74 if (ptr_align == 1 && samples_align == 1) {
82 snprintf(chan_str,
sizeof(chan_str),
"[%d to %d] ",
83 in_channels, out_channels);
85 snprintf(chan_str,
sizeof(chan_str),
"[%d to any] ",
87 }
else if (out_channels) {
88 snprintf(chan_str,
sizeof(chan_str),
"[any to %d] ",
91 snprintf(chan_str,
sizeof(chan_str),
"[any to any] ");
99 #define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c 101 #define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \ 102 static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \ 103 int len, int out_ch, int in_ch) \ 106 stype temp[AVRESAMPLE_MAX_CHANNELS]; \ 107 for (i = 0; i < len; i++) { \ 108 for (out = 0; out < out_ch; out++) { \ 110 for (in = 0; in < in_ch; in++) \ 111 sum += samples[in][i] * matrix[out][in]; \ 114 for (out = 0; out < out_ch; out++) \ 115 samples[out][i] = temp[out]; \ 122 MIX_FUNC_GENERIC(S16P, Q8, int16_t, int16_t, int32_t, av_clip_int16(sum >> 8))
126 static
void mix_2_to_1_fltp_flt_c(
float **samples,
float **
matrix,
int len,
127 int out_ch,
int in_ch)
129 float *
src0 = samples[0];
130 float *
src1 = samples[1];
132 float m0 = matrix[0][0];
133 float m1 = matrix[0][1];
136 *dst++ = *src0++ * m0 + *src1++ * m1;
137 *dst++ = *src0++ * m0 + *src1++ * m1;
138 *dst++ = *src0++ * m0 + *src1++ * m1;
139 *dst++ = *src0++ * m0 + *src1++ * m1;
143 *dst++ = *src0++ * m0 + *src1++ * m1;
149 int out_ch,
int in_ch)
151 int16_t *
src0 = samples[0];
152 int16_t *
src1 = samples[1];
154 float m0 = matrix[0][0];
155 float m1 = matrix[0][1];
158 *dst++ = av_clip_int16(
lrintf(*src0++ * m0 + *src1++ * m1));
159 *dst++ = av_clip_int16(
lrintf(*src0++ * m0 + *src1++ * m1));
160 *dst++ = av_clip_int16(
lrintf(*src0++ * m0 + *src1++ * m1));
161 *dst++ = av_clip_int16(
lrintf(*src0++ * m0 + *src1++ * m1));
165 *dst++ = av_clip_int16(
lrintf(*src0++ * m0 + *src1++ * m1));
171 int out_ch,
int in_ch)
173 int16_t *
src0 = samples[0];
174 int16_t *
src1 = samples[1];
176 int16_t m0 = matrix[0][0];
177 int16_t m1 = matrix[0][1];
180 *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
181 *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
182 *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
183 *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
187 *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
193 int out_ch,
int in_ch)
196 float *dst0 = samples[0];
197 float *dst1 = samples[1];
199 float m0 = matrix[0][0];
200 float m1 = matrix[1][0];
226 int out_ch,
int in_ch)
229 float *
src0 = samples[0];
230 float *
src1 = samples[1];
231 float *src2 = samples[2];
232 float *src3 = samples[3];
233 float *src4 = samples[4];
234 float *src5 = samples[5];
237 float *m0 = matrix[0];
238 float *m1 = matrix[1];
243 *dst0++ = v0 * m0[0] +
249 *dst1++ = v0 * m1[0] +
260 int out_ch,
int in_ch)
263 float *dst0 = samples[0];
264 float *dst1 = samples[1];
265 float *dst2 = samples[2];
266 float *dst3 = samples[3];
267 float *dst4 = samples[4];
268 float *dst5 = samples[5];
275 *dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1];
276 *dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1];
277 *dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1];
278 *dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1];
279 *dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1];
280 *dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1];
311 2, 1, 1, 1,
"C", mix_2_to_1_fltp_flt_c);
374 sizeof(*matrix_dbl));
457 data0[j++] = src->
data[i];
493 #define GET_MATRIX_CONVERT(suffix, scale) \ 494 if (!am->matrix_ ## suffix[0]) { \ 495 av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \ 496 return AVERROR(EINVAL); \ 498 for (o = 0, o0 = 0; o < am->out_channels; o++) { \ 499 for (i = 0, i0 = 0; i < am->in_channels; i++) { \ 500 if (am->input_skip[i] || am->output_zero[o]) \ 501 matrix[o * stride + i] = 0.0; \ 503 matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \ 505 if (!am->input_skip[i]) \ 508 if (!am->output_zero[o]) \ 544 if (matrix[o * stride + i] != 0.0) {
553 if (matrix[i * stride + o] != 0.0) {
562 if (o < am->in_channels)
579 if ((o != i && matrix[o * stride + i] != 0.0) ||
580 (o == i && matrix[o * stride + i] != 1.0)) {
588 if (i0 != i && matrix[o * stride + i0] != 0.0) {
605 if (matrix[o * stride + i] != 0.0) {
627 if ((o != i && matrix[o * stride + i] != 0.0) ||
628 (o == i && matrix[o * stride + i] != 1.0)) {
637 if (o0 != i && matrix[o0 * stride + i] != 0.0) {
655 int i, o, i0, o0, ret;
656 char in_layout_name[128];
657 char out_layout_name[128];
675 #define CONVERT_MATRIX(type, expr) \ 676 am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \ 677 am->in_matrix_channels * \ 678 sizeof(*am->matrix_## type[0])); \ 679 if (!am->matrix_## type[0]) \ 680 return AVERROR(ENOMEM); \ 681 for (o = 0, o0 = 0; o < am->out_channels; o++) { \ 682 if (am->output_zero[o] || am->output_skip[o]) \ 685 am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \ 686 am->in_matrix_channels; \ 687 for (i = 0, i0 = 0; i < am->in_channels; i++) { \ 689 if (am->input_skip[i] || am->output_zero[i]) \ 691 v = matrix[o * stride + i]; \ 692 am->matrix_## type[o0][i0] = expr; \ 697 am->matrix = (void **)am->matrix_## type; 725 in_layout_name, out_layout_name);
#define MIX_FUNC_NAME(fmt, cfmt)
int in_channels
number of input channels
int output_skip[AVRESAMPLE_MAX_CHANNELS]
Audio buffer used for intermediate storage between conversion phases.
double * mix_matrix
mix matrix only used if avresample_set_matrix() is called before avresample_open() ...
uint64_t out_channel_layout
output channel layout
static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len, int out_ch, int in_ch)
enum attribute_deprecated AVMixCoeffType
int16_t * matrix_q8[AVRESAMPLE_MAX_CHANNELS]
int ff_audio_mix(AudioMix *am, AudioData *src)
Apply channel mixing to audio data using the current mixing matrix.
int nb_samples
current number of samples
double surround_mix_level
surround mix level
enum AVMixCoeffType mix_coeff_type
mixing coefficient type
int out_channels
number of output channels
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
int normalize_mix_level
enable mix level normalization
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
Set the current mixing matrix.
static void mix_2_to_1_s16p_flt_c(int16_t **samples, float **matrix, int len, int out_ch, int in_ch)
int output_zero[AVRESAMPLE_MAX_CHANNELS]
int ff_audio_data_set_channels(AudioData *a, int channels)
enum AVMixCoeffType coeff_type
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
double center_mix_level
center mix level
static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len, int out_ch, int in_ch)
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
static const char *const coeff_type_names[]
uint64_t in_channel_layout
input channel layout
static void error(const char *err)
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
AVSampleFormat
Audio sample formats.
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout, double center_mix_level, double surround_mix_level, double lfe_mix_level, int normalize, double *matrix_out, int stride, enum AVMatrixEncoding matrix_encoding)
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
void ff_audio_mix_init_x86(AudioMix *am)
#define AVRESAMPLE_MAX_CHANNELS
enum AVSampleFormat internal_sample_fmt
internal sample format
static void reduce_matrix(AudioMix *am, const double *matrix, int stride)
void() mix_func(uint8_t **src, void **matrix, int len, int out_ch, int in_ch)
void ff_audio_mix_free(AudioMix **am_p)
Free an AudioMix context.
Replacements for frequently missing libm functions.
int32_t * matrix_q15[AVRESAMPLE_MAX_CHANNELS]
const char * func_descr_generic
float * matrix_flt[AVRESAMPLE_MAX_CHANNELS]
int input_skip[AVRESAMPLE_MAX_CHANNELS]
static av_cold int mix_function_init(AudioMix *am)
int samples_align
allocated samples alignment
#define GET_MATRIX_CONVERT(suffix, scale)
common internal and external API header
enum AVMatrixEncoding matrix_encoding
matrixed stereo encoding
AudioMix * ff_audio_mix_alloc(AVAudioResampleContext *avr)
Allocate and initialize an AudioMix context.
#define CONVERT_MATRIX(type, expr)
static void mix_2_to_1_s16p_q8_c(int16_t **samples, int16_t **matrix, int len, int out_ch, int in_ch)
static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len, int out_ch, int in_ch)
AV_MIX_COEFF_TYPE_FLT
32-bit 17.15 fixed-point
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)
Set mixing function if the parameters match.
int ptr_align
minimum data pointer alignment
AV_MIX_COEFF_TYPE_Q15
16-bit 8.8 fixed-point
double lfe_mix_level
lfe mix level
int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
Get the current mixing matrix.
#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr)
AVAudioResampleContext * avr