497 lines
		
	
	
		
			18 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			497 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->mix_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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                                            avr->in_sample_rate);
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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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                                             avr->out_sample_rate);
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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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    ff_audio_convert_free(&avr->ac_in);
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    ff_audio_convert_free(&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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 | 
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    if (avr->resample_needed) {
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        AudioData *resample_out;
 | 
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 | 
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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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 | 
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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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 | 
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        current_buffer = resample_out;
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    }
 | 
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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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 | 
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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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 | 
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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);
 | 
						|
            ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer);
 | 
						|
            if (ret < 0)
 | 
						|
                return ret;
 | 
						|
 | 
						|
            av_dlog(avr, "[end conversion]\n");
 | 
						|
            return output_buffer.nb_samples;
 | 
						|
        } else {
 | 
						|
            ret = ff_audio_data_realloc(avr->out_buffer,
 | 
						|
                                        current_buffer->nb_samples);
 | 
						|
            if (ret < 0)
 | 
						|
                return ret;
 | 
						|
            av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
 | 
						|
            ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
 | 
						|
                                   current_buffer);
 | 
						|
            if (ret < 0)
 | 
						|
                return ret;
 | 
						|
            current_buffer = avr->out_buffer;
 | 
						|
        }
 | 
						|
    }
 | 
						|
 | 
						|
    return handle_buffered_output(avr, output ? &output_buffer : NULL,
 | 
						|
                                  current_buffer);
 | 
						|
}
 | 
						|
 | 
						|
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
 | 
						|
                          int stride)
 | 
						|
{
 | 
						|
    int in_channels, out_channels, i, o;
 | 
						|
 | 
						|
    if (avr->am)
 | 
						|
        return ff_audio_mix_get_matrix(avr->am, matrix, stride);
 | 
						|
 | 
						|
    in_channels  = av_get_channel_layout_nb_channels(avr->in_channel_layout);
 | 
						|
    out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
 | 
						|
 | 
						|
    if ( in_channels <= 0 ||  in_channels > AVRESAMPLE_MAX_CHANNELS ||
 | 
						|
        out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
 | 
						|
        av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
 | 
						|
        return AVERROR(EINVAL);
 | 
						|
    }
 | 
						|
 | 
						|
    if (!avr->mix_matrix) {
 | 
						|
        av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
 | 
						|
        return AVERROR(EINVAL);
 | 
						|
    }
 | 
						|
 | 
						|
    for (o = 0; o < out_channels; o++)
 | 
						|
        for (i = 0; i < in_channels; i++)
 | 
						|
            matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i];
 | 
						|
 | 
						|
    return 0;
 | 
						|
}
 | 
						|
 | 
						|
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
 | 
						|
                          int stride)
 | 
						|
{
 | 
						|
    int in_channels, out_channels, i, o;
 | 
						|
 | 
						|
    if (avr->am)
 | 
						|
        return ff_audio_mix_set_matrix(avr->am, matrix, stride);
 | 
						|
 | 
						|
    in_channels  = av_get_channel_layout_nb_channels(avr->in_channel_layout);
 | 
						|
    out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
 | 
						|
 | 
						|
    if ( in_channels <= 0 ||  in_channels > AVRESAMPLE_MAX_CHANNELS ||
 | 
						|
        out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
 | 
						|
        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;
 | 
						|
}
 |