apm->voice_detection()->Enable(true);
ec->params.webrtc.apm = apm;
- ec->params.webrtc.sample_spec = *out_ss;
- ec->params.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
- *nframes = ec->params.webrtc.blocksize / pa_frame_size(out_ss);
+ ec->params.webrtc.rec_ss = *rec_ss;
+ ec->params.webrtc.play_ss = *play_ss;
+ ec->params.webrtc.blocksize = (uint64_t) out_ss->rate * BLOCK_SIZE_US / PA_USEC_PER_SEC;
+ *nframes = ec->params.webrtc.blocksize;
ec->params.webrtc.first = true;
pa_modargs_free(ma);
void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
webrtc::AudioFrame play_frame;
- const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
+ const pa_sample_spec *ss = &ec->params.webrtc.play_ss;
play_frame.num_channels_ = ss->channels;
play_frame.sample_rate_hz_ = ss->rate;
play_frame.interleaved_ = true;
- play_frame.samples_per_channel_ = ec->params.webrtc.blocksize / pa_frame_size(ss);
+ play_frame.samples_per_channel_ = ec->params.webrtc.blocksize;
pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
- memcpy(play_frame.data_, play, ec->params.webrtc.blocksize);
+ memcpy(play_frame.data_, play, ec->params.webrtc.blocksize * pa_frame_size(ss));
apm->ProcessReverseStream(&play_frame);
void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
webrtc::AudioFrame out_frame;
- const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
+ const pa_sample_spec *ss = &ec->params.webrtc.rec_ss;
pa_cvolume v;
int old_volume, new_volume;
out_frame.num_channels_ = ss->channels;
out_frame.sample_rate_hz_ = ss->rate;
out_frame.interleaved_ = true;
- out_frame.samples_per_channel_ = ec->params.webrtc.blocksize / pa_frame_size(ss);
+ out_frame.samples_per_channel_ = ec->params.webrtc.blocksize;
pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
- memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize);
+ memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize * pa_frame_size(ss));
if (ec->params.webrtc.agc) {
pa_cvolume_init(&v);
}
}
- memcpy(out, out_frame.data_, ec->params.webrtc.blocksize);
+ memcpy(out, out_frame.data_, ec->params.webrtc.blocksize * pa_frame_size(ss));
}
void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
- const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
- apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.webrtc.blocksize / pa_frame_size(ss));
+ apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.webrtc.blocksize);
}
void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {