int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len)
{
int ch;
//FIXME optimize common cases
for(ch=0; ch<ctx->out_channels; ch++){
const int is= in_stride[ch];
const int os= out_stride[ch];
const uint8_t *pi= in[ch];
uint8_t *po= out[ch];
uint8_t *end= po + os*len;
if(!out[ch])
continue;
#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\
do{\
*(otype*)po = expr; pi += is; po += os;\
}while(po < end);\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding ?
CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else return -1;
}
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->avr, 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->avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
return mix_function_init(am);
}
static void set(uint8_t *a[], int ch, int index, int ch_count, enum AVSampleFormat f, double v){
uint8_t *p;
if(av_sample_fmt_is_planar(f)){
f= av_get_alt_sample_fmt(f, 0);
p= a[ch];
}else{
p= a[0];
index= ch + index*ch_count;
}
switch(f){
case AV_SAMPLE_FMT_U8 : ((uint8_t*)p)[index]= av_clip_uint8 (lrint((v+1.0)*127)); break;
case AV_SAMPLE_FMT_S16: ((int16_t*)p)[index]= av_clip_int16 (lrint(v*32767)); break;
case AV_SAMPLE_FMT_S32: ((int32_t*)p)[index]= av_clipl_int32(llrint(v*2147483647)); break;
case AV_SAMPLE_FMT_FLT: ((float *)p)[index]= v; break;
case AV_SAMPLE_FMT_DBL: ((double *)p)[index]= v; break;
default: av_assert2(0);
}
}
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<24)
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0f/ (1<<7)))
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0f/ (1<<15)))
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0f/ (1U<<31)))
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
#define FMT_PAIR_FUNC(out, in) [(out) + AV_SAMPLE_FMT_NB*(in)] = CONV_FUNC_NAME(out, in)
static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = {
FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8 ),
FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8 ),
/**
* builds a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param filter_type filter type
* @param kaiser_beta kaiser window beta
* @return 0 on success, negative on error
*/
static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
int filter_type, double kaiser_beta){
int ph, i;
int ph_nb = phase_count % 2 ? phase_count : phase_count / 2 + 1;
double x, y, w, t, s;
double *tab = av_malloc_array(tap_count+1, sizeof(*tab));
double *sin_lut = av_malloc_array(ph_nb, sizeof(*sin_lut));
const int center= (tap_count-1)/2;
int ret = AVERROR(ENOMEM);
if (!tab || !sin_lut)
goto fail;
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
if (factor == 1.0) {
for (ph = 0; ph < ph_nb; ph++)
sin_lut[ph] = sin(M_PI * ph / phase_count);
}
for(ph = 0; ph < ph_nb; ph++) {
double norm = 0;
s = sin_lut[ph];
for(i=0;i<=tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else if (factor == 1.0)
y = s / x;
else
y = sin(x) / x;
switch(filter_type){
case SWR_FILTER_TYPE_CUBIC:{
const float d= -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
break;}
case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
w = 2.0*x / (factor*tap_count);
t = -cos(w);
y *= 0.3635819 - 0.4891775 * t + 0.1365995 * (2*t*t-1) - 0.0106411 * (4*t*t*t - 3*t);
break;
case SWR_FILTER_TYPE_KAISER:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
break;
default:
av_assert0(0);
}
tab[i] = y;
s = -s;
if (i < tap_count)
norm += y;
}
/* normalize so that an uniform color remains the same */
switch(c->format){
case AV_SAMPLE_FMT_S16P:
for(i=0;i<tap_count;i++)
((int16_t*)filter)[ph * alloc + i] = av_clip_int16(lrintf(tab[i] * scale / norm));
if (phase_count % 2) break;
if (tap_count % 2 == 0 || tap_count == 1) {
for (i = 0; i < tap_count; i++)
((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i];
}
else {
for (i = 1; i <= tap_count; i++)
((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-i] =
av_clip_int16(lrintf(tab[i] * scale / (norm - tab[0] + tab[tap_count])));
}
break;
case AV_SAMPLE_FMT_S32P:
for(i=0;i<tap_count;i++)
((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));
if (phase_count % 2) break;
if (tap_count % 2 == 0 || tap_count == 1) {
for (i = 0; i < tap_count; i++)
((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i];
}
else {
for (i = 1; i <= tap_count; i++)
((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-i] =
av_clipl_int32(llrint(tab[i] * scale / (norm - tab[0] + tab[tap_count])));
}
break;
case AV_SAMPLE_FMT_FLTP:
for(i=0;i<tap_count;i++)
((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
if (phase_count % 2) break;
if (tap_count % 2 == 0 || tap_count == 1) {
for (i = 0; i < tap_count; i++)
((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i];
}
else {
for (i = 1; i <= tap_count; i++)
((float*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]);
}
break;
case AV_SAMPLE_FMT_DBLP:
for(i=0;i<tap_count;i++)
((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
if (phase_count % 2) break;
if (tap_count % 2 == 0 || tap_count == 1) {
for (i = 0; i < tap_count; i++)
((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i];
}
else {
for (i = 1; i <= tap_count; i++)
((double*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]);
}
break;
}
}
#if 0
{
#define LEN 1024
int j,k;
double sine[LEN + tap_count];
double filtered[LEN];
double maxff=-2, minff=2, maxsf=-2, minsf=2;
for(i=0; i<LEN; i++){
double ss=0, sf=0, ff=0;
for(j=0; j<LEN+tap_count; j++)
sine[j]= cos(i*j*M_PI/LEN);
for(j=0; j<LEN; j++){
double sum=0;
ph=0;
for(k=0; k<tap_count; k++)
sum += filter[ph * tap_count + k] * sine[k+j];
filtered[j]= sum / (1<<FILTER_SHIFT);
ss+= sine[j + center] * sine[j + center];
ff+= filtered[j] * filtered[j];
sf+= sine[j + center] * filtered[j];
}
ss= sqrt(2*ss/LEN);
ff= sqrt(2*ff/LEN);
sf= 2*sf/LEN;
maxff= FFMAX(maxff, ff);
minff= FFMIN(minff, ff);
maxsf= FFMAX(maxsf, sf);
minsf= FFMIN(minsf, sf);
if(i%11==0){
av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
minff=minsf= 2;
maxff=maxsf= -2;
}
}
}
#endif
ret = 0;
fail:
av_free(tab);
av_free(sin_lut);
return ret;
}
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;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
ASyncContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout);
int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
int out_size, ret;
int64_t delta;
int64_t new_pts;
/* buffer data until we get the next timestamp */
if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) {
if (pts != AV_NOPTS_VALUE) {
s->pts = pts - get_delay(s);
}
return write_to_fifo(s, buf);
}
if (s->first_pts != AV_NOPTS_VALUE) {
handle_trimming(ctx);
if (!avresample_available(s->avr))
return write_to_fifo(s, buf);
}
/* when we have two timestamps, compute how many samples would we have
* to add/remove to get proper sync between data and timestamps */
delta = pts - s->pts - get_delay(s);
out_size = avresample_available(s->avr);
if (llabs(delta) > s->min_delta ||
(s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) {
av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
out_size = av_clipl_int32((int64_t)out_size + delta);
} else {
if (s->resample) {
// adjust the compensation if delta is non-zero
int delay = get_delay(s);
int comp = s->comp + av_clip(delta * inlink->sample_rate / delay,
-s->max_comp, s->max_comp);
if (comp != s->comp) {
av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) {
s->comp = comp;
}
}
}
// adjust PTS to avoid monotonicity errors with input PTS jitter
pts -= delta;
delta = 0;
}
if (out_size > 0) {
AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size);
if (!buf_out) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (s->first_frame && delta > 0) {
int planar = av_sample_fmt_is_planar(buf_out->format);
int planes = planar ? nb_channels : 1;
int block_size = av_get_bytes_per_sample(buf_out->format) *
(planar ? 1 : nb_channels);
int ch;
av_samples_set_silence(buf_out->extended_data, 0, delta,
nb_channels, buf->format);
for (ch = 0; ch < planes; ch++)
buf_out->extended_data[ch] += delta * block_size;
avresample_read(s->avr, buf_out->extended_data, out_size);
for (ch = 0; ch < planes; ch++)
buf_out->extended_data[ch] -= delta * block_size;
} else {
avresample_read(s->avr, buf_out->extended_data, out_size);
if (delta > 0) {
av_samples_set_silence(buf_out->extended_data, out_size - delta,
delta, nb_channels, buf->format);
}
}
buf_out->pts = s->pts;
ret = ff_filter_frame(outlink, buf_out);
if (ret < 0)
goto fail;
s->got_output = 1;
} else if (avresample_available(s->avr)) {
av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
"whole buffer.\n");
}
/* drain any remaining buffered data */
avresample_read(s->avr, NULL, avresample_available(s->avr));
new_pts = pts - avresample_get_delay(s->avr);
/* check for s->pts monotonicity */
if (new_pts > s->pts) {
s->pts = new_pts;
ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
buf->linesize[0], buf->nb_samples);
} else {
av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
"whole buffer.\n");
ret = 0;
}
s->first_frame = 0;
fail:
av_frame_free(&buf);
return ret;
}
int ff_audio_mix_set_matrix(AudioMix *am, const 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);
}
if (am->matrix) {
av_free(am->matrix[0]);
am->matrix = NULL;
}
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_zero));
am->in_matrix_channels = am->in_channels;
am->out_matrix_channels = am->out_channels;
/* 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 0;
}
/* 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++) {
if ((o != i && matrix[o * stride + i] != 0.0) ||
(o == i && matrix[o * stride + i] != 1.0)) {
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 0;
}
/* 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;
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;
}
}
if (skip) {
am->output_skip[o] = 1;
am->out_matrix_channels--;
}
}
if (am->out_matrix_channels == 0) {
am->in_matrix_channels = 0;
return 0;
}
#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;
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);
}
return mix_function_init(am);
}
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
int stride)
{
int in_channels, out_channels, i, o;
in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
return AVERROR(EINVAL);
}
if (avr->am->matrix) {
av_free(avr->am->matrix[0]);
avr->am->matrix = NULL;
}
#define CONVERT_MATRIX(type, expr) \
avr->am->matrix_## type[0] = av_mallocz(out_channels * in_channels * \
sizeof(*avr->am->matrix_## type[0])); \
if (!avr->am->matrix_## type[0]) \
return AVERROR(ENOMEM); \
for (o = 0; o < out_channels; o++) { \
if (o > 0) \
avr->am->matrix_## type[o] = avr->am->matrix_## type[o - 1] + \
in_channels; \
for (i = 0; i < in_channels; i++) { \
double v = matrix[o * stride + i]; \
avr->am->matrix_## type[o][i] = expr; \
} \
} \
avr->am->matrix = (void **)avr->am->matrix_## type;
switch (avr->mix_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(avr, 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 */
/* set AudioMix params */
avr->am->in_layout = avr->in_channel_layout;
avr->am->out_layout = avr->out_channel_layout;
avr->am->in_channels = in_channels;
avr->am->out_channels = out_channels;
return 0;
}
