/*
  * 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_log(avr, AV_LOG_TRACE, "remap channels during in_copy\n");

         } else if (avr->in_convert_needed) {
             avr->remap_point = REMAP_IN_CONVERT;

             av_log(avr, AV_LOG_TRACE, "remap channels during in_convert\n");

         } else if (avr->out_convert_needed) {
             avr->remap_point = REMAP_OUT_CONVERT;

             av_log(avr, AV_LOG_TRACE, "remap channels during out_convert\n");

         } else {
             avr->remap_point = REMAP_OUT_COPY;

             av_log(avr, AV_LOG_TRACE, "remap channels during out_copy\n");

         }
 
 #ifdef DEBUG
         {
             int ch;

             av_log(avr, AV_LOG_TRACE, "output map: ");

             if (avr->ch_map_info.do_remap)
                 for (ch = 0; ch < avr->in_channels; ch++)

                     av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_map[ch]);

             else

                 av_log(avr, AV_LOG_TRACE, "n/a");
             av_log(avr, AV_LOG_TRACE, "\n");
             av_log(avr, AV_LOG_TRACE, "copy map:   ");

             if (avr->ch_map_info.do_copy)
                 for (ch = 0; ch < avr->in_channels; ch++)

                     av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_copy[ch]);

             else

                 av_log(avr, AV_LOG_TRACE, "n/a");
             av_log(avr, AV_LOG_TRACE, "\n");
             av_log(avr, AV_LOG_TRACE, "zero map:   ");

             if (avr->ch_map_info.do_zero)
                 for (ch = 0; ch < avr->in_channels; ch++)

                     av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_zero[ch]);

             else

                 av_log(avr, AV_LOG_TRACE, "n/a");
             av_log(avr, AV_LOG_TRACE, "\n");
             av_log(avr, AV_LOG_TRACE, "input map:  ");

             for (ch = 0; ch < avr->in_channels; ch++)

                 av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.input_map[ch]);
             av_log(avr, AV_LOG_TRACE, "\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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[FIFO] read from out_fifo to output\n");
             av_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[end conversion]\n");

         return output->nb_samples;
     }

     av_log(avr, AV_LOG_TRACE, "[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 * const *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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[end conversion]\n");

             return 0;
         }
 
         current_buffer = resample_out;
     }
 
     if (avr->upmix_needed) {

         av_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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_log(avr, AV_LOG_TRACE, "[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;

 }