forked from KolibriOS/kolibrios
upload sdk
git-svn-id: svn://kolibrios.org@4349 a494cfbc-eb01-0410-851d-a64ba20cac60
This commit is contained in:
Sergey Semyonov (Serge)
435
contrib/sdk/sources/ffmpeg/libavcodec/resample.c
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435
contrib/sdk/sources/ffmpeg/libavcodec/resample.c
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@@ -0,0 +1,435 @@
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/*
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* samplerate conversion for both audio and video
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* Copyright (c) 2000 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* samplerate conversion for both audio and video
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*/
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#include <string.h>
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#include "avcodec.h"
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#include "audioconvert.h"
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#include "libavutil/opt.h"
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#include "libavutil/mem.h"
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#include "libavutil/samplefmt.h"
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#if FF_API_AVCODEC_RESAMPLE
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#define MAX_CHANNELS 8
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struct AVResampleContext;
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static const char *context_to_name(void *ptr)
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{
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return "audioresample";
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}
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static const AVOption options[] = {{NULL}};
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static const AVClass audioresample_context_class = {
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"ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
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};
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struct ReSampleContext {
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struct AVResampleContext *resample_context;
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short *temp[MAX_CHANNELS];
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int temp_len;
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float ratio;
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/* channel convert */
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int input_channels, output_channels, filter_channels;
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AVAudioConvert *convert_ctx[2];
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enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
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unsigned sample_size[2]; ///< size of one sample in sample_fmt
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short *buffer[2]; ///< buffers used for conversion to S16
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unsigned buffer_size[2]; ///< sizes of allocated buffers
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};
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/* n1: number of samples */
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static void stereo_to_mono(short *output, short *input, int n1)
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{
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short *p, *q;
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int n = n1;
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p = input;
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q = output;
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while (n >= 4) {
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q[0] = (p[0] + p[1]) >> 1;
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q[1] = (p[2] + p[3]) >> 1;
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q[2] = (p[4] + p[5]) >> 1;
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q[3] = (p[6] + p[7]) >> 1;
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q += 4;
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p += 8;
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n -= 4;
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}
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while (n > 0) {
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q[0] = (p[0] + p[1]) >> 1;
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q++;
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p += 2;
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n--;
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}
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}
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/* n1: number of samples */
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static void mono_to_stereo(short *output, short *input, int n1)
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{
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short *p, *q;
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int n = n1;
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int v;
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p = input;
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q = output;
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while (n >= 4) {
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v = p[0]; q[0] = v; q[1] = v;
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v = p[1]; q[2] = v; q[3] = v;
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v = p[2]; q[4] = v; q[5] = v;
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v = p[3]; q[6] = v; q[7] = v;
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q += 8;
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p += 4;
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n -= 4;
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}
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while (n > 0) {
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v = p[0]; q[0] = v; q[1] = v;
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q += 2;
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p += 1;
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n--;
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}
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}
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/*
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5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
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- Left = front_left + rear_gain * rear_left + center_gain * center
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- Right = front_right + rear_gain * rear_right + center_gain * center
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Where rear_gain is usually around 0.5-1.0 and
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center_gain is almost always 0.7 (-3 dB)
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*/
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static void surround_to_stereo(short **output, short *input, int channels, int samples)
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{
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int i;
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short l, r;
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for (i = 0; i < samples; i++) {
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int fl,fr,c,rl,rr;
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fl = input[0];
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fr = input[1];
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c = input[2];
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// lfe = input[3];
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rl = input[4];
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rr = input[5];
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l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
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r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));
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/* output l & r. */
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*output[0]++ = l;
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*output[1]++ = r;
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/* increment input. */
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input += channels;
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}
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}
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static void deinterleave(short **output, short *input, int channels, int samples)
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{
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int i, j;
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for (i = 0; i < samples; i++) {
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for (j = 0; j < channels; j++) {
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*output[j]++ = *input++;
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}
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}
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}
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static void interleave(short *output, short **input, int channels, int samples)
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{
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int i, j;
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for (i = 0; i < samples; i++) {
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for (j = 0; j < channels; j++) {
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*output++ = *input[j]++;
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}
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}
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}
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static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
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{
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int i;
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short l, r;
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for (i = 0; i < n; i++) {
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l = *input1++;
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r = *input2++;
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*output++ = l; /* left */
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*output++ = (l / 2) + (r / 2); /* center */
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*output++ = r; /* right */
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*output++ = 0; /* left surround */
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*output++ = 0; /* right surroud */
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*output++ = 0; /* low freq */
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}
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}
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#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
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ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0
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static const uint8_t supported_resampling[MAX_CHANNELS] = {
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// output ch: 1 2 3 4 5 6 7 8
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SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
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SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
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SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
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SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
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};
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ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
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int output_rate, int input_rate,
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enum AVSampleFormat sample_fmt_out,
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enum AVSampleFormat sample_fmt_in,
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int filter_length, int log2_phase_count,
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int linear, double cutoff)
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{
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ReSampleContext *s;
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if (input_channels > MAX_CHANNELS) {
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av_log(NULL, AV_LOG_ERROR,
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"Resampling with input channels greater than %d is unsupported.\n",
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MAX_CHANNELS);
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return NULL;
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}
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if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
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int i;
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av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
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"output channels for %d input channel%s", input_channels,
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input_channels > 1 ? "s:" : ":");
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for (i = 0; i < MAX_CHANNELS; i++)
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if (supported_resampling[input_channels-1] & (1<<i))
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av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
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av_log(NULL, AV_LOG_ERROR, "\n");
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return NULL;
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}
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s = av_mallocz(sizeof(ReSampleContext));
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if (!s) {
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av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
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return NULL;
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}
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s->ratio = (float)output_rate / (float)input_rate;
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s->input_channels = input_channels;
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s->output_channels = output_channels;
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s->filter_channels = s->input_channels;
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if (s->output_channels < s->filter_channels)
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s->filter_channels = s->output_channels;
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s->sample_fmt[0] = sample_fmt_in;
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s->sample_fmt[1] = sample_fmt_out;
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s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
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s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
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if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
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if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
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s->sample_fmt[0], 1, NULL, 0))) {
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av_log(s, AV_LOG_ERROR,
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"Cannot convert %s sample format to s16 sample format\n",
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av_get_sample_fmt_name(s->sample_fmt[0]));
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av_free(s);
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return NULL;
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}
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}
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if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
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if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
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AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
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av_log(s, AV_LOG_ERROR,
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"Cannot convert s16 sample format to %s sample format\n",
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av_get_sample_fmt_name(s->sample_fmt[1]));
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av_audio_convert_free(s->convert_ctx[0]);
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av_free(s);
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return NULL;
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}
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}
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s->resample_context = av_resample_init(output_rate, input_rate,
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filter_length, log2_phase_count,
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linear, cutoff);
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*(const AVClass**)s->resample_context = &audioresample_context_class;
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return s;
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}
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/* resample audio. 'nb_samples' is the number of input samples */
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/* XXX: optimize it ! */
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int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
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{
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int i, nb_samples1;
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short *bufin[MAX_CHANNELS];
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short *bufout[MAX_CHANNELS];
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short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
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short *output_bak = NULL;
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int lenout;
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if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
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/* nothing to do */
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memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
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return nb_samples;
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}
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if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
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int istride[1] = { s->sample_size[0] };
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int ostride[1] = { 2 };
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const void *ibuf[1] = { input };
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void *obuf[1];
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unsigned input_size = nb_samples * s->input_channels * 2;
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if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
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av_free(s->buffer[0]);
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s->buffer_size[0] = input_size;
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s->buffer[0] = av_malloc(s->buffer_size[0]);
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if (!s->buffer[0]) {
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av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
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return 0;
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}
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}
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obuf[0] = s->buffer[0];
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if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
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ibuf, istride, nb_samples * s->input_channels) < 0) {
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av_log(s->resample_context, AV_LOG_ERROR,
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"Audio sample format conversion failed\n");
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return 0;
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}
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input = s->buffer[0];
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}
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lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
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if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
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int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
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s->output_channels;
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output_bak = output;
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if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
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av_free(s->buffer[1]);
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s->buffer_size[1] = out_size;
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s->buffer[1] = av_malloc(s->buffer_size[1]);
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if (!s->buffer[1]) {
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av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
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return 0;
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}
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}
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output = s->buffer[1];
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}
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/* XXX: move those malloc to resample init code */
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for (i = 0; i < s->filter_channels; i++) {
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bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
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memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
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buftmp2[i] = bufin[i] + s->temp_len;
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bufout[i] = av_malloc(lenout * sizeof(short));
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}
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if (s->input_channels == 2 && s->output_channels == 1) {
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buftmp3[0] = output;
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stereo_to_mono(buftmp2[0], input, nb_samples);
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} else if (s->output_channels >= 2 && s->input_channels == 1) {
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buftmp3[0] = bufout[0];
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memcpy(buftmp2[0], input, nb_samples * sizeof(short));
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} else if (s->input_channels == 6 && s->output_channels ==2) {
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buftmp3[0] = bufout[0];
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buftmp3[1] = bufout[1];
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surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
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} else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
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for (i = 0; i < s->input_channels; i++) {
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buftmp3[i] = bufout[i];
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}
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deinterleave(buftmp2, input, s->input_channels, nb_samples);
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} else {
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buftmp3[0] = output;
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memcpy(buftmp2[0], input, nb_samples * sizeof(short));
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}
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nb_samples += s->temp_len;
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/* resample each channel */
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nb_samples1 = 0; /* avoid warning */
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for (i = 0; i < s->filter_channels; i++) {
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int consumed;
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int is_last = i + 1 == s->filter_channels;
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nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
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&consumed, nb_samples, lenout, is_last);
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s->temp_len = nb_samples - consumed;
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s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
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memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
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}
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if (s->output_channels == 2 && s->input_channels == 1) {
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mono_to_stereo(output, buftmp3[0], nb_samples1);
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} else if (s->output_channels == 6 && s->input_channels == 2) {
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ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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} else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
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(s->output_channels == 2 && s->input_channels == 6)) {
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interleave(output, buftmp3, s->output_channels, nb_samples1);
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}
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if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
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int istride[1] = { 2 };
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int ostride[1] = { s->sample_size[1] };
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const void *ibuf[1] = { output };
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void *obuf[1] = { output_bak };
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if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
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ibuf, istride, nb_samples1 * s->output_channels) < 0) {
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av_log(s->resample_context, AV_LOG_ERROR,
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"Audio sample format conversion failed\n");
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return 0;
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}
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}
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for (i = 0; i < s->filter_channels; i++) {
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av_free(bufin[i]);
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av_free(bufout[i]);
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}
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return nb_samples1;
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}
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void audio_resample_close(ReSampleContext *s)
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{
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int i;
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av_resample_close(s->resample_context);
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for (i = 0; i < s->filter_channels; i++)
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av_freep(&s->temp[i]);
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av_freep(&s->buffer[0]);
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av_freep(&s->buffer[1]);
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av_audio_convert_free(s->convert_ctx[0]);
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av_audio_convert_free(s->convert_ctx[1]);
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av_free(s);
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}
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#endif
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