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FFmpeg/libavresample/audio_mix.c
Justin Ruggles 157542ebc1 lavr: fix mixing matrix reduction when normalization is disabled
In some cases when an input contributes fully to the corresponding
output, other inputs may also contribute to the same output. This is the
case, for example, for the default 5.1 to stereo downmix matrix without
normalization.
2013-02-12 13:41:13 -05:00

740 lines
26 KiB
C

/*
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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 *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 * m1;
*dst1++ = v * m0;
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
len -= 4;
}
while (len > 0) {
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
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 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_dlog(am->avr, "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];
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 (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]) \
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_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;
}