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/*

 * 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 

#include "avcodec.h"

#include "audioconvert.h"

#include "libavutil/opt.h"

#include "libavutil/mem.h"

#include "libavutil/samplefmt.h"

#if FF_API_AVCODEC_RESAMPLE

#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<

                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((nb_samples + s->temp_len) * sizeof(short));

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

        buftmp2[i] = bufin[i] + s->temp_len;

        bufout[i] = av_malloc(lenout * sizeof(short));

    }

    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(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;

        }

    }

    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);

}

#endif