408 lines
15 KiB
C
408 lines
15 KiB
C
/*
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/dict.h"
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#include "libavutil/error.h"
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#include "libavutil/log.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "avresample.h"
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#include "audio_data.h"
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#include "internal.h"
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int avresample_open(AVAudioResampleContext *avr)
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{
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int ret;
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/* set channel mixing parameters */
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avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
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if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
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av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
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avr->in_channel_layout);
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return AVERROR(EINVAL);
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}
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avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
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if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
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av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
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avr->out_channel_layout);
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return AVERROR(EINVAL);
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}
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avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
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avr->downmix_needed = avr->in_channels > avr->out_channels;
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avr->upmix_needed = avr->out_channels > avr->in_channels ||
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avr->am->matrix ||
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(avr->out_channels == avr->in_channels &&
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avr->in_channel_layout != avr->out_channel_layout);
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avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
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/* set resampling parameters */
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avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
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avr->force_resampling;
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/* set sample format conversion parameters */
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/* override user-requested internal format to avoid unexpected failures
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TODO: support more internal formats */
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if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
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av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
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avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
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} else if (avr->mixing_needed &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
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av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
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avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
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}
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if (avr->in_channels == 1)
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avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
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if (avr->out_channels == 1)
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avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
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avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
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avr->in_sample_fmt != avr->internal_sample_fmt;
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if (avr->resample_needed || avr->mixing_needed)
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avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
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else
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avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
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/* allocate buffers */
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if (avr->mixing_needed || avr->in_convert_needed) {
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avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
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0, avr->internal_sample_fmt,
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"in_buffer");
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if (!avr->in_buffer) {
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ret = AVERROR(EINVAL);
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goto error;
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}
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}
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if (avr->resample_needed) {
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avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
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0, avr->internal_sample_fmt,
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"resample_out_buffer");
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if (!avr->resample_out_buffer) {
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ret = AVERROR(EINVAL);
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goto error;
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}
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}
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if (avr->out_convert_needed) {
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avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
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avr->out_sample_fmt, "out_buffer");
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if (!avr->out_buffer) {
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ret = AVERROR(EINVAL);
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goto error;
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}
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}
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avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
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1024);
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if (!avr->out_fifo) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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/* setup contexts */
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if (avr->in_convert_needed) {
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avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
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avr->in_sample_fmt, avr->in_channels);
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if (!avr->ac_in) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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}
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if (avr->out_convert_needed) {
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enum AVSampleFormat src_fmt;
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if (avr->in_convert_needed)
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src_fmt = avr->internal_sample_fmt;
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else
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src_fmt = avr->in_sample_fmt;
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avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
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avr->out_channels);
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if (!avr->ac_out) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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}
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if (avr->resample_needed) {
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avr->resample = ff_audio_resample_init(avr);
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if (!avr->resample) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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}
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if (avr->mixing_needed) {
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ret = ff_audio_mix_init(avr);
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if (ret < 0)
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goto error;
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}
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return 0;
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error:
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avresample_close(avr);
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return ret;
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}
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void avresample_close(AVAudioResampleContext *avr)
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{
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ff_audio_data_free(&avr->in_buffer);
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ff_audio_data_free(&avr->resample_out_buffer);
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ff_audio_data_free(&avr->out_buffer);
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av_audio_fifo_free(avr->out_fifo);
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avr->out_fifo = NULL;
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av_freep(&avr->ac_in);
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av_freep(&avr->ac_out);
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ff_audio_resample_free(&avr->resample);
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ff_audio_mix_close(avr->am);
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return;
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}
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void avresample_free(AVAudioResampleContext **avr)
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{
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if (!*avr)
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return;
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avresample_close(*avr);
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av_freep(&(*avr)->am);
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av_opt_free(*avr);
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av_freep(avr);
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}
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static int handle_buffered_output(AVAudioResampleContext *avr,
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AudioData *output, AudioData *converted)
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{
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int ret;
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if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
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(converted && output->allocated_samples < converted->nb_samples)) {
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if (converted) {
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/* if there are any samples in the output FIFO or if the
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user-supplied output buffer is not large enough for all samples,
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we add to the output FIFO */
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av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name);
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ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
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converted->nb_samples);
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if (ret < 0)
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return ret;
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}
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/* if the user specified an output buffer, read samples from the output
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FIFO to the user output */
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if (output && output->allocated_samples > 0) {
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av_dlog(avr, "[FIFO] read from out_fifo to output\n");
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av_dlog(avr, "[end conversion]\n");
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return ff_audio_data_read_from_fifo(avr->out_fifo, output,
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output->allocated_samples);
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}
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} else if (converted) {
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/* copy directly to output if it is large enough or there is not any
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data in the output FIFO */
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av_dlog(avr, "[copy] %s to output\n", converted->name);
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output->nb_samples = 0;
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ret = ff_audio_data_copy(output, converted);
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if (ret < 0)
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return ret;
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av_dlog(avr, "[end conversion]\n");
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return output->nb_samples;
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}
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av_dlog(avr, "[end conversion]\n");
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return 0;
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}
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int avresample_convert(AVAudioResampleContext *avr, void **output,
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int out_plane_size, int out_samples, void **input,
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int in_plane_size, int in_samples)
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{
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AudioData input_buffer;
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AudioData output_buffer;
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AudioData *current_buffer;
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int ret;
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/* reset internal buffers */
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if (avr->in_buffer) {
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avr->in_buffer->nb_samples = 0;
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ff_audio_data_set_channels(avr->in_buffer,
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avr->in_buffer->allocated_channels);
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}
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if (avr->resample_out_buffer) {
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avr->resample_out_buffer->nb_samples = 0;
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ff_audio_data_set_channels(avr->resample_out_buffer,
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avr->resample_out_buffer->allocated_channels);
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}
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if (avr->out_buffer) {
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avr->out_buffer->nb_samples = 0;
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ff_audio_data_set_channels(avr->out_buffer,
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avr->out_buffer->allocated_channels);
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}
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av_dlog(avr, "[start conversion]\n");
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/* initialize output_buffer with output data */
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if (output) {
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ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
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avr->out_channels, out_samples,
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avr->out_sample_fmt, 0, "output");
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if (ret < 0)
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return ret;
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output_buffer.nb_samples = 0;
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}
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if (input) {
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/* initialize input_buffer with input data */
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ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
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avr->in_channels, in_samples,
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avr->in_sample_fmt, 1, "input");
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if (ret < 0)
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return ret;
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current_buffer = &input_buffer;
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if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
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!avr->out_convert_needed && output && out_samples >= in_samples) {
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/* in some rare cases we can copy input to output and upmix
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directly in the output buffer */
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av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
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ret = ff_audio_data_copy(&output_buffer, current_buffer);
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if (ret < 0)
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return ret;
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current_buffer = &output_buffer;
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} else if (avr->mixing_needed || avr->in_convert_needed) {
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/* if needed, copy or convert input to in_buffer, and downmix if
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applicable */
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if (avr->in_convert_needed) {
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ret = ff_audio_data_realloc(avr->in_buffer,
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current_buffer->nb_samples);
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if (ret < 0)
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return ret;
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av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
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ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer,
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current_buffer->nb_samples);
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if (ret < 0)
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return ret;
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} else {
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av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
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ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
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if (ret < 0)
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return ret;
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}
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ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
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if (avr->downmix_needed) {
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av_dlog(avr, "[downmix] in_buffer\n");
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ret = ff_audio_mix(avr->am, avr->in_buffer);
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if (ret < 0)
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return ret;
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}
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current_buffer = avr->in_buffer;
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}
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} else {
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/* flush resampling buffer and/or output FIFO if input is NULL */
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if (!avr->resample_needed)
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return handle_buffered_output(avr, output ? &output_buffer : NULL,
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NULL);
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current_buffer = NULL;
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}
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if (avr->resample_needed) {
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AudioData *resample_out;
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int consumed = 0;
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if (!avr->out_convert_needed && output && out_samples > 0)
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resample_out = &output_buffer;
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else
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resample_out = avr->resample_out_buffer;
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av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
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resample_out->name);
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ret = ff_audio_resample(avr->resample, resample_out,
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current_buffer, &consumed);
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if (ret < 0)
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return ret;
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/* if resampling did not produce any samples, just return 0 */
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if (resample_out->nb_samples == 0) {
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av_dlog(avr, "[end conversion]\n");
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return 0;
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}
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current_buffer = resample_out;
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}
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if (avr->upmix_needed) {
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av_dlog(avr, "[upmix] %s\n", current_buffer->name);
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ret = ff_audio_mix(avr->am, current_buffer);
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if (ret < 0)
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return ret;
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}
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/* if we resampled or upmixed directly to output, return here */
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if (current_buffer == &output_buffer) {
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av_dlog(avr, "[end conversion]\n");
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return current_buffer->nb_samples;
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}
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if (avr->out_convert_needed) {
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if (output && out_samples >= current_buffer->nb_samples) {
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/* convert directly to output */
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av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
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ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer,
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current_buffer->nb_samples);
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if (ret < 0)
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return ret;
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av_dlog(avr, "[end conversion]\n");
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return output_buffer.nb_samples;
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} else {
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ret = ff_audio_data_realloc(avr->out_buffer,
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current_buffer->nb_samples);
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if (ret < 0)
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return ret;
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av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
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ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
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current_buffer, current_buffer->nb_samples);
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if (ret < 0)
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return ret;
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current_buffer = avr->out_buffer;
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}
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}
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return handle_buffered_output(avr, &output_buffer, current_buffer);
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}
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int avresample_available(AVAudioResampleContext *avr)
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{
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return av_audio_fifo_size(avr->out_fifo);
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}
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int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples)
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{
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if (!output)
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return av_audio_fifo_drain(avr->out_fifo, nb_samples);
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return av_audio_fifo_read(avr->out_fifo, output, nb_samples);
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}
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unsigned avresample_version(void)
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{
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return LIBAVRESAMPLE_VERSION_INT;
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}
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const char *avresample_license(void)
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{
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#define LICENSE_PREFIX "libavresample license: "
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return LICENSE_PREFIX LIBAV_LICENSE + sizeof(LICENSE_PREFIX) - 1;
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}
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const char *avresample_configuration(void)
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{
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return LIBAV_CONFIGURATION;
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}
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