int CFfAudioResample::Resample( sqbind::CSqBinary *pSrc, sqbind::CSqBinary *pDst )
{_STT();
// Validate resampler
if ( !m_ctx || 0 >= m_nInRate || 0 >= m_nOutRate )
return 0;
// Validate params
if ( !pDst || !pSrc )
return 0;
// Do we have data?
if ( 0 >= pSrc->getUsed() )
return 0;
// Calculate required output buffer size
oex::oexINT64 nOutSize = pSrc->getUsed() * m_nOutRate / m_nInRate + FF_INPUT_BUFFER_PADDING_SIZE + 1024;
// Allocate space for output data if needed
if ( ( pDst->Size() - pDst->getUsed() ) < nOutSize )
if ( !pDst->Resize( pDst->getUsed() + nOutSize ) )
return 0;
// Build resampling context
int consumed = 0;
int out = av_resample( m_ctx, (short*)pDst->Ptr( pDst->getUsed() ), (short*)pSrc->Ptr(),
&consumed, pSrc->getUsed() / 2, pDst->Size() / 2, 0 );
if ( 0 > out )
{ pDst->setUsed( 0 );
return 0;
} // end if
// Number of bytes used
pDst->setUsed( out );
return out;
}
void AudioProcessor::Resample()
{
if (!m_resample_ctx) {
m_consumer->Consume(m_buffer.data(), m_buffer_offset);
m_buffer_offset = 0;
return;
}
int consumed = 0;
int length = av_resample(m_resample_ctx, m_resample_buffer.data(), m_buffer.data(), &consumed, m_buffer_offset, kMaxBufferSize, 1);
if (length > kMaxBufferSize) {
DEBUG("chromaprint::AudioProcessor::Resample() -- Resampling overwrote output buffer.");
length = kMaxBufferSize;
}
m_consumer->Consume(m_resample_buffer.data(), length);
int remaining = m_buffer_offset - consumed;
if (remaining > 0) {
std::copy(m_buffer.begin() + consumed, m_buffer.begin() + m_buffer_offset, m_buffer.begin());
}
else if (remaining < 0) {
DEBUG("chromaprint::AudioProcessor::Resample() -- Resampling overread input buffer.");
remaining = 0;
}
m_buffer_offset = remaining;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
AVFilterLink * const outlink = inlink->dst->outputs[0];
int i,
in_nb_samples = insamplesref->audio->nb_samples,
cached_nb_samples = in_nb_samples + aresample->unconsumed_nb_samples,
requested_out_nb_samples = aresample->ratio * cached_nb_samples,
nb_channels =
av_get_channel_layout_nb_channels(inlink->channel_layout);
if (cached_nb_samples > aresample->max_cached_nb_samples) {
for (i = 0; i < nb_channels; i++) {
aresample->cached_data[i] =
av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
aresample->resampled_data[i] =
av_realloc(aresample->resampled_data[i],
FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
return;
}
aresample->max_cached_nb_samples = cached_nb_samples;
if (aresample->outsamplesref)
avfilter_unref_buffer(aresample->outsamplesref);
aresample->outsamplesref =
avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, requested_out_nb_samples);
outlink->out_buf = aresample->outsamplesref;
}
avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
aresample->outsamplesref->pts =
av_rescale(outlink->sample_rate, insamplesref->pts, inlink->sample_rate);
/* av_resample() works with planar audio buffers */
if (!inlink->planar && nb_channels > 1) {
int16_t *out[8];
for (i = 0; i < nb_channels; i++)
out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
deinterleave(out, (int16_t *)insamplesref->data[0],
nb_channels, in_nb_samples);
} else {
for (i = 0; i < nb_channels; i++)
memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
insamplesref->data[i],
in_nb_samples * sizeof(int16_t));
}
for (i = 0; i < nb_channels; i++) {
int consumed_nb_samples;
const int is_last = i+1 == nb_channels;
aresample->outsamplesref->audio->nb_samples =
av_resample(aresample->resample,
aresample->resampled_data[i], aresample->cached_data[i],
&consumed_nb_samples,
cached_nb_samples,
requested_out_nb_samples, is_last);
/* move unconsumed data back to the beginning of the cache */
aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
memmove(aresample->cached_data[i],
aresample->cached_data[i] + consumed_nb_samples,
aresample->unconsumed_nb_samples * sizeof(int16_t));
}
/* copy resampled data to the output samplesref */
if (!inlink->planar && nb_channels > 1) {
interleave((int16_t *)aresample->outsamplesref->data[0],
aresample->resampled_data,
nb_channels, aresample->outsamplesref->audio->nb_samples);
} else {
for (i = 0; i < nb_channels; i++)
memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
}
avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
avfilter_unref_buffer(insamplesref);
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
af_resample_t *s = af->setup;
int i, j, consumed, ret;
int16_t *in = (int16_t*)data->audio;
int16_t *out;
int chans = data->nch;
int in_len = data->len/(2*chans);
int out_len = in_len * af->mul + 10;
int16_t tmp[AF_NCH][out_len];
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
return NULL;
out= (int16_t*)af->data->audio;
out_len= min(out_len, af->data->len/(2*chans));
if(s->in_alloc < in_len + s->index){
s->in_alloc= in_len + s->index;
for(i=0; i<chans; i++){
s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t));
}
}
if(chans==1){
memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
}else if(chans==2){
for(j=0; j<in_len; j++){
s->in[0][j + s->index]= *(in++);
s->in[1][j + s->index]= *(in++);
}
}else{
for(j=0; j<in_len; j++){
for(i=0; i<chans; i++){
s->in[i][j + s->index]= *(in++);
}
}
}
in_len += s->index;
for(i=0; i<chans; i++){
ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans);
}
out_len= ret;
s->index= in_len - consumed;
for(i=0; i<chans; i++){
memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
}
if(chans==1){
memcpy(out, tmp[0], out_len*sizeof(int16_t));
}else if(chans==2){
for(j=0; j<out_len; j++){
*(out++)= tmp[0][j];
*(out++)= tmp[1][j];
}
}else{
for(j=0; j<out_len; j++){
for(i=0; i<chans; i++){
*(out++)= tmp[i][j];
}
}
}
data->audio = af->data->audio;
data->len = out_len*chans*2;
data->rate = af->data->rate;
return data;
}
