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ffmpeg/libavresample/audio_mix.c

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/*
* 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;
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_channels || in_channels == 0) &&
(out_channels == am->out_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);
}
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],
(in_channels || out_channels) ? 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)
{
/* 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 {
int i, j;
char in_layout_name[128];
char out_layout_name[128];
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;
}
av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name),
avr->in_channels, avr->in_channel_layout);
av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name),
avr->out_channels, avr->out_channel_layout);
av_log(avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n",
in_layout_name, out_layout_name);
for (i = 0; i < avr->out_channels; i++) {
for (j = 0; j < avr->in_channels; j++) {
av_log(avr, AV_LOG_DEBUG, " %0.3f ",
matrix_dbl[i * avr->in_channels + j]);
}
av_log(avr, AV_LOG_DEBUG, "\n");
}
ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels);
if (ret < 0) {
av_free(matrix_dbl);
goto error;
}
av_free(matrix_dbl);
}
ret = mix_function_init(am);
if (ret < 0)
goto error;
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;
/* 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 (use_generic)
am->mix_generic(src->data, am->matrix, len, am->out_channels,
am->in_channels);
else
am->mix(src->data, am->matrix, len, am->out_channels, am->in_channels);
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;
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(am, AV_LOG_ERROR, "Invalid channel counts\n");
return AVERROR(EINVAL);
}
#define GET_MATRIX_CONVERT(suffix, scale) \
if (!am->matrix_ ## suffix[0]) { \
av_log(am, AV_LOG_ERROR, "matrix is not set\n"); \
return AVERROR(EINVAL); \
} \
for (o = 0; o < am->out_channels; o++) \
for (i = 0; i < am->in_channels; i++) \
matrix[o * stride + i] = am->matrix_ ## suffix[o][i] * (scale);
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, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
return 0;
}
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
{
int i, o;
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(am, AV_LOG_ERROR, "Invalid channel counts\n");
return AVERROR(EINVAL);
}
if (am->matrix) {
av_free(am->matrix[0]);
am->matrix = NULL;
}
#define CONVERT_MATRIX(type, expr) \
am->matrix_## type[0] = av_mallocz(am->out_channels * am->in_channels * \
sizeof(*am->matrix_## type[0])); \
if (!am->matrix_## type[0]) \
return AVERROR(ENOMEM); \
for (o = 0; o < am->out_channels; o++) { \
if (o > 0) \
am->matrix_## type[o] = am->matrix_## type[o - 1] + \
am->in_channels; \
for (i = 0; i < am->in_channels; i++) { \
double v = matrix[o * stride + i]; \
am->matrix_## type[o][i] = expr; \
} \
} \
am->matrix = (void **)am->matrix_## type;
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, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
/* TODO: detect situations where we can just swap around pointers
instead of doing matrix multiplications with 0.0 and 1.0 */
return 0;
}
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