root/libavresample/utils.c

DEFINITIONS

1. avresample_open
2. avresample_is_open
3. avresample_close
4. avresample_free
5. handle_buffered_output
6. avresample_convert
7. avresample_config
8. config_changed
9. convert_frame
10. available_samples
11. avresample_convert_frame
12. avresample_get_matrix
13. avresample_set_matrix
14. avresample_set_channel_mapping
15. avresample_available
16. avresample_get_out_samples
17. avresample_read
18. avresample_version
19. avresample_license
20. avresample_configuration

/*
 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/common.h"
#include "libavutil/dict.h"
// #include "libavutil/error.h"
#include "libavutil/frame.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"

#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_convert.h"
#include "audio_mix.h"
#include "resample.h"

int avresample_open(AVAudioResampleContext *avr)
{
    int ret;

    if (avresample_is_open(avr)) {
        av_log(avr, AV_LOG_ERROR, "The resampling context is already open.\n");
        return AVERROR(EINVAL);
    }

    /* set channel mixing parameters */
    avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
    if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
        av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
               avr->in_channel_layout);
        return AVERROR(EINVAL);
    }
    avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
    if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
        av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
               avr->out_channel_layout);
        return AVERROR(EINVAL);
    }
    avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
    avr->downmix_needed    = avr->in_channels  > avr->out_channels;
    avr->upmix_needed      = avr->out_channels > avr->in_channels ||
                             (!avr->downmix_needed && (avr->mix_matrix ||
                              avr->in_channel_layout != avr->out_channel_layout));
    avr->mixing_needed     = avr->downmix_needed || avr->upmix_needed;

    /* set resampling parameters */
    avr->resample_needed   = avr->in_sample_rate != avr->out_sample_rate ||
                             avr->force_resampling;

    /* select internal sample format if not specified by the user */
    if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
        (avr->mixing_needed || avr->resample_needed)) {
        enum AVSampleFormat  in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
        enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
        int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
                            av_get_bytes_per_sample(out_fmt));
        if (max_bps <= 2) {
            avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
        } else if (avr->mixing_needed) {
            avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
        } else {
            if (max_bps <= 4) {
                if (in_fmt  == AV_SAMPLE_FMT_S32P ||
                    out_fmt == AV_SAMPLE_FMT_S32P) {
                    if (in_fmt  == AV_SAMPLE_FMT_FLTP ||
                        out_fmt == AV_SAMPLE_FMT_FLTP) {
                        /* if one is s32 and the other is flt, use dbl */
                        avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
                    } else {
                        /* if one is s32 and the other is s32, s16, or u8, use s32 */
                        avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
                    }
                } else {
                    /* if one is flt and the other is flt, s16 or u8, use flt */
                    avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
                }
            } else {
                /* if either is dbl, use dbl */
                avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
            }
        }
        av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
               av_get_sample_fmt_name(avr->internal_sample_fmt));
    }

    /* we may need to add an extra conversion in order to remap channels if
       the output format is not planar */
    if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
        !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels)) {
        avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
    }

    /* set sample format conversion parameters */
    if (avr->resample_needed || avr->mixing_needed)
        avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
    else
        avr->in_convert_needed = avr->use_channel_map &&
                                 !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels);

    if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
        avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
    else
        avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;

    avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
                          (avr->use_channel_map && avr->resample_needed));

    if (avr->use_channel_map) {
        if (avr->in_copy_needed) {
            avr->remap_point = REMAP_IN_COPY;
            av_dlog(avr, "remap channels during in_copy\n");
        } else if (avr->in_convert_needed) {
            avr->remap_point = REMAP_IN_CONVERT;
            av_dlog(avr, "remap channels during in_convert\n");
        } else if (avr->out_convert_needed) {
            avr->remap_point = REMAP_OUT_CONVERT;
            av_dlog(avr, "remap channels during out_convert\n");
        } else {
            avr->remap_point = REMAP_OUT_COPY;
            av_dlog(avr, "remap channels during out_copy\n");
        }

#ifdef DEBUG
        {
            int ch;
            av_dlog(avr, "output map: ");
            if (avr->ch_map_info.do_remap)
                for (ch = 0; ch < avr->in_channels; ch++)
                    av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]);
            else
                av_dlog(avr, "n/a");
            av_dlog(avr, "\n");
            av_dlog(avr, "copy map:   ");
            if (avr->ch_map_info.do_copy)
                for (ch = 0; ch < avr->in_channels; ch++)
                    av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]);
            else
                av_dlog(avr, "n/a");
            av_dlog(avr, "\n");
            av_dlog(avr, "zero map:   ");
            if (avr->ch_map_info.do_zero)
                for (ch = 0; ch < avr->in_channels; ch++)
                    av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]);
            else
                av_dlog(avr, "n/a");
            av_dlog(avr, "\n");
            av_dlog(avr, "input map:  ");
            for (ch = 0; ch < avr->in_channels; ch++)
                av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]);
            av_dlog(avr, "\n");
        }
#endif
    } else
        avr->remap_point = REMAP_NONE;

    /* allocate buffers */
    if (avr->in_copy_needed || avr->in_convert_needed) {
        avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
                                             0, avr->internal_sample_fmt,
                                             "in_buffer");
        if (!avr->in_buffer) {
            ret = AVERROR(EINVAL);
            goto error;
        }
    }
    if (avr->resample_needed) {
        avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
                                                       1024, avr->internal_sample_fmt,
                                                       "resample_out_buffer");
        if (!avr->resample_out_buffer) {
            ret = AVERROR(EINVAL);
            goto error;
        }
    }
    if (avr->out_convert_needed) {
        avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
                                              avr->out_sample_fmt, "out_buffer");
        if (!avr->out_buffer) {
            ret = AVERROR(EINVAL);
            goto error;
        }
    }
    avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
                                        1024);
    if (!avr->out_fifo) {
        ret = AVERROR(ENOMEM);
        goto error;
    }

    /* setup contexts */
    if (avr->in_convert_needed) {
        avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
                                            avr->in_sample_fmt, avr->in_channels,
                                            avr->in_sample_rate,
                                            avr->remap_point == REMAP_IN_CONVERT);
        if (!avr->ac_in) {
            ret = AVERROR(ENOMEM);
            goto error;
        }
    }
    if (avr->out_convert_needed) {
        enum AVSampleFormat src_fmt;
        if (avr->in_convert_needed)
            src_fmt = avr->internal_sample_fmt;
        else
            src_fmt = avr->in_sample_fmt;
        avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
                                             avr->out_channels,
                                             avr->out_sample_rate,
                                             avr->remap_point == REMAP_OUT_CONVERT);
        if (!avr->ac_out) {
            ret = AVERROR(ENOMEM);
            goto error;
        }
    }
    if (avr->resample_needed) {
        avr->resample = ff_audio_resample_init(avr);
        if (!avr->resample) {
            ret = AVERROR(ENOMEM);
            goto error;
        }
    }
    if (avr->mixing_needed) {
        avr->am = ff_audio_mix_alloc(avr);
        if (!avr->am) {
            ret = AVERROR(ENOMEM);
            goto error;
        }
    }

    return 0;

error:
    avresample_close(avr);
    return ret;
}

int avresample_is_open(AVAudioResampleContext *avr)
{
    return !!avr->out_fifo;
}

void avresample_close(AVAudioResampleContext *avr)
{
    ff_audio_data_free(&avr->in_buffer);
    ff_audio_data_free(&avr->resample_out_buffer);
    ff_audio_data_free(&avr->out_buffer);
    av_audio_fifo_free(avr->out_fifo);
    avr->out_fifo = NULL;
    ff_audio_convert_free(&avr->ac_in);
    ff_audio_convert_free(&avr->ac_out);
    ff_audio_resample_free(&avr->resample);
    ff_audio_mix_free(&avr->am);
    av_freep(&avr->mix_matrix);

    avr->use_channel_map = 0;
}

void avresample_free(AVAudioResampleContext **avr)
{
    if (!*avr)
        return;
    avresample_close(*avr);
    av_opt_free(*avr);
    av_freep(avr);
}

static int handle_buffered_output(AVAudioResampleContext *avr,
                                  AudioData *output, AudioData *converted)
{
    int ret;

    if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
        (converted && output->allocated_samples < converted->nb_samples)) {
        if (converted) {
            /* if there are any samples in the output FIFO or if the
               user-supplied output buffer is not large enough for all samples,
               we add to the output FIFO */
            av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name);
            ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
                                            converted->nb_samples);
            if (ret < 0)
                return ret;
        }

        /* if the user specified an output buffer, read samples from the output
           FIFO to the user output */
        if (output && output->allocated_samples > 0) {
            av_dlog(avr, "[FIFO] read from out_fifo to output\n");
            av_dlog(avr, "[end conversion]\n");
            return ff_audio_data_read_from_fifo(avr->out_fifo, output,
                                                output->allocated_samples);
        }
    } else if (converted) {
        /* copy directly to output if it is large enough or there is not any
           data in the output FIFO */
        av_dlog(avr, "[copy] %s to output\n", converted->name);
        output->nb_samples = 0;
        ret = ff_audio_data_copy(output, converted,
                                 avr->remap_point == REMAP_OUT_COPY ?
                                 &avr->ch_map_info : NULL);
        if (ret < 0)
            return ret;
        av_dlog(avr, "[end conversion]\n");
        return output->nb_samples;
    }
    av_dlog(avr, "[end conversion]\n");
    return 0;
}

int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
                                           uint8_t **output, int out_plane_size,
                                           int out_samples, uint8_t **input,
                                           int in_plane_size, int in_samples)
{
    AudioData input_buffer;
    AudioData output_buffer;
    AudioData *current_buffer;
    int ret, direct_output;

    /* reset internal buffers */
    if (avr->in_buffer) {
        avr->in_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->in_buffer,
                                   avr->in_buffer->allocated_channels);
    }
    if (avr->resample_out_buffer) {
        avr->resample_out_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->resample_out_buffer,
                                   avr->resample_out_buffer->allocated_channels);
    }
    if (avr->out_buffer) {
        avr->out_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->out_buffer,
                                   avr->out_buffer->allocated_channels);
    }

    av_dlog(avr, "[start conversion]\n");

    /* initialize output_buffer with output data */
    direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0;
    if (output) {
        ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
                                 avr->out_channels, out_samples,
                                 avr->out_sample_fmt, 0, "output");
        if (ret < 0)
            return ret;
        output_buffer.nb_samples = 0;
    }

    if (input) {
        /* initialize input_buffer with input data */
        ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
                                 avr->in_channels, in_samples,
                                 avr->in_sample_fmt, 1, "input");
        if (ret < 0)
            return ret;
        current_buffer = &input_buffer;

        if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
            !avr->out_convert_needed && direct_output && out_samples >= in_samples) {
            /* in some rare cases we can copy input to output and upmix
               directly in the output buffer */
            av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
            ret = ff_audio_data_copy(&output_buffer, current_buffer,
                                     avr->remap_point == REMAP_OUT_COPY ?
                                     &avr->ch_map_info : NULL);
            if (ret < 0)
                return ret;
            current_buffer = &output_buffer;
        } else if (avr->remap_point == REMAP_OUT_COPY &&
                   (!direct_output || out_samples < in_samples)) {
            /* if remapping channels during output copy, we may need to
             * use an intermediate buffer in order to remap before adding
             * samples to the output fifo */
            av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name);
            ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
                                     &avr->ch_map_info);
            if (ret < 0)
                return ret;
            current_buffer = avr->out_buffer;
        } else if (avr->in_copy_needed || avr->in_convert_needed) {
            /* if needed, copy or convert input to in_buffer, and downmix if
               applicable */
            if (avr->in_convert_needed) {
                ret = ff_audio_data_realloc(avr->in_buffer,
                                            current_buffer->nb_samples);
                if (ret < 0)
                    return ret;
                av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
                ret = ff_audio_convert(avr->ac_in, avr->in_buffer,
                                       current_buffer);
                if (ret < 0)
                    return ret;
            } else {
                av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
                ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
                                         avr->remap_point == REMAP_IN_COPY ?
                                         &avr->ch_map_info : NULL);
                if (ret < 0)
                    return ret;
            }
            ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
            if (avr->downmix_needed) {
                av_dlog(avr, "[downmix] in_buffer\n");
                ret = ff_audio_mix(avr->am, avr->in_buffer);
                if (ret < 0)
                    return ret;
            }
            current_buffer = avr->in_buffer;
        }
    } else {
        /* flush resampling buffer and/or output FIFO if input is NULL */
        if (!avr->resample_needed)
            return handle_buffered_output(avr, output ? &output_buffer : NULL,
                                          NULL);
        current_buffer = NULL;
    }

    if (avr->resample_needed) {
        AudioData *resample_out;

        if (!avr->out_convert_needed && direct_output && out_samples > 0)
            resample_out = &output_buffer;
        else
            resample_out = avr->resample_out_buffer;
        av_dlog(avr, "[resample] %s to %s\n",
                current_buffer ? current_buffer->name : "null",
                resample_out->name);
        ret = ff_audio_resample(avr->resample, resample_out,
                                current_buffer);
        if (ret < 0)
            return ret;

        /* if resampling did not produce any samples, just return 0 */
        if (resample_out->nb_samples == 0) {
            av_dlog(avr, "[end conversion]\n");
            return 0;
        }

        current_buffer = resample_out;
    }

    if (avr->upmix_needed) {
        av_dlog(avr, "[upmix] %s\n", current_buffer->name);
        ret = ff_audio_mix(avr->am, current_buffer);
        if (ret < 0)
            return ret;
    }

    /* if we resampled or upmixed directly to output, return here */
    if (current_buffer == &output_buffer) {
        av_dlog(avr, "[end conversion]\n");
        return current_buffer->nb_samples;
    }

    if (avr->out_convert_needed) {
        if (direct_output && out_samples >= current_buffer->nb_samples) {
            /* convert directly to output */
            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_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
{
    if (avresample_is_open(avr)) {
        avresample_close(avr);
    }

    if (in) {
        avr->in_channel_layout  = in->channel_layout;
        avr->in_sample_rate     = in->sample_rate;
        avr->in_sample_fmt      = in->format;
    }

    if (out) {
        avr->out_channel_layout = out->channel_layout;
        avr->out_sample_rate    = out->sample_rate;
        avr->out_sample_fmt     = out->format;
    }

    return 0;
}

static int config_changed(AVAudioResampleContext *avr,
                          AVFrame *out, AVFrame *in)
{
    int ret = 0;

    if (in) {
        if (avr->in_channel_layout != in->channel_layout ||
            avr->in_sample_rate    != in->sample_rate ||
            avr->in_sample_fmt     != in->format) {
            ret |= AVERROR_INPUT_CHANGED;
        }
    }

    if (out) {
        if (avr->out_channel_layout != out->channel_layout ||
            avr->out_sample_rate    != out->sample_rate ||
            avr->out_sample_fmt     != out->format) {
            ret |= AVERROR_OUTPUT_CHANGED;
        }
    }

    return ret;
}

static inline int convert_frame(AVAudioResampleContext *avr,
                                AVFrame *out, AVFrame *in)
{
    int ret;
    uint8_t **out_data = NULL, **in_data = NULL;
    int out_linesize = 0, in_linesize = 0;
    int out_nb_samples = 0, in_nb_samples = 0;

    if (out) {
        out_data       = out->extended_data;
        out_linesize   = out->linesize[0];
        out_nb_samples = out->nb_samples;
    }

    if (in) {
        in_data       = in->extended_data;
        in_linesize   = in->linesize[0];
        in_nb_samples = in->nb_samples;
    }

    ret = avresample_convert(avr, out_data, out_linesize,
                             out_nb_samples,
                             in_data, in_linesize,
                             in_nb_samples);

    if (ret < 0) {
        if (out)
            out->nb_samples = 0;
        return ret;
    }

    if (out)
        out->nb_samples = ret;

    return 0;
}

static inline int available_samples(AVFrame *out)
{
    int samples;
    int bytes_per_sample = av_get_bytes_per_sample(out->format);
    if (!bytes_per_sample)
        return AVERROR(EINVAL);

    samples = out->linesize[0] / bytes_per_sample;
    if (av_sample_fmt_is_planar(out->format)) {
        return samples;
    } else {
        int channels = av_get_channel_layout_nb_channels(out->channel_layout);
        return samples / channels;
    }
}

int avresample_convert_frame(AVAudioResampleContext *avr,
                             AVFrame *out, AVFrame *in)
{
    int ret, setup = 0;

    if (!avresample_is_open(avr)) {
        if ((ret = avresample_config(avr, out, in)) < 0)
            return ret;
        if ((ret = avresample_open(avr)) < 0)
            return ret;
        setup = 1;
    } else {
        // return as is or reconfigure for input changes?
        if ((ret = config_changed(avr, out, in)))
            return ret;
    }

    if (out) {
        if (!out->linesize[0]) {
            out->nb_samples = avresample_get_out_samples(avr, in->nb_samples);
            if ((ret = av_frame_get_buffer(out, 0)) < 0) {
                if (setup)
                    avresample_close(avr);
                return ret;
            }
        } else {
            if (!out->nb_samples)
                out->nb_samples = available_samples(out);
        }
    }

    return convert_frame(avr, out, in);
}

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_set_channel_mapping(AVAudioResampleContext *avr,
                                   const int *channel_map)
{
    ChannelMapInfo *info = &avr->ch_map_info;
    int in_channels, ch, i;

    in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
    if (in_channels <= 0 ||  in_channels > AVRESAMPLE_MAX_CHANNELS) {
        av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n");
        return AVERROR(EINVAL);
    }

    memset(info, 0, sizeof(*info));
    memset(info->input_map, -1, sizeof(info->input_map));

    for (ch = 0; ch < in_channels; ch++) {
        if (channel_map[ch] >= in_channels) {
            av_log(avr, AV_LOG_ERROR, "Invalid channel map\n");
            return AVERROR(EINVAL);
        }
        if (channel_map[ch] < 0) {
            info->channel_zero[ch] =  1;
            info->channel_map[ch]  = -1;
            info->do_zero          =  1;
        } else if (info->input_map[channel_map[ch]] >= 0) {
            info->channel_copy[ch] = info->input_map[channel_map[ch]];
            info->channel_map[ch]  = -1;
            info->do_copy          =  1;
        } else {
            info->channel_map[ch]            = channel_map[ch];
            info->input_map[channel_map[ch]] = ch;
            info->do_remap                   =  1;
        }
    }
    /* Fill-in unmapped input channels with unmapped output channels.
       This is used when remapping during conversion from interleaved to
       planar format. */
    for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
        while (ch < in_channels && info->input_map[ch] >= 0)
            ch++;
        while (i < in_channels && info->channel_map[i] >= 0)
            i++;
        if (ch >= in_channels || i >= in_channels)
            break;
        info->input_map[ch] = i;
    }

    avr->use_channel_map = 1;
    return 0;
}

int avresample_available(AVAudioResampleContext *avr)
{
    return av_audio_fifo_size(avr->out_fifo);
}

int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
{
    int64_t samples = avresample_get_delay(avr) + (int64_t)in_nb_samples;

    if (avr->resample_needed) {
        samples = av_rescale_rnd(samples,
                                 avr->out_sample_rate,
                                 avr->in_sample_rate,
                                 AV_ROUND_UP);
    }

    samples += avresample_available(avr);

    if (samples > INT_MAX)
        return AVERROR(EINVAL);

    return samples;
}

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 FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
}

const char *avresample_configuration(void)
{
    return FFMPEG_CONFIGURATION;
}