yading@10: /*
yading@10:  * samplerate conversion for both audio and video
yading@10:  * Copyright (c) 2000 Fabrice Bellard
yading@10:  *
yading@10:  * This file is part of FFmpeg.
yading@10:  *
yading@10:  * FFmpeg is free software; you can redistribute it and/or
yading@10:  * modify it under the terms of the GNU Lesser General Public
yading@10:  * License as published by the Free Software Foundation; either
yading@10:  * version 2.1 of the License, or (at your option) any later version.
yading@10:  *
yading@10:  * FFmpeg is distributed in the hope that it will be useful,
yading@10:  * but WITHOUT ANY WARRANTY; without even the implied warranty of
yading@10:  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
yading@10:  * Lesser General Public License for more details.
yading@10:  *
yading@10:  * You should have received a copy of the GNU Lesser General Public
yading@10:  * License along with FFmpeg; if not, write to the Free Software
yading@10:  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
yading@10:  */
yading@10: 
yading@10: /**
yading@10:  * @file
yading@10:  * samplerate conversion for both audio and video
yading@10:  */
yading@10: 
yading@10: #include <string.h>
yading@10: 
yading@10: #include "avcodec.h"
yading@10: #include "audioconvert.h"
yading@10: #include "libavutil/opt.h"
yading@10: #include "libavutil/mem.h"
yading@10: #include "libavutil/samplefmt.h"
yading@10: 
yading@10: #if FF_API_AVCODEC_RESAMPLE
yading@10: 
yading@10: #define MAX_CHANNELS 8
yading@10: 
yading@10: struct AVResampleContext;
yading@10: 
yading@10: static const char *context_to_name(void *ptr)
yading@10: {
yading@10:     return "audioresample";
yading@10: }
yading@10: 
yading@10: static const AVOption options[] = {{NULL}};
yading@10: static const AVClass audioresample_context_class = {
yading@10:     "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
yading@10: };
yading@10: 
yading@10: struct ReSampleContext {
yading@10:     struct AVResampleContext *resample_context;
yading@10:     short *temp[MAX_CHANNELS];
yading@10:     int temp_len;
yading@10:     float ratio;
yading@10:     /* channel convert */
yading@10:     int input_channels, output_channels, filter_channels;
yading@10:     AVAudioConvert *convert_ctx[2];
yading@10:     enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
yading@10:     unsigned sample_size[2];           ///< size of one sample in sample_fmt
yading@10:     short *buffer[2];                  ///< buffers used for conversion to S16
yading@10:     unsigned buffer_size[2];           ///< sizes of allocated buffers
yading@10: };
yading@10: 
yading@10: /* n1: number of samples */
yading@10: static void stereo_to_mono(short *output, short *input, int n1)
yading@10: {
yading@10:     short *p, *q;
yading@10:     int n = n1;
yading@10: 
yading@10:     p = input;
yading@10:     q = output;
yading@10:     while (n >= 4) {
yading@10:         q[0] = (p[0] + p[1]) >> 1;
yading@10:         q[1] = (p[2] + p[3]) >> 1;
yading@10:         q[2] = (p[4] + p[5]) >> 1;
yading@10:         q[3] = (p[6] + p[7]) >> 1;
yading@10:         q += 4;
yading@10:         p += 8;
yading@10:         n -= 4;
yading@10:     }
yading@10:     while (n > 0) {
yading@10:         q[0] = (p[0] + p[1]) >> 1;
yading@10:         q++;
yading@10:         p += 2;
yading@10:         n--;
yading@10:     }
yading@10: }
yading@10: 
yading@10: /* n1: number of samples */
yading@10: static void mono_to_stereo(short *output, short *input, int n1)
yading@10: {
yading@10:     short *p, *q;
yading@10:     int n = n1;
yading@10:     int v;
yading@10: 
yading@10:     p = input;
yading@10:     q = output;
yading@10:     while (n >= 4) {
yading@10:         v = p[0]; q[0] = v; q[1] = v;
yading@10:         v = p[1]; q[2] = v; q[3] = v;
yading@10:         v = p[2]; q[4] = v; q[5] = v;
yading@10:         v = p[3]; q[6] = v; q[7] = v;
yading@10:         q += 8;
yading@10:         p += 4;
yading@10:         n -= 4;
yading@10:     }
yading@10:     while (n > 0) {
yading@10:         v = p[0]; q[0] = v; q[1] = v;
yading@10:         q += 2;
yading@10:         p += 1;
yading@10:         n--;
yading@10:     }
yading@10: }
yading@10: 
yading@10: /*
yading@10: 5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
yading@10: - Left = front_left + rear_gain * rear_left + center_gain * center
yading@10: - Right = front_right + rear_gain * rear_right + center_gain * center
yading@10: Where rear_gain is usually around 0.5-1.0 and
yading@10:       center_gain is almost always 0.7 (-3 dB)
yading@10: */
yading@10: static void surround_to_stereo(short **output, short *input, int channels, int samples)
yading@10: {
yading@10:     int i;
yading@10:     short l, r;
yading@10: 
yading@10:     for (i = 0; i < samples; i++) {
yading@10:         int fl,fr,c,rl,rr;
yading@10:         fl = input[0];
yading@10:         fr = input[1];
yading@10:         c = input[2];
yading@10:         // lfe = input[3];
yading@10:         rl = input[4];
yading@10:         rr = input[5];
yading@10: 
yading@10:         l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
yading@10:         r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));
yading@10: 
yading@10:         /* output l & r. */
yading@10:         *output[0]++ = l;
yading@10:         *output[1]++ = r;
yading@10: 
yading@10:         /* increment input. */
yading@10:         input += channels;
yading@10:     }
yading@10: }
yading@10: 
yading@10: static void deinterleave(short **output, short *input, int channels, int samples)
yading@10: {
yading@10:     int i, j;
yading@10: 
yading@10:     for (i = 0; i < samples; i++) {
yading@10:         for (j = 0; j < channels; j++) {
yading@10:             *output[j]++ = *input++;
yading@10:         }
yading@10:     }
yading@10: }
yading@10: 
yading@10: static void interleave(short *output, short **input, int channels, int samples)
yading@10: {
yading@10:     int i, j;
yading@10: 
yading@10:     for (i = 0; i < samples; i++) {
yading@10:         for (j = 0; j < channels; j++) {
yading@10:             *output++ = *input[j]++;
yading@10:         }
yading@10:     }
yading@10: }
yading@10: 
yading@10: static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
yading@10: {
yading@10:     int i;
yading@10:     short l, r;
yading@10: 
yading@10:     for (i = 0; i < n; i++) {
yading@10:         l = *input1++;
yading@10:         r = *input2++;
yading@10:         *output++ = l;                  /* left */
yading@10:         *output++ = (l / 2) + (r / 2);  /* center */
yading@10:         *output++ = r;                  /* right */
yading@10:         *output++ = 0;                  /* left surround */
yading@10:         *output++ = 0;                  /* right surroud */
yading@10:         *output++ = 0;                  /* low freq */
yading@10:     }
yading@10: }
yading@10: 
yading@10: #define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
yading@10:     ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0
yading@10: 
yading@10: static const uint8_t supported_resampling[MAX_CHANNELS] = {
yading@10:     // output ch:    1  2  3  4  5  6  7  8
yading@10:     SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
yading@10:     SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
yading@10:     SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
yading@10:     SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
yading@10:     SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
yading@10:     SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
yading@10:     SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
yading@10:     SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
yading@10: };
yading@10: 
yading@10: ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
yading@10:                                         int output_rate, int input_rate,
yading@10:                                         enum AVSampleFormat sample_fmt_out,
yading@10:                                         enum AVSampleFormat sample_fmt_in,
yading@10:                                         int filter_length, int log2_phase_count,
yading@10:                                         int linear, double cutoff)
yading@10: {
yading@10:     ReSampleContext *s;
yading@10: 
yading@10:     if (input_channels > MAX_CHANNELS) {
yading@10:         av_log(NULL, AV_LOG_ERROR,
yading@10:                "Resampling with input channels greater than %d is unsupported.\n",
yading@10:                MAX_CHANNELS);
yading@10:         return NULL;
yading@10:     }
yading@10:     if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
yading@10:         int i;
yading@10:         av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
yading@10:                "output channels for %d input channel%s", input_channels,
yading@10:                input_channels > 1 ? "s:" : ":");
yading@10:         for (i = 0; i < MAX_CHANNELS; i++)
yading@10:             if (supported_resampling[input_channels-1] & (1<<i))
yading@10:                 av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
yading@10:         av_log(NULL, AV_LOG_ERROR, "\n");
yading@10:         return NULL;
yading@10:     }
yading@10: 
yading@10:     s = av_mallocz(sizeof(ReSampleContext));
yading@10:     if (!s) {
yading@10:         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
yading@10:         return NULL;
yading@10:     }
yading@10: 
yading@10:     s->ratio = (float)output_rate / (float)input_rate;
yading@10: 
yading@10:     s->input_channels = input_channels;
yading@10:     s->output_channels = output_channels;
yading@10: 
yading@10:     s->filter_channels = s->input_channels;
yading@10:     if (s->output_channels < s->filter_channels)
yading@10:         s->filter_channels = s->output_channels;
yading@10: 
yading@10:     s->sample_fmt[0]  = sample_fmt_in;
yading@10:     s->sample_fmt[1]  = sample_fmt_out;
yading@10:     s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
yading@10:     s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
yading@10: 
yading@10:     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
yading@10:         if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
yading@10:                                                          s->sample_fmt[0], 1, NULL, 0))) {
yading@10:             av_log(s, AV_LOG_ERROR,
yading@10:                    "Cannot convert %s sample format to s16 sample format\n",
yading@10:                    av_get_sample_fmt_name(s->sample_fmt[0]));
yading@10:             av_free(s);
yading@10:             return NULL;
yading@10:         }
yading@10:     }
yading@10: 
yading@10:     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
yading@10:         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
yading@10:                                                          AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
yading@10:             av_log(s, AV_LOG_ERROR,
yading@10:                    "Cannot convert s16 sample format to %s sample format\n",
yading@10:                    av_get_sample_fmt_name(s->sample_fmt[1]));
yading@10:             av_audio_convert_free(s->convert_ctx[0]);
yading@10:             av_free(s);
yading@10:             return NULL;
yading@10:         }
yading@10:     }
yading@10: 
yading@10:     s->resample_context = av_resample_init(output_rate, input_rate,
yading@10:                                            filter_length, log2_phase_count,
yading@10:                                            linear, cutoff);
yading@10: 
yading@10:     *(const AVClass**)s->resample_context = &audioresample_context_class;
yading@10: 
yading@10:     return s;
yading@10: }
yading@10: 
yading@10: /* resample audio. 'nb_samples' is the number of input samples */
yading@10: /* XXX: optimize it ! */
yading@10: int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
yading@10: {
yading@10:     int i, nb_samples1;
yading@10:     short *bufin[MAX_CHANNELS];
yading@10:     short *bufout[MAX_CHANNELS];
yading@10:     short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
yading@10:     short *output_bak = NULL;
yading@10:     int lenout;
yading@10: 
yading@10:     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
yading@10:         /* nothing to do */
yading@10:         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
yading@10:         return nb_samples;
yading@10:     }
yading@10: 
yading@10:     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
yading@10:         int istride[1] = { s->sample_size[0] };
yading@10:         int ostride[1] = { 2 };
yading@10:         const void *ibuf[1] = { input };
yading@10:         void       *obuf[1];
yading@10:         unsigned input_size = nb_samples * s->input_channels * 2;
yading@10: 
yading@10:         if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
yading@10:             av_free(s->buffer[0]);
yading@10:             s->buffer_size[0] = input_size;
yading@10:             s->buffer[0] = av_malloc(s->buffer_size[0]);
yading@10:             if (!s->buffer[0]) {
yading@10:                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
yading@10:                 return 0;
yading@10:             }
yading@10:         }
yading@10: 
yading@10:         obuf[0] = s->buffer[0];
yading@10: 
yading@10:         if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
yading@10:                              ibuf, istride, nb_samples * s->input_channels) < 0) {
yading@10:             av_log(s->resample_context, AV_LOG_ERROR,
yading@10:                    "Audio sample format conversion failed\n");
yading@10:             return 0;
yading@10:         }
yading@10: 
yading@10:         input = s->buffer[0];
yading@10:     }
yading@10: 
yading@10:     lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
yading@10: 
yading@10:     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
yading@10:         int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
yading@10:                        s->output_channels;
yading@10:         output_bak = output;
yading@10: 
yading@10:         if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
yading@10:             av_free(s->buffer[1]);
yading@10:             s->buffer_size[1] = out_size;
yading@10:             s->buffer[1] = av_malloc(s->buffer_size[1]);
yading@10:             if (!s->buffer[1]) {
yading@10:                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
yading@10:                 return 0;
yading@10:             }
yading@10:         }
yading@10: 
yading@10:         output = s->buffer[1];
yading@10:     }
yading@10: 
yading@10:     /* XXX: move those malloc to resample init code */
yading@10:     for (i = 0; i < s->filter_channels; i++) {
yading@10:         bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
yading@10:         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
yading@10:         buftmp2[i] = bufin[i] + s->temp_len;
yading@10:         bufout[i] = av_malloc(lenout * sizeof(short));
yading@10:     }
yading@10: 
yading@10:     if (s->input_channels == 2 && s->output_channels == 1) {
yading@10:         buftmp3[0] = output;
yading@10:         stereo_to_mono(buftmp2[0], input, nb_samples);
yading@10:     } else if (s->output_channels >= 2 && s->input_channels == 1) {
yading@10:         buftmp3[0] = bufout[0];
yading@10:         memcpy(buftmp2[0], input, nb_samples * sizeof(short));
yading@10:     } else if (s->input_channels == 6 && s->output_channels ==2) {
yading@10:         buftmp3[0] = bufout[0];
yading@10:         buftmp3[1] = bufout[1];
yading@10:         surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
yading@10:     } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
yading@10:         for (i = 0; i < s->input_channels; i++) {
yading@10:             buftmp3[i] = bufout[i];
yading@10:         }
yading@10:         deinterleave(buftmp2, input, s->input_channels, nb_samples);
yading@10:     } else {
yading@10:         buftmp3[0] = output;
yading@10:         memcpy(buftmp2[0], input, nb_samples * sizeof(short));
yading@10:     }
yading@10: 
yading@10:     nb_samples += s->temp_len;
yading@10: 
yading@10:     /* resample each channel */
yading@10:     nb_samples1 = 0; /* avoid warning */
yading@10:     for (i = 0; i < s->filter_channels; i++) {
yading@10:         int consumed;
yading@10:         int is_last = i + 1 == s->filter_channels;
yading@10: 
yading@10:         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
yading@10:                                   &consumed, nb_samples, lenout, is_last);
yading@10:         s->temp_len = nb_samples - consumed;
yading@10:         s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
yading@10:         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
yading@10:     }
yading@10: 
yading@10:     if (s->output_channels == 2 && s->input_channels == 1) {
yading@10:         mono_to_stereo(output, buftmp3[0], nb_samples1);
yading@10:     } else if (s->output_channels == 6 && s->input_channels == 2) {
yading@10:         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
yading@10:     } else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
yading@10:                (s->output_channels == 2 && s->input_channels == 6)) {
yading@10:         interleave(output, buftmp3, s->output_channels, nb_samples1);
yading@10:     }
yading@10: 
yading@10:     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
yading@10:         int istride[1] = { 2 };
yading@10:         int ostride[1] = { s->sample_size[1] };
yading@10:         const void *ibuf[1] = { output };
yading@10:         void       *obuf[1] = { output_bak };
yading@10: 
yading@10:         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
yading@10:                              ibuf, istride, nb_samples1 * s->output_channels) < 0) {
yading@10:             av_log(s->resample_context, AV_LOG_ERROR,
yading@10:                    "Audio sample format conversion failed\n");
yading@10:             return 0;
yading@10:         }
yading@10:     }
yading@10: 
yading@10:     for (i = 0; i < s->filter_channels; i++) {
yading@10:         av_free(bufin[i]);
yading@10:         av_free(bufout[i]);
yading@10:     }
yading@10: 
yading@10:     return nb_samples1;
yading@10: }
yading@10: 
yading@10: void audio_resample_close(ReSampleContext *s)
yading@10: {
yading@10:     int i;
yading@10:     av_resample_close(s->resample_context);
yading@10:     for (i = 0; i < s->filter_channels; i++)
yading@10:         av_freep(&s->temp[i]);
yading@10:     av_freep(&s->buffer[0]);
yading@10:     av_freep(&s->buffer[1]);
yading@10:     av_audio_convert_free(s->convert_ctx[0]);
yading@10:     av_audio_convert_free(s->convert_ctx[1]);
yading@10:     av_free(s);
yading@10: }
yading@10: 
yading@10: #endif