mirror of https://git.ffmpeg.org/ffmpeg.git
743 lines
26 KiB
C
743 lines
26 KiB
C
/*
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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#include <stdint.h>
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#include "libavutil/common.h"
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#include "libavutil/libm.h"
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#include "libavutil/samplefmt.h"
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#include "avresample.h"
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#include "internal.h"
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#include "audio_data.h"
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#include "audio_mix.h"
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static const char * const coeff_type_names[] = { "q8", "q15", "flt" };
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struct AudioMix {
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AVAudioResampleContext *avr;
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enum AVSampleFormat fmt;
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enum AVMixCoeffType coeff_type;
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uint64_t in_layout;
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uint64_t out_layout;
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int in_channels;
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int out_channels;
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int ptr_align;
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int samples_align;
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int has_optimized_func;
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const char *func_descr;
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const char *func_descr_generic;
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mix_func *mix;
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mix_func *mix_generic;
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int in_matrix_channels;
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int out_matrix_channels;
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int output_zero[AVRESAMPLE_MAX_CHANNELS];
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int input_skip[AVRESAMPLE_MAX_CHANNELS];
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int output_skip[AVRESAMPLE_MAX_CHANNELS];
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int16_t *matrix_q8[AVRESAMPLE_MAX_CHANNELS];
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int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS];
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float *matrix_flt[AVRESAMPLE_MAX_CHANNELS];
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void **matrix;
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};
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void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
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enum AVMixCoeffType coeff_type, int in_channels,
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int out_channels, int ptr_align, int samples_align,
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const char *descr, void *mix_func)
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{
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if (fmt == am->fmt && coeff_type == am->coeff_type &&
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( in_channels == am->in_matrix_channels || in_channels == 0) &&
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(out_channels == am->out_matrix_channels || out_channels == 0)) {
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char chan_str[16];
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am->mix = mix_func;
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am->func_descr = descr;
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am->ptr_align = ptr_align;
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am->samples_align = samples_align;
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if (ptr_align == 1 && samples_align == 1) {
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am->mix_generic = mix_func;
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am->func_descr_generic = descr;
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} else {
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am->has_optimized_func = 1;
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}
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if (in_channels) {
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if (out_channels)
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snprintf(chan_str, sizeof(chan_str), "[%d to %d] ",
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in_channels, out_channels);
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else
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snprintf(chan_str, sizeof(chan_str), "[%d to any] ",
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in_channels);
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} else if (out_channels) {
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snprintf(chan_str, sizeof(chan_str), "[any to %d] ",
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out_channels);
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} else {
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snprintf(chan_str, sizeof(chan_str), "[any to any] ");
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}
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av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] "
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"[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt),
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coeff_type_names[coeff_type], chan_str, descr);
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}
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}
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#define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c
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#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \
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static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \
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int len, int out_ch, int in_ch) \
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{ \
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int i, in, out; \
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stype temp[AVRESAMPLE_MAX_CHANNELS]; \
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for (i = 0; i < len; i++) { \
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for (out = 0; out < out_ch; out++) { \
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sumtype sum = 0; \
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for (in = 0; in < in_ch; in++) \
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sum += samples[in][i] * matrix[out][in]; \
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temp[out] = expr; \
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} \
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for (out = 0; out < out_ch; out++) \
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samples[out][i] = temp[out]; \
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} \
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}
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MIX_FUNC_GENERIC(FLTP, FLT, float, float, float, sum)
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MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum)))
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MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15))
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MIX_FUNC_GENERIC(S16P, Q8, int16_t, int16_t, int32_t, av_clip_int16(sum >> 8))
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/* TODO: templatize the channel-specific C functions */
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static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len,
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int out_ch, int in_ch)
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{
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float *src0 = samples[0];
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float *src1 = samples[1];
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float *dst = src0;
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float m0 = matrix[0][0];
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float m1 = matrix[0][1];
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while (len > 4) {
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*dst++ = *src0++ * m0 + *src1++ * m1;
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*dst++ = *src0++ * m0 + *src1++ * m1;
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*dst++ = *src0++ * m0 + *src1++ * m1;
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*dst++ = *src0++ * m0 + *src1++ * m1;
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len -= 4;
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}
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while (len > 0) {
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*dst++ = *src0++ * m0 + *src1++ * m1;
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len--;
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}
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}
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static void mix_2_to_1_s16p_flt_c(int16_t **samples, float **matrix, int len,
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int out_ch, int in_ch)
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{
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int16_t *src0 = samples[0];
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int16_t *src1 = samples[1];
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int16_t *dst = src0;
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float m0 = matrix[0][0];
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float m1 = matrix[0][1];
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while (len > 4) {
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*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
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*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
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*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
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*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
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len -= 4;
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}
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while (len > 0) {
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*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
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len--;
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}
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}
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static void mix_2_to_1_s16p_q8_c(int16_t **samples, int16_t **matrix, int len,
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int out_ch, int in_ch)
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{
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int16_t *src0 = samples[0];
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int16_t *src1 = samples[1];
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int16_t *dst = src0;
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int16_t m0 = matrix[0][0];
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int16_t m1 = matrix[0][1];
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while (len > 4) {
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*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
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*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
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*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
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*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
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len -= 4;
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}
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while (len > 0) {
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*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
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len--;
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}
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}
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static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len,
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int out_ch, int in_ch)
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{
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float v;
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float *dst0 = samples[0];
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float *dst1 = samples[1];
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float *src = dst0;
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float m0 = matrix[0][0];
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float m1 = matrix[1][0];
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while (len > 4) {
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v = *src++;
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*dst0++ = v * m0;
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*dst1++ = v * m1;
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v = *src++;
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*dst0++ = v * m0;
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*dst1++ = v * m1;
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v = *src++;
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*dst0++ = v * m0;
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*dst1++ = v * m1;
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v = *src++;
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*dst0++ = v * m0;
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*dst1++ = v * m1;
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len -= 4;
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}
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while (len > 0) {
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v = *src++;
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*dst0++ = v * m0;
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*dst1++ = v * m1;
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len--;
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}
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}
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static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len,
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int out_ch, int in_ch)
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{
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float v0, v1;
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float *src0 = samples[0];
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float *src1 = samples[1];
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float *src2 = samples[2];
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float *src3 = samples[3];
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float *src4 = samples[4];
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float *src5 = samples[5];
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float *dst0 = src0;
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float *dst1 = src1;
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float *m0 = matrix[0];
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float *m1 = matrix[1];
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while (len > 0) {
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v0 = *src0++;
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v1 = *src1++;
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*dst0++ = v0 * m0[0] +
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v1 * m0[1] +
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*src2 * m0[2] +
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*src3 * m0[3] +
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*src4 * m0[4] +
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*src5 * m0[5];
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*dst1++ = v0 * m1[0] +
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v1 * m1[1] +
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*src2++ * m1[2] +
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*src3++ * m1[3] +
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*src4++ * m1[4] +
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*src5++ * m1[5];
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len--;
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}
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}
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static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len,
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int out_ch, int in_ch)
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{
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float v0, v1;
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float *dst0 = samples[0];
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float *dst1 = samples[1];
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float *dst2 = samples[2];
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float *dst3 = samples[3];
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float *dst4 = samples[4];
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float *dst5 = samples[5];
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float *src0 = dst0;
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float *src1 = dst1;
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while (len > 0) {
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v0 = *src0++;
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v1 = *src1++;
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*dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1];
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*dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1];
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*dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1];
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*dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1];
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*dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1];
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*dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1];
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len--;
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}
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}
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static av_cold int mix_function_init(AudioMix *am)
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{
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am->func_descr = am->func_descr_generic = "n/a";
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am->mix = am->mix_generic = NULL;
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/* no need to set a mix function when we're skipping mixing */
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if (!am->in_matrix_channels || !am->out_matrix_channels)
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return 0;
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/* any-to-any C versions */
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
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0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP, FLT));
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
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0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, FLT));
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q15,
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0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q15));
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
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0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q8));
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/* channel-specific C versions */
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
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2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c);
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
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2, 1, 1, 1, "C", mix_2_to_1_s16p_flt_c);
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
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2, 1, 1, 1, "C", mix_2_to_1_s16p_q8_c);
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
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1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c);
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
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6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c);
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ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
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2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c);
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if (ARCH_X86)
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ff_audio_mix_init_x86(am);
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if (!am->mix) {
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av_log(am->avr, AV_LOG_ERROR, "audio_mix: NO FUNCTION FOUND: [fmt=%s] "
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"[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am->fmt),
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coeff_type_names[am->coeff_type], am->in_channels,
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am->out_channels);
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return AVERROR_PATCHWELCOME;
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}
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return 0;
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}
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AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr)
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{
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AudioMix *am;
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int ret;
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am = av_mallocz(sizeof(*am));
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if (!am)
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return NULL;
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am->avr = avr;
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if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
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av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
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"mixing: %s\n",
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av_get_sample_fmt_name(avr->internal_sample_fmt));
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goto error;
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}
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am->fmt = avr->internal_sample_fmt;
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am->coeff_type = avr->mix_coeff_type;
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am->in_layout = avr->in_channel_layout;
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am->out_layout = avr->out_channel_layout;
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am->in_channels = avr->in_channels;
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am->out_channels = avr->out_channels;
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/* build matrix if the user did not already set one */
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if (avr->mix_matrix) {
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ret = ff_audio_mix_set_matrix(am, avr->mix_matrix, avr->in_channels);
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if (ret < 0)
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goto error;
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av_freep(&avr->mix_matrix);
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} else {
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double *matrix_dbl = av_mallocz(avr->out_channels * avr->in_channels *
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sizeof(*matrix_dbl));
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if (!matrix_dbl)
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goto error;
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|
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ret = avresample_build_matrix(avr->in_channel_layout,
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avr->out_channel_layout,
|
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avr->center_mix_level,
|
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avr->surround_mix_level,
|
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avr->lfe_mix_level,
|
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avr->normalize_mix_level,
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matrix_dbl,
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avr->in_channels,
|
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avr->matrix_encoding);
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if (ret < 0) {
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av_free(matrix_dbl);
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goto error;
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}
|
|
|
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ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels);
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if (ret < 0) {
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av_log(avr, AV_LOG_ERROR, "error setting mix matrix\n");
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av_free(matrix_dbl);
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goto error;
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}
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|
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av_free(matrix_dbl);
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}
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|
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return am;
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|
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error:
|
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av_free(am);
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return NULL;
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}
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|
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void ff_audio_mix_free(AudioMix **am_p)
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{
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AudioMix *am;
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if (!*am_p)
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return;
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am = *am_p;
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|
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if (am->matrix) {
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av_free(am->matrix[0]);
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am->matrix = NULL;
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}
|
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memset(am->matrix_q8, 0, sizeof(am->matrix_q8 ));
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memset(am->matrix_q15, 0, sizeof(am->matrix_q15));
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memset(am->matrix_flt, 0, sizeof(am->matrix_flt));
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|
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av_freep(am_p);
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}
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|
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int ff_audio_mix(AudioMix *am, AudioData *src)
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{
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int use_generic = 1;
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int len = src->nb_samples;
|
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int i, j;
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|
|
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/* determine whether to use the optimized function based on pointer and
|
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samples alignment in both the input and output */
|
|
if (am->has_optimized_func) {
|
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int aligned_len = FFALIGN(len, am->samples_align);
|
|
if (!(src->ptr_align % am->ptr_align) &&
|
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src->samples_align >= aligned_len) {
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len = aligned_len;
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use_generic = 0;
|
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}
|
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}
|
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av_log(am->avr, AV_LOG_TRACE, "audio_mix: %d samples - %d to %d channels (%s)\n",
|
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src->nb_samples, am->in_channels, am->out_channels,
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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;
|
|
}
|