=0;i<(decoded_data_size / sizeof(short));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); + } } }