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FFmpeg/libavresample/audio_mix.c
Michael Niedermayer 40d552dae6 Merge commit '1a3eb042c704dea190c644def5b32c9cee8832b8' 
* commit '1a3eb042c704dea190c644def5b32c9cee8832b8':
  Replace av_dlog with normal av_log at trace level

Conflicts:
	ffplay.c
	libavdevice/fbdev_dec.c
	libavfilter/avfilter.c
	libavfilter/internal.h
	libavfilter/setpts.c
	libavfilter/src_movie.c
	libavfilter/vf_crop.c
	libavfilter/vf_drawtext.c
	libavfilter/vf_fieldorder.c
	libavformat/assdec.c
	libavformat/avidec.c
	libavformat/flvdec.c
	libavformat/http.c
	libavformat/ipmovie.c
	libavformat/isom.c
	libavformat/mov.c
	libavformat/mpegenc.c
	libavformat/mpegts.c
	libavformat/mpegtsenc.c
	libavformat/mux.c
	libavformat/mxfdec.c
	libavformat/nsvdec.c
	libavformat/oggdec.c
	libavformat/r3d.c
	libavformat/rtspdec.c
	libavformat/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2015-04-20 03:19:47 +02:00

743 lines
26 KiB
C
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/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* 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
*/
#include <stdint.h>
#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_mix.h"
static const char * const coeff_type_names[] = { "q8", "q15", "flt" };
struct AudioMix {
AVAudioResampleContext *avr;
enum AVSampleFormat fmt;
enum AVMixCoeffType coeff_type;
uint64_t in_layout;
uint64_t out_layout;
int in_channels;
int out_channels;
int ptr_align;
int samples_align;
int has_optimized_func;
const char *func_descr;
const char *func_descr_generic;
mix_func *mix;
mix_func *mix_generic;
int in_matrix_channels;
int out_matrix_channels;
int output_zero[AVRESAMPLE_MAX_CHANNELS];
int input_skip[AVRESAMPLE_MAX_CHANNELS];
int output_skip[AVRESAMPLE_MAX_CHANNELS];
int16_t *matrix_q8[AVRESAMPLE_MAX_CHANNELS];
int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS];
float *matrix_flt[AVRESAMPLE_MAX_CHANNELS];
void **matrix;
};
void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
enum AVMixCoeffType coeff_type, int in_channels,
int out_channels, int ptr_align, int samples_align,
const char *descr, void *mix_func)
{
if (fmt == am->fmt && coeff_type == am->coeff_type &&
( in_channels == am->in_matrix_channels || in_channels == 0) &&
(out_channels == am->out_matrix_channels || out_channels == 0)) {
char chan_str[16];
am->mix = mix_func;
am->func_descr = descr;
am->ptr_align = ptr_align;
am->samples_align = samples_align;
if (ptr_align == 1 && samples_align == 1) {
am->mix_generic = mix_func;
am->func_descr_generic = descr;
} else {
am->has_optimized_func = 1;
}
if (in_channels) {
if (out_channels)
snprintf(chan_str, sizeof(chan_str), "[%d to %d] ",
in_channels, out_channels);
else
snprintf(chan_str, sizeof(chan_str), "[%d to any] ",
in_channels);
} else if (out_channels) {
snprintf(chan_str, sizeof(chan_str), "[any to %d] ",
out_channels);
} else {
snprintf(chan_str, sizeof(chan_str), "[any to any] ");
}
av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] "
"[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt),
coeff_type_names[coeff_type], chan_str, descr);
}
}
#define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c
#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \
static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \
int len, int out_ch, int in_ch) \
{ \
int i, in, out; \
stype temp[AVRESAMPLE_MAX_CHANNELS]; \
for (i = 0; i < len; i++) { \
for (out = 0; out < out_ch; out++) { \
sumtype sum = 0; \
for (in = 0; in < in_ch; in++) \
sum += samples[in][i] * matrix[out][in]; \
temp[out] = expr; \
} \
for (out = 0; out < out_ch; out++) \
samples[out][i] = temp[out]; \
} \
}
MIX_FUNC_GENERIC(FLTP, FLT, float, float, float, sum)
MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum)))
MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15))
MIX_FUNC_GENERIC(S16P, Q8, int16_t, int16_t, int32_t, av_clip_int16(sum >> 8))
/* TODO: templatize the channel-specific C functions */
static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float *src0 = samples[0];
float *src1 = samples[1];
float *dst = src0;
float m0 = matrix[0][0];
float m1 = matrix[0][1];
while (len > 4) {
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
len -= 4;
}
while (len > 0) {
*dst++ = *src0++ * m0 + *src1++ * m1;
len--;
}
}
static void mix_2_to_1_s16p_flt_c(int16_t **samples, float **matrix, int len,
int out_ch, int in_ch)
{
int16_t *src0 = samples[0];
int16_t *src1 = samples[1];
int16_t *dst = src0;
float m0 = matrix[0][0];
float m1 = matrix[0][1];
while (len > 4) {
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
len -= 4;
}
while (len > 0) {
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
len--;
}
}
static void mix_2_to_1_s16p_q8_c(int16_t **samples, int16_t **matrix, int len,
int out_ch, int in_ch)
{
int16_t *src0 = samples[0];
int16_t *src1 = samples[1];
int16_t *dst = src0;
int16_t m0 = matrix[0][0];
int16_t m1 = matrix[0][1];
while (len > 4) {
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
len -= 4;
}
while (len > 0) {
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
len--;
}
}
static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v;
float *dst0 = samples[0];
float *dst1 = samples[1];
float *src = dst0;
float m0 = matrix[0][0];
float m1 = matrix[1][0];
while (len > 4) {
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
len -= 4;
}
while (len > 0) {
v = *src++;
*dst0++ = v * m0;
*dst1++ = v * m1;
len--;
}
}
static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v0, v1;
float *src0 = samples[0];
float *src1 = samples[1];
float *src2 = samples[2];
float *src3 = samples[3];
float *src4 = samples[4];
float *src5 = samples[5];
float *dst0 = src0;
float *dst1 = src1;
float *m0 = matrix[0];
float *m1 = matrix[1];
while (len > 0) {
v0 = *src0++;
v1 = *src1++;
*dst0++ = v0 * m0[0] +
v1 * m0[1] +
*src2 * m0[2] +
*src3 * m0[3] +
*src4 * m0[4] +
*src5 * m0[5];
*dst1++ = v0 * m1[0] +
v1 * m1[1] +
*src2++ * m1[2] +
*src3++ * m1[3] +
*src4++ * m1[4] +
*src5++ * m1[5];
len--;
}
}
static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v0, v1;
float *dst0 = samples[0];
float *dst1 = samples[1];
float *dst2 = samples[2];
float *dst3 = samples[3];
float *dst4 = samples[4];
float *dst5 = samples[5];
float *src0 = dst0;
float *src1 = dst1;
while (len > 0) {
v0 = *src0++;
v1 = *src1++;
*dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1];
*dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1];
*dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1];
*dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1];
*dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1];
*dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1];
len--;
}
}
static av_cold int mix_function_init(AudioMix *am)
{
am->func_descr = am->func_descr_generic = "n/a";
am->mix = am->mix_generic = NULL;
/* no need to set a mix function when we're skipping mixing */
if (!am->in_matrix_channels || !am->out_matrix_channels)
return 0;
/* any-to-any C versions */
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP, FLT));
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, FLT));
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q15,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q15));
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q8));
/* channel-specific C versions */
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 1, 1, "C", mix_2_to_1_s16p_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
2, 1, 1, 1, "C", mix_2_to_1_s16p_q8_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c);
if (ARCH_X86)
ff_audio_mix_init_x86(am);
if (!am->mix) {
av_log(am->avr, AV_LOG_ERROR, "audio_mix: NO FUNCTION FOUND: [fmt=%s] "
"[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am->fmt),
coeff_type_names[am->coeff_type], am->in_channels,
am->out_channels);
return AVERROR_PATCHWELCOME;
}
return 0;
}
AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr)
{
AudioMix *am;
int ret;
am = av_mallocz(sizeof(*am));
if (!am)
return NULL;
am->avr = avr;
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"mixing: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
goto error;
}
am->fmt = avr->internal_sample_fmt;
am->coeff_type = avr->mix_coeff_type;
am->in_layout = avr->in_channel_layout;
am->out_layout = avr->out_channel_layout;
am->in_channels = avr->in_channels;
am->out_channels = avr->out_channels;
/* build matrix if the user did not already set one */
if (avr->mix_matrix) {
ret = ff_audio_mix_set_matrix(am, avr->mix_matrix, avr->in_channels);
if (ret < 0)
goto error;
av_freep(&avr->mix_matrix);
} else {
double *matrix_dbl = av_mallocz(avr->out_channels * avr->in_channels *
sizeof(*matrix_dbl));
if (!matrix_dbl)
goto error;
ret = avresample_build_matrix(avr->in_channel_layout,
avr->out_channel_layout,
avr->center_mix_level,
avr->surround_mix_level,
avr->lfe_mix_level,
avr->normalize_mix_level,
matrix_dbl,
avr->in_channels,
avr->matrix_encoding);
if (ret < 0) {
av_free(matrix_dbl);
goto error;
}
ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels);
if (ret < 0) {
av_log(avr, AV_LOG_ERROR, "error setting mix matrix\n");
av_free(matrix_dbl);
goto error;
}
av_free(matrix_dbl);
}
return am;
error:
av_free(am);
return NULL;
}
void ff_audio_mix_free(AudioMix **am_p)
{
AudioMix *am;
if (!*am_p)
return;
am = *am_p;
if (am->matrix) {
av_free(am->matrix[0]);
am->matrix = NULL;
}
memset(am->matrix_q8, 0, sizeof(am->matrix_q8 ));
memset(am->matrix_q15, 0, sizeof(am->matrix_q15));
memset(am->matrix_flt, 0, sizeof(am->matrix_flt));
av_freep(am_p);
}
int ff_audio_mix(AudioMix *am, AudioData *src)
{
int use_generic = 1;
int len = src->nb_samples;
int i, j;
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
if (am->has_optimized_func) {
int aligned_len = FFALIGN(len, am->samples_align);
if (!(src->ptr_align % am->ptr_align) &&
src->samples_align >= aligned_len) {
len = aligned_len;
use_generic = 0;
}
}
av_log(am->avr, AV_LOG_TRACE, "audio_mix: %d samples - %d to %d channels (%s)\n",
src->nb_samples, am->in_channels, am->out_channels,
use_generic ? am->func_descr_generic : am->func_descr);
if (am->in_matrix_channels && am->out_matrix_channels) {
uint8_t **data;
uint8_t *data0[AVRESAMPLE_MAX_CHANNELS] = { NULL };
if (am->out_matrix_channels < am->out_channels ||
am->in_matrix_channels < am->in_channels) {
for (i = 0, j = 0; i < FFMAX(am->in_channels, am->out_channels); i++) {
if (am->input_skip[i] || am->output_skip[i] || am->output_zero[i])
continue;
data0[j++] = src->data[i];
}
data = data0;
} else {
data = src->data;
}
if (use_generic)
am->mix_generic(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
else
am->mix(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
}
if (am->out_matrix_channels < am->out_channels) {
for (i = 0; i < am->out_channels; i++)
if (am->output_zero[i])
av_samples_set_silence(&src->data[i], 0, len, 1, am->fmt);
}
ff_audio_data_set_channels(src, am->out_channels);
return 0;
}
int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
{
int i, o, i0, o0;
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n");
return AVERROR(EINVAL);
}
#define GET_MATRIX_CONVERT(suffix, scale) \
if (!am->matrix_ ## suffix[0]) { \
av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \
return AVERROR(EINVAL); \
} \
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
if (am->input_skip[i] || am->output_zero[o]) \
matrix[o * stride + i] = 0.0; \
else \
matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \
(scale); \
if (!am->input_skip[i]) \
i0++; \
} \
if (!am->output_zero[o]) \
o0++; \
}
switch (am->coeff_type) {
case AV_MIX_COEFF_TYPE_Q8:
GET_MATRIX_CONVERT(q8, 1.0 / 256.0);
break;
case AV_MIX_COEFF_TYPE_Q15:
GET_MATRIX_CONVERT(q15, 1.0 / 32768.0);
break;
case AV_MIX_COEFF_TYPE_FLT:
GET_MATRIX_CONVERT(flt, 1.0);
break;
default:
av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
return 0;
}
static void reduce_matrix(AudioMix *am, const double *matrix, int stride)
{
int i, o;
memset(am->output_zero, 0, sizeof(am->output_zero));
memset(am->input_skip, 0, sizeof(am->input_skip));
memset(am->output_skip, 0, sizeof(am->output_skip));
/* exclude output channels if they can be zeroed instead of mixed */
for (o = 0; o < am->out_channels; o++) {
int zero = 1;
/* check if the output is always silent */
for (i = 0; i < am->in_channels; i++) {
if (matrix[o * stride + i] != 0.0) {
zero = 0;
break;
}
}
/* check if the corresponding input channel makes a contribution to
any output channel */
if (o < am->in_channels) {
for (i = 0; i < am->out_channels; i++) {
if (matrix[i * stride + o] != 0.0) {
zero = 0;
break;
}
}
}
if (zero) {
am->output_zero[o] = 1;
am->out_matrix_channels--;
if (o < am->in_channels)
am->in_matrix_channels--;
}
}
if (am->out_matrix_channels == 0 || am->in_matrix_channels == 0) {
am->out_matrix_channels = 0;
am->in_matrix_channels = 0;
return;
}
/* skip input channels that contribute fully only to the corresponding
output channel */
for (i = 0; i < FFMIN(am->in_channels, am->out_channels); i++) {
int skip = 1;
for (o = 0; o < am->out_channels; o++) {
int i0;
if ((o != i && matrix[o * stride + i] != 0.0) ||
(o == i && matrix[o * stride + i] != 1.0)) {
skip = 0;
break;
}
/* if the input contributes fully to the output, also check that no
other inputs contribute to this output */
if (o == i) {
for (i0 = 0; i0 < am->in_channels; i0++) {
if (i0 != i && matrix[o * stride + i0] != 0.0) {
skip = 0;
break;
}
}
}
}
if (skip) {
am->input_skip[i] = 1;
am->in_matrix_channels--;
}
}
/* skip input channels that do not contribute to any output channel */
for (; i < am->in_channels; i++) {
int contrib = 0;
for (o = 0; o < am->out_channels; o++) {
if (matrix[o * stride + i] != 0.0) {
contrib = 1;
break;
}
}
if (!contrib) {
am->input_skip[i] = 1;
am->in_matrix_channels--;
}
}
if (am->in_matrix_channels == 0) {
am->out_matrix_channels = 0;
return;
}
/* skip output channels that only get full contribution from the
corresponding input channel */
for (o = 0; o < FFMIN(am->in_channels, am->out_channels); o++) {
int skip = 1;
int o0;
for (i = 0; i < am->in_channels; i++) {
if ((o != i && matrix[o * stride + i] != 0.0) ||
(o == i && matrix[o * stride + i] != 1.0)) {
skip = 0;
break;
}
}
/* check if the corresponding input channel makes a contribution to
any other output channel */
i = o;
for (o0 = 0; o0 < am->out_channels; o0++) {
if (o0 != i && matrix[o0 * stride + i] != 0.0) {
skip = 0;
break;
}
}
if (skip) {
am->output_skip[o] = 1;
am->out_matrix_channels--;
}
}
if (am->out_matrix_channels == 0) {
am->in_matrix_channels = 0;
return;
}
}
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
{
int i, o, i0, o0, ret;
char in_layout_name[128];
char out_layout_name[128];
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n");
return AVERROR(EINVAL);
}
if (am->matrix) {
av_free(am->matrix[0]);
am->matrix = NULL;
}
am->in_matrix_channels = am->in_channels;
am->out_matrix_channels = am->out_channels;
reduce_matrix(am, matrix, stride);
#define CONVERT_MATRIX(type, expr) \
am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \
am->in_matrix_channels * \
sizeof(*am->matrix_## type[0])); \
if (!am->matrix_## type[0]) \
return AVERROR(ENOMEM); \
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
if (am->output_zero[o] || am->output_skip[o]) \
continue; \
if (o0 > 0) \
am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \
am->in_matrix_channels; \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
double v; \
if (am->input_skip[i] || am->output_zero[i]) \
continue; \
v = matrix[o * stride + i]; \
am->matrix_## type[o0][i0] = expr; \
i0++; \
} \
o0++; \
} \
am->matrix = (void **)am->matrix_## type;
if (am->in_matrix_channels && am->out_matrix_channels) {
switch (am->coeff_type) {
case AV_MIX_COEFF_TYPE_Q8:
CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v)))
break;
case AV_MIX_COEFF_TYPE_Q15:
CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v)))
break;
case AV_MIX_COEFF_TYPE_FLT:
CONVERT_MATRIX(flt, v)
break;
default:
av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
}
ret = mix_function_init(am);
if (ret < 0)
return ret;
av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name),
am->in_channels, am->in_layout);
av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name),
am->out_channels, am->out_layout);
av_log(am->avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n",
in_layout_name, out_layout_name);
av_log(am->avr, AV_LOG_DEBUG, "matrix size: %d x %d\n",
am->in_matrix_channels, am->out_matrix_channels);
for (o = 0; o < am->out_channels; o++) {
for (i = 0; i < am->in_channels; i++) {
if (am->output_zero[o])
av_log(am->avr, AV_LOG_DEBUG, " (ZERO)");
else if (am->input_skip[i] || am->output_zero[i] || am->output_skip[o])
av_log(am->avr, AV_LOG_DEBUG, " (SKIP)");
else
av_log(am->avr, AV_LOG_DEBUG, " %0.3f ",
matrix[o * am->in_channels + i]);
}
av_log(am->avr, AV_LOG_DEBUG, "\n");
}
return 0;
}
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