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authortrav90 <travawine@palemoon.org>2018-09-30 10:40:30 -0500
committertrav90 <travawine@palemoon.org>2018-09-30 10:40:30 -0500
commitedc124b92beccd55e5277062e95efb62a8b3ec7b (patch)
tree3486b32f85152ff76b1bee03a8d84b3c34c70a5f /media/ffvpx/libavcodec/resample.c
parent8ba6dd1bd12a3d13f9e2c683216dd8778011a72e (diff)
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[ffvpx] Update ffvp9/ffvp8 to release 4.0.2
Diffstat (limited to 'media/ffvpx/libavcodec/resample.c')
-rw-r--r--media/ffvpx/libavcodec/resample.c439
1 files changed, 0 insertions, 439 deletions
diff --git a/media/ffvpx/libavcodec/resample.c b/media/ffvpx/libavcodec/resample.c
deleted file mode 100644
index 4c5eb9f10..000000000
--- a/media/ffvpx/libavcodec/resample.c
+++ /dev/null
@@ -1,439 +0,0 @@
-/*
- * 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->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