14758e3211
This allows AudioMix to be treated the same way as other conversion contexts and removes the requirement to allocate it at the same time as the AVAudioResampleContext. The current matrix get/set functions are split between the public interface and AudioMix private functions.
495 lines
18 KiB
C
495 lines
18 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/common.h"
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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->downmix_needed && (avr->am->matrix ||
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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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/* select internal sample format if not specified by the user */
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if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
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(avr->mixing_needed || avr->resample_needed)) {
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enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
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enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
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int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
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av_get_bytes_per_sample(out_fmt));
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if (max_bps <= 2) {
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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_FLTP;
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} else {
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if (max_bps <= 4) {
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if (in_fmt == AV_SAMPLE_FMT_S32P ||
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out_fmt == AV_SAMPLE_FMT_S32P) {
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if (in_fmt == AV_SAMPLE_FMT_FLTP ||
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out_fmt == AV_SAMPLE_FMT_FLTP) {
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/* if one is s32 and the other is flt, use dbl */
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avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
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} else {
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/* if one is s32 and the other is s32, s16, or u8, use s32 */
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avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
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}
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} else {
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/* if one is flt and the other is flt, s16 or u8, use flt */
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avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
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}
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} else {
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/* if either is dbl, use dbl */
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avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
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}
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}
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av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
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av_get_sample_fmt_name(avr->internal_sample_fmt));
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}
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/* set sample format conversion parameters */
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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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avr->am = ff_audio_mix_alloc(avr);
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if (!avr->am) {
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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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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_free(&avr->am);
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av_freep(&avr->mix_matrix);
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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_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 attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
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uint8_t **output, int out_plane_size,
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int out_samples, uint8_t **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, direct_output;
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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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direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0;
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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 && direct_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,
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current_buffer);
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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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if (!avr->out_convert_needed && direct_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);
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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 (direct_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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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);
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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 ? &output_buffer : NULL,
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current_buffer);
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}
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int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
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int stride)
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{
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int in_channels, out_channels, i, o;
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if (avr->am)
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return ff_audio_mix_get_matrix(avr->am, matrix, stride);
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in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
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out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
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if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
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out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
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av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
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return AVERROR(EINVAL);
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}
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if (!avr->mix_matrix) {
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av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
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return AVERROR(EINVAL);
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}
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for (o = 0; o < out_channels; o++)
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for (i = 0; i < in_channels; i++)
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matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i];
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return 0;
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}
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int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
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int stride)
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{
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int in_channels, out_channels, i, o;
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if (avr->am)
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return ff_audio_mix_set_matrix(avr->am, matrix, stride);
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in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
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out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
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if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
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out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
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av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
if (avr->mix_matrix)
|
|
av_freep(&avr->mix_matrix);
|
|
avr->mix_matrix = av_malloc(in_channels * out_channels *
|
|
sizeof(*avr->mix_matrix));
|
|
if (!avr->mix_matrix)
|
|
return AVERROR(ENOMEM);
|
|
|
|
for (o = 0; o < out_channels; o++)
|
|
for (i = 0; i < in_channels; i++)
|
|
avr->mix_matrix[o * in_channels + i] = matrix[o * stride + i];
|
|
|
|
return 0;
|
|
}
|
|
|
|
int avresample_available(AVAudioResampleContext *avr)
|
|
{
|
|
return av_audio_fifo_size(avr->out_fifo);
|
|
}
|
|
|
|
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
|
|
{
|
|
if (!output)
|
|
return av_audio_fifo_drain(avr->out_fifo, nb_samples);
|
|
return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples);
|
|
}
|
|
|
|
unsigned avresample_version(void)
|
|
{
|
|
return LIBAVRESAMPLE_VERSION_INT;
|
|
}
|
|
|
|
const char *avresample_license(void)
|
|
{
|
|
#define LICENSE_PREFIX "libavresample license: "
|
|
return LICENSE_PREFIX LIBAV_LICENSE + sizeof(LICENSE_PREFIX) - 1;
|
|
}
|
|
|
|
const char *avresample_configuration(void)
|
|
{
|
|
return LIBAV_CONFIGURATION;
|
|
}
|