static int transcode_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; if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame())) return AVERROR(ENOMEM); else avcodec_get_frame_defaults(ist->decoded_frame); decoded_frame = ist->decoded_frame; ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt); if (ret < 0) { return ret; } if (!*got_output) { /* no audio frame */ return ret; } /* if the decoder provides a pts, use it instead of the last packet pts. the decoder could be delaying output by a packet or more. */ if (decoded_frame->pts != AV_NOPTS_VALUE) ist->next_dts = decoded_frame->pts; /* increment next_dts to use for the case where the input stream does not have timestamps or there are multiple frames in the packet */ ist->next_dts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) / avctx->sample_rate; // 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_program(1); } } rate_emu_sleep(ist); for (i = 0; i < nb_output_streams; i++) { OutputStream *ost = output_streams[i]; if (!check_output_constraints(ist, ost) || !ost->encoding_needed) continue; do_audio_out(output_files[ost->file_index]->ctx, ost, ist, decoded_frame); } return ret; }