mirror of https://git.ffmpeg.org/ffmpeg.git
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.
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parent
b30a363331
commit
9a71d362a6
59
avconv.c
59
avconv.c
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@ -1081,7 +1081,6 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
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{
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AVFrame *decoded_frame;
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AVCodecContext *avctx = ist->st->codec;
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int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
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int i, ret, resample_changed;
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if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
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@ -1106,64 +1105,6 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
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pkt->pts = AV_NOPTS_VALUE;
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}
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// preprocess audio (volume)
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if (audio_volume != 256) {
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int decoded_data_size = decoded_frame->nb_samples * avctx->channels * bps;
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void *samples = decoded_frame->data[0];
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switch (avctx->sample_fmt) {
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case AV_SAMPLE_FMT_U8:
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{
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uint8_t *volp = samples;
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for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
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int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;
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*volp++ = av_clip_uint8(v);
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}
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break;
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}
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case AV_SAMPLE_FMT_S16:
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{
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int16_t *volp = samples;
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for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
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int v = ((*volp) * audio_volume + 128) >> 8;
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*volp++ = av_clip_int16(v);
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}
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break;
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}
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case AV_SAMPLE_FMT_S32:
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{
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int32_t *volp = samples;
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for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
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int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);
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*volp++ = av_clipl_int32(v);
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}
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break;
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}
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case AV_SAMPLE_FMT_FLT:
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{
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float *volp = samples;
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float scale = audio_volume / 256.f;
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for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
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*volp++ *= scale;
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}
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break;
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}
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case AV_SAMPLE_FMT_DBL:
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{
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double *volp = samples;
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double scale = audio_volume / 256.;
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for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
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*volp++ *= scale;
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}
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break;
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}
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default:
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av_log(NULL, AV_LOG_FATAL,
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"Audio volume adjustment on sample format %s is not supported.\n",
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av_get_sample_fmt_name(ist->st->codec->sample_fmt));
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exit(1);
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}
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}
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rate_emu_sleep(ist);
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resample_changed = ist->resample_sample_fmt != decoded_frame->format ||
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@ -452,6 +452,29 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
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first_filter = async;
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pad_idx = 0;
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}
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if (audio_volume != 256) {
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AVFilterContext *volume;
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av_log(NULL, AV_LOG_WARNING, "-vol has been deprecated. Use the volume "
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"audio filter instead.\n");
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snprintf(args, sizeof(args), "volume=%f", audio_volume / 256.0);
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snprintf(name, sizeof(name), "graph %d volume for input stream %d:%d",
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fg->index, ist->file_index, ist->st->index);
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ret = avfilter_graph_create_filter(&volume,
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avfilter_get_by_name("volume"),
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name, args, NULL, fg->graph);
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if (ret < 0)
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return ret;
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ret = avfilter_link(volume, 0, first_filter, pad_idx);
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if (ret < 0)
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return ret;
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first_filter = volume;
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pad_idx = 0;
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}
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if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
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return ret;
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