libavcodec/resample.c - nest-android-app/ffmpeg - Git at Google

 /*
  * samplerate conversion for both audio and video
  * Copyright (c) 2000 Fabrice Bellard
  *
  * 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
  */

 /**
  * @file
  * samplerate conversion for both audio and video
  */

 #include <string.h>

 #include "avcodec.h"
 #include "audioconvert.h"
 #include "libavutil/opt.h"
 #include "libavutil/mem.h"
 #include "libavutil/samplefmt.h"

 #if FF_API_AVCODEC_RESAMPLE
 FF_DISABLE_DEPRECATION_WARNINGS

 #define MAX_CHANNELS 8

 struct AVResampleContext;

 static const char *context_to_name(void *ptr)
 {
     return "audioresample";
 }

 static const AVOption options[] = {{NULL}};
 static const AVClass audioresample_context_class = {
     "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
 };

 struct ReSampleContext {
     struct AVResampleContext *resample_context;
     short *temp[MAX_CHANNELS];
     int temp_len;
     float ratio;
     /* channel convert */
     int input_channels, output_channels, filter_channels;
     AVAudioConvert *convert_ctx[2];
     enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
     unsigned sample_size[2];           ///< size of one sample in sample_fmt
     short *buffer[2];                  ///< buffers used for conversion to S16
     unsigned buffer_size[2];           ///< sizes of allocated buffers
 };

 /* n1: number of samples */
 static void stereo_to_mono(short *output, short *input, int n1)
 {
     short *p, *q;
     int n = n1;

     p = input;
     q = output;
     while (n >= 4) {
         q[0] = (p[0] + p[1]) >> 1;
         q[1] = (p[2] + p[3]) >> 1;
         q[2] = (p[4] + p[5]) >> 1;
         q[3] = (p[6] + p[7]) >> 1;
         q += 4;
         p += 8;
         n -= 4;
     }
     while (n > 0) {
         q[0] = (p[0] + p[1]) >> 1;
         q++;
         p += 2;
         n--;
     }
 }

 /* n1: number of samples */
 static void mono_to_stereo(short *output, short *input, int n1)
 {
     short *p, *q;
     int n = n1;
     int v;

     p = input;
     q = output;
     while (n >= 4) {
         v = p[0]; q[0] = v; q[1] = v;
         v = p[1]; q[2] = v; q[3] = v;
         v = p[2]; q[4] = v; q[5] = v;
         v = p[3]; q[6] = v; q[7] = v;
         q += 8;
         p += 4;
         n -= 4;
     }
     while (n > 0) {
         v = p[0]; q[0] = v; q[1] = v;
         q += 2;
         p += 1;
         n--;
     }
 }

 /*
 5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
 - Left = front_left + rear_gain * rear_left + center_gain * center
 - Right = front_right + rear_gain * rear_right + center_gain * center
 Where rear_gain is usually around 0.5-1.0 and
       center_gain is almost always 0.7 (-3 dB)
 */
 static void surround_to_stereo(short **output, short *input, int channels, int samples)
 {
     int i;
     short l, r;

     for (i = 0; i < samples; i++) {
         int fl,fr,c,rl,rr;
         fl = input[0];
         fr = input[1];
         c = input[2];
         // lfe = input[3];
         rl = input[4];
         rr = input[5];

         l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
         r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));

         /* output l & r. */
         *output[0]++ = l;
         *output[1]++ = r;

         /* increment input. */
         input += channels;
     }
 }

 static void deinterleave(short **output, short *input, int channels, int samples)
 {
     int i, j;

     for (i = 0; i < samples; i++) {
         for (j = 0; j < channels; j++) {
             *output[j]++ = *input++;
         }
     }
 }

 static void interleave(short *output, short **input, int channels, int samples)
 {
     int i, j;

     for (i = 0; i < samples; i++) {
         for (j = 0; j < channels; j++) {
             *output++ = *input[j]++;
         }
     }
 }

 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 {
     int i;
     short l, r;

     for (i = 0; i < n; i++) {
         l = *input1++;
         r = *input2++;
         *output++ = l;                  /* left */
         *output++ = (l / 2) + (r / 2);  /* center */
         *output++ = r;                  /* right */
         *output++ = 0;                  /* left surround */
         *output++ = 0;                  /* right surroud */
         *output++ = 0;                  /* low freq */
     }
 }

 #define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
     ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0

 static const uint8_t supported_resampling[MAX_CHANNELS] = {
     // output ch:    1  2  3  4  5  6  7  8
     SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
     SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
     SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
     SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
     SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
     SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
     SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
     SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
 };

 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
                                         int output_rate, int input_rate,
                                         enum AVSampleFormat sample_fmt_out,
                                         enum AVSampleFormat sample_fmt_in,
                                         int filter_length, int log2_phase_count,
                                         int linear, double cutoff)
 {
     ReSampleContext *s;

     if (input_channels > MAX_CHANNELS) {
         av_log(NULL, AV_LOG_ERROR,
                "Resampling with input channels greater than %d is unsupported.\n",
                MAX_CHANNELS);
         return NULL;
     }
     if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
         int i;
         av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
                "output channels for %d input channel%s", input_channels,
                input_channels > 1 ? "s:" : ":");
         for (i = 0; i < MAX_CHANNELS; i++)
             if (supported_resampling[input_channels-1] & (1<<i))
                 av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
         av_log(NULL, AV_LOG_ERROR, "\n");
         return NULL;
     }

     s = av_mallocz(sizeof(ReSampleContext));
     if (!s) {
         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
         return NULL;
     }

     s->ratio = (float)output_rate / (float)input_rate;

     s->input_channels = input_channels;
     s->output_channels = output_channels;

     s->filter_channels = s->input_channels;
     if (s->output_channels < s->filter_channels)
         s->filter_channels = s->output_channels;

     s->sample_fmt[0]  = sample_fmt_in;
     s->sample_fmt[1]  = sample_fmt_out;
     s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
     s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);

     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
         if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
                                                          s->sample_fmt[0], 1, NULL, 0))) {
             av_log(s, AV_LOG_ERROR,
                    "Cannot convert %s sample format to s16 sample format\n",
                    av_get_sample_fmt_name(s->sample_fmt[0]));
             av_free(s);
             return NULL;
         }
     }

     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
                                                          AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
             av_log(s, AV_LOG_ERROR,
                    "Cannot convert s16 sample format to %s sample format\n",
                    av_get_sample_fmt_name(s->sample_fmt[1]));
             av_audio_convert_free(s->convert_ctx[0]);
             av_free(s);
             return NULL;
         }
     }

     s->resample_context = av_resample_init(output_rate, input_rate,
                                            filter_length, log2_phase_count,
                                            linear, cutoff);

     *(const AVClass**)s->resample_context = &audioresample_context_class;

     return s;
 }

 /* resample audio. 'nb_samples' is the number of input samples */
 /* XXX: optimize it ! */
 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 {
     int i, nb_samples1;
     short *bufin[MAX_CHANNELS];
     short *bufout[MAX_CHANNELS];
     short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
     short *output_bak = NULL;
     int lenout;

     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
         /* nothing to do */
         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
         return nb_samples;
     }

     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
         int istride[1] = { s->sample_size[0] };
         int ostride[1] = { 2 };
         const void *ibuf[1] = { input };
         void       *obuf[1];
         unsigned input_size = nb_samples * s->input_channels * 2;

         if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
             av_free(s->buffer[0]);
             s->buffer_size[0] = input_size;
             s->buffer[0] = av_malloc(s->buffer_size[0]);
             if (!s->buffer[0]) {
                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
                 return 0;
             }
         }

         obuf[0] = s->buffer[0];

         if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
                              ibuf, istride, nb_samples * s->input_channels) < 0) {
             av_log(s->resample_context, AV_LOG_ERROR,
                    "Audio sample format conversion failed\n");
             return 0;
         }

         input = s->buffer[0];
     }

     lenout= 2*s->output_channels*nb_samples * s->ratio + 16;

     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
         int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
                        s->output_channels;
         output_bak = output;

         if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
             av_free(s->buffer[1]);
             s->buffer_size[1] = out_size;
             s->buffer[1] = av_malloc(s->buffer_size[1]);
             if (!s->buffer[1]) {
                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
                 return 0;
             }
         }

         output = s->buffer[1];
     }

     /* XXX: move those malloc to resample init code */
     for (i = 0; i < s->filter_channels; i++) {
         bufin[i] = av_malloc_array((nb_samples + s->temp_len), sizeof(short));
         bufout[i] = av_malloc_array(lenout, sizeof(short));

         if (!bufin[i] || !bufout[i]) {
             av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
             nb_samples1 = 0;
             goto fail;
         }

         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
         buftmp2[i] = bufin[i] + s->temp_len;
     }

     if (s->input_channels == 2 && s->output_channels == 1) {
         buftmp3[0] = output;
         stereo_to_mono(buftmp2[0], input, nb_samples);
     } else if (s->output_channels >= 2 && s->input_channels == 1) {
         buftmp3[0] = bufout[0];
         memcpy(buftmp2[0], input, nb_samples * sizeof(short));
     } else if (s->input_channels == 6 && s->output_channels ==2) {
         buftmp3[0] = bufout[0];
         buftmp3[1] = bufout[1];
         surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
     } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
         for (i = 0; i < s->input_channels; i++) {
             buftmp3[i] = bufout[i];
         }
         deinterleave(buftmp2, input, s->input_channels, nb_samples);
     } else {
         buftmp3[0] = output;
         memcpy(buftmp2[0], input, nb_samples * sizeof(short));
     }

     nb_samples += s->temp_len;

     /* resample each channel */
     nb_samples1 = 0; /* avoid warning */
     for (i = 0; i < s->filter_channels; i++) {
         int consumed;
         int is_last = i + 1 == s->filter_channels;

         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
                                   &consumed, nb_samples, lenout, is_last);
         s->temp_len = nb_samples - consumed;
         s->temp[i] = av_realloc_array(s->temp[i], s->temp_len, sizeof(short));
         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
     }

     if (s->output_channels == 2 && s->input_channels == 1) {
         mono_to_stereo(output, buftmp3[0], nb_samples1);
     } else if (s->output_channels == 6 && s->input_channels == 2) {
         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
     } else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
                (s->output_channels == 2 && s->input_channels == 6)) {
         interleave(output, buftmp3, s->output_channels, nb_samples1);
     }

     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
         int istride[1] = { 2 };
         int ostride[1] = { s->sample_size[1] };
         const void *ibuf[1] = { output };
         void       *obuf[1] = { output_bak };

         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
                              ibuf, istride, nb_samples1 * s->output_channels) < 0) {
             av_log(s->resample_context, AV_LOG_ERROR,
                    "Audio sample format conversion failed\n");
             return 0;
         }
     }

 fail:
     for (i = 0; i < s->filter_channels; i++) {
         av_free(bufin[i]);
         av_free(bufout[i]);
     }

     return nb_samples1;
 }

 void audio_resample_close(ReSampleContext *s)
 {
     int i;
     av_resample_close(s->resample_context);
     for (i = 0; i < s->filter_channels; i++)
         av_freep(&s->temp[i]);
     av_freep(&s->buffer[0]);
     av_freep(&s->buffer[1]);
     av_audio_convert_free(s->convert_ctx[0]);
     av_audio_convert_free(s->convert_ctx[1]);
     av_free(s);
 }

 FF_ENABLE_DEPRECATION_WARNINGS
 #endif
	/*
	* samplerate conversion for both audio and video
	* Copyright (c) 2000 Fabrice Bellard
	*
	* 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
	*/

	/**
	* @file
	* samplerate conversion for both audio and video
	*/

	#include <string.h>

	#include "avcodec.h"
	#include "audioconvert.h"
	#include "libavutil/opt.h"
	#include "libavutil/mem.h"
	#include "libavutil/samplefmt.h"

	#if FF_API_AVCODEC_RESAMPLE
	FF_DISABLE_DEPRECATION_WARNINGS

	#define MAX_CHANNELS 8

	struct AVResampleContext;

	static const char context_to_name(void ptr)
	{
	return "audioresample";
	}

	static const AVOption options[] = {{NULL}};
	static const AVClass audioresample_context_class = {
	"ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
	};

	struct ReSampleContext {
	struct AVResampleContext *resample_context;
	short *temp[MAX_CHANNELS];
	int temp_len;
	float ratio;
	/* channel convert */
	int input_channels, output_channels, filter_channels;
	AVAudioConvert *convert_ctx[2];
	enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
	unsigned sample_size[2]; ///< size of one sample in sample_fmt
	short *buffer[2]; ///< buffers used for conversion to S16
	unsigned buffer_size[2]; ///< sizes of allocated buffers
	};

	/* n1: number of samples */
	static void stereo_to_mono(short output, short input, int n1)
	{
	short p, q;
	int n = n1;

	p = input;
	q = output;
	while (n >= 4) {
	q[0] = (p[0] + p[1]) >> 1;
	q[1] = (p[2] + p[3]) >> 1;
	q[2] = (p[4] + p[5]) >> 1;
	q[3] = (p[6] + p[7]) >> 1;
	q += 4;
	p += 8;
	n -= 4;
	}
	while (n > 0) {
	q[0] = (p[0] + p[1]) >> 1;
	q++;
	p += 2;
	n--;
	}
	}

	/* n1: number of samples */
	static void mono_to_stereo(short output, short input, int n1)
	{
	short p, q;
	int n = n1;
	int v;

	p = input;
	q = output;
	while (n >= 4) {
	v = p[0]; q[0] = v; q[1] = v;
	v = p[1]; q[2] = v; q[3] = v;
	v = p[2]; q[4] = v; q[5] = v;
	v = p[3]; q[6] = v; q[7] = v;
	q += 8;
	p += 4;
	n -= 4;
	}
	while (n > 0) {
	v = p[0]; q[0] = v; q[1] = v;
	q += 2;
	p += 1;
	n--;
	}
	}

	/*
	5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
	- Left = front_left + rear_gain * rear_left + center_gain * center
	- Right = front_right + rear_gain * rear_right + center_gain * center
	Where rear_gain is usually around 0.5-1.0 and
	center_gain is almost always 0.7 (-3 dB)
	*/
	static void surround_to_stereo(short *output, short input, int channels, int samples)
	{
	int i;
	short l, r;

	for (i = 0; i < samples; i++) {
	int fl,fr,c,rl,rr;
	fl = input[0];
	fr = input[1];
	c = input[2];
	// lfe = input[3];
	rl = input[4];
	rr = input[5];

	l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
	r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));

	/* output l & r. */
	*output[0]++ = l;
	*output[1]++ = r;

	/* increment input. */
	input += channels;
	}
	}

	static void deinterleave(short *output, short input, int channels, int samples)
	{
	int i, j;

	for (i = 0; i < samples; i++) {
	for (j = 0; j < channels; j++) {
	output[j]++ = input++;
	}
	}
	}

	static void interleave(short output, short *input, int channels, int samples)
	{
	int i, j;

	for (i = 0; i < samples; i++) {
	for (j = 0; j < channels; j++) {
	output++ = input[j]++;
	}
	}
	}

	static void ac3_5p1_mux(short output, short input1, short *input2, int n)
	{
	int i;
	short l, r;

	for (i = 0; i < n; i++) {
	l = *input1++;
	r = *input2++;
	output++ = l; / left */
	output++ = (l / 2) + (r / 2); / center */
	output++ = r; / right */
	output++ = 0; / left surround */
	output++ = 0; / right surroud */
	output++ = 0; / low freq */
	}
	}

	#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
	ch8<<7 \| ch7<<6 \| ch6<<5 \| ch5<<4 \| ch4<<3 \| ch3<<2 \| ch2<<1 \| ch1<<0

	static const uint8_t supported_resampling[MAX_CHANNELS] = {
	// output ch: 1 2 3 4 5 6 7 8
	SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
	SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
	SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
	SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
	SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
	SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
	SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
	SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
	};

	ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
	int output_rate, int input_rate,
	enum AVSampleFormat sample_fmt_out,
	enum AVSampleFormat sample_fmt_in,
	int filter_length, int log2_phase_count,
	int linear, double cutoff)
	{
	ReSampleContext *s;

	if (input_channels > MAX_CHANNELS) {
	av_log(NULL, AV_LOG_ERROR,
	"Resampling with input channels greater than %d is unsupported.\n",
	MAX_CHANNELS);
	return NULL;
	}
	if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
	int i;
	av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
	"output channels for %d input channel%s", input_channels,
	input_channels > 1 ? "s:" : ":");
	for (i = 0; i < MAX_CHANNELS; i++)
	if (supported_resampling[input_channels-1] & (1<<i))
	av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
	av_log(NULL, AV_LOG_ERROR, "\n");
	return NULL;
	}

	s = av_mallocz(sizeof(ReSampleContext));
	if (!s) {
	av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
	return NULL;
	}

	s->ratio = (float)output_rate / (float)input_rate;

	s->input_channels = input_channels;
	s->output_channels = output_channels;

	s->filter_channels = s->input_channels;
	if (s->output_channels < s->filter_channels)
	s->filter_channels = s->output_channels;

	s->sample_fmt[0] = sample_fmt_in;
	s->sample_fmt[1] = sample_fmt_out;
	s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
	s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);

	if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
	if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
	s->sample_fmt[0], 1, NULL, 0))) {
	av_log(s, AV_LOG_ERROR,
	"Cannot convert %s sample format to s16 sample format\n",
	av_get_sample_fmt_name(s->sample_fmt[0]));
	av_free(s);
	return NULL;
	}
	}

	if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
	if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
	AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
	av_log(s, AV_LOG_ERROR,
	"Cannot convert s16 sample format to %s sample format\n",
	av_get_sample_fmt_name(s->sample_fmt[1]));
	av_audio_convert_free(s->convert_ctx[0]);
	av_free(s);
	return NULL;
	}
	}

	s->resample_context = av_resample_init(output_rate, input_rate,
	filter_length, log2_phase_count,
	linear, cutoff);

	(const AVClass*)s->resample_context = &audioresample_context_class;

	return s;
	}

	/* resample audio. 'nb_samples' is the number of input samples */
	/* XXX: optimize it ! */
	int audio_resample(ReSampleContext s, short output, short *input, int nb_samples)
	{
	int i, nb_samples1;
	short *bufin[MAX_CHANNELS];
	short *bufout[MAX_CHANNELS];
	short buftmp2[MAX_CHANNELS], buftmp3[MAX_CHANNELS];
	short *output_bak = NULL;
	int lenout;

	if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
	/* nothing to do */
	memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
	return nb_samples;
	}

	if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
	int istride[1] = { s->sample_size[0] };
	int ostride[1] = { 2 };
	const void *ibuf[1] = { input };
	void *obuf[1];
	unsigned input_size = nb_samples * s->input_channels * 2;

	if (!s->buffer_size[0] \|\| s->buffer_size[0] < input_size) {
	av_free(s->buffer[0]);
	s->buffer_size[0] = input_size;
	s->buffer[0] = av_malloc(s->buffer_size[0]);
	if (!s->buffer[0]) {
	av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
	return 0;
	}
	}

	obuf[0] = s->buffer[0];

	if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
	ibuf, istride, nb_samples * s->input_channels) < 0) {
	av_log(s->resample_context, AV_LOG_ERROR,
	"Audio sample format conversion failed\n");
	return 0;
	}

	input = s->buffer[0];
	}

	lenout= 2s->output_channelsnb_samples * s->ratio + 16;

	if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
	int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
	s->output_channels;
	output_bak = output;

	if (!s->buffer_size[1] \|\| s->buffer_size[1] < out_size) {
	av_free(s->buffer[1]);
	s->buffer_size[1] = out_size;
	s->buffer[1] = av_malloc(s->buffer_size[1]);
	if (!s->buffer[1]) {
	av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
	return 0;
	}
	}

	output = s->buffer[1];
	}

	/* XXX: move those malloc to resample init code */
	for (i = 0; i < s->filter_channels; i++) {
	bufin[i] = av_malloc_array((nb_samples + s->temp_len), sizeof(short));
	bufout[i] = av_malloc_array(lenout, sizeof(short));

	if (!bufin[i] \|\| !bufout[i]) {
	av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
	nb_samples1 = 0;
	goto fail;
	}

	memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
	buftmp2[i] = bufin[i] + s->temp_len;
	}

	if (s->input_channels == 2 && s->output_channels == 1) {
	buftmp3[0] = output;
	stereo_to_mono(buftmp2[0], input, nb_samples);
	} else if (s->output_channels >= 2 && s->input_channels == 1) {
	buftmp3[0] = bufout[0];
	memcpy(buftmp2[0], input, nb_samples * sizeof(short));
	} else if (s->input_channels == 6 && s->output_channels ==2) {
	buftmp3[0] = bufout[0];
	buftmp3[1] = bufout[1];
	surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
	} else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
	for (i = 0; i < s->input_channels; i++) {
	buftmp3[i] = bufout[i];
	}
	deinterleave(buftmp2, input, s->input_channels, nb_samples);
	} else {
	buftmp3[0] = output;
	memcpy(buftmp2[0], input, nb_samples * sizeof(short));
	}

	nb_samples += s->temp_len;

	/* resample each channel */
	nb_samples1 = 0; /* avoid warning */
	for (i = 0; i < s->filter_channels; i++) {
	int consumed;
	int is_last = i + 1 == s->filter_channels;

	nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
	&consumed, nb_samples, lenout, is_last);
	s->temp_len = nb_samples - consumed;
	s->temp[i] = av_realloc_array(s->temp[i], s->temp_len, sizeof(short));
	memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
	}

	if (s->output_channels == 2 && s->input_channels == 1) {
	mono_to_stereo(output, buftmp3[0], nb_samples1);
	} else if (s->output_channels == 6 && s->input_channels == 2) {
	ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
	} else if ((s->output_channels == s->input_channels && s->input_channels >= 2) \|\|
	(s->output_channels == 2 && s->input_channels == 6)) {
	interleave(output, buftmp3, s->output_channels, nb_samples1);
	}

	if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
	int istride[1] = { 2 };
	int ostride[1] = { s->sample_size[1] };
	const void *ibuf[1] = { output };
	void *obuf[1] = { output_bak };

	if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
	ibuf, istride, nb_samples1 * s->output_channels) < 0) {
	av_log(s->resample_context, AV_LOG_ERROR,
	"Audio sample format conversion failed\n");
	return 0;
	}
	}

	fail:
	for (i = 0; i < s->filter_channels; i++) {
	av_free(bufin[i]);
	av_free(bufout[i]);
	}

	return nb_samples1;
	}

	void audio_resample_close(ReSampleContext *s)
	{
	int i;
	av_resample_close(s->resample_context);
	for (i = 0; i < s->filter_channels; i++)
	av_freep(&s->temp[i]);
	av_freep(&s->buffer[0]);
	av_freep(&s->buffer[1]);
	av_audio_convert_free(s->convert_ctx[0]);
	av_audio_convert_free(s->convert_ctx[1]);
	av_free(s);
	}

	FF_ENABLE_DEPRECATION_WARNINGS
	#endif