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echo-cancel: webrtc canceller supports different in/out channel counts

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Needed for upcoming beamforming code.
This commit is contained in:
Arun Raghavan 2016-02-17 19:47:10 +05:30
parent 222a98846c
commit 3133ff8e11
2 changed files with 9 additions and 7 deletions
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2
src/modules/echo-cancel/echo-cancel.h
View file

@ -65,7 +65,7 @@ struct pa_echo_canceller_params {
* to C++ linkage. apm is a pointer to an AudioProcessing object */ * to C++ linkage. apm is a pointer to an AudioProcessing object */
void *apm; void *apm;
unsigned int blocksize; /* in frames */ unsigned int blocksize; /* in frames */
pa_sample_spec rec_ss, play_ss; pa_sample_spec rec_ss, play_ss, out_ss;
void *trace_callback; void *trace_callback;
bool agc; bool agc;
bool first; bool first;

14
src/modules/echo-cancel/webrtc.cc
View file

@ -335,6 +335,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
ec->params.webrtc.apm = apm; ec->params.webrtc.apm = apm;
ec->params.webrtc.rec_ss = *rec_ss; ec->params.webrtc.rec_ss = *rec_ss;
ec->params.webrtc.play_ss = *play_ss; ec->params.webrtc.play_ss = *play_ss;
ec->params.webrtc.out_ss = *out_ss;
ec->params.webrtc.blocksize = (uint64_t) out_ss->rate * BLOCK_SIZE_US / PA_USEC_PER_SEC; ec->params.webrtc.blocksize = (uint64_t) out_ss->rate * BLOCK_SIZE_US / PA_USEC_PER_SEC;
*nframes = ec->params.webrtc.blocksize; *nframes = ec->params.webrtc.blocksize;
ec->params.webrtc.first = true; ec->params.webrtc.first = true;
@ -379,17 +380,18 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) { 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::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
webrtc::AudioFrame out_frame; webrtc::AudioFrame out_frame;
const pa_sample_spec *ss = &ec->params.webrtc.rec_ss; const pa_sample_spec *rec_ss = &ec->params.webrtc.rec_ss;
const pa_sample_spec *out_ss = &ec->params.webrtc.out_ss;
pa_cvolume v; pa_cvolume v;
int old_volume, new_volume; int old_volume, new_volume;
out_frame.num_channels_ = ss->channels; out_frame.num_channels_ = rec_ss->channels;
out_frame.sample_rate_hz_ = ss->rate; out_frame.sample_rate_hz_ = rec_ss->rate;
out_frame.interleaved_ = true; out_frame.interleaved_ = true;
out_frame.samples_per_channel_ = ec->params.webrtc.blocksize; out_frame.samples_per_channel_ = ec->params.webrtc.blocksize;
pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples); pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize * pa_frame_size(ss)); memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize * pa_frame_size(rec_ss));
if (ec->params.webrtc.agc) { if (ec->params.webrtc.agc) {
pa_cvolume_init(&v); pa_cvolume_init(&v);
@ -414,12 +416,12 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
} }
if (old_volume != new_volume) { if (old_volume != new_volume) {
pa_cvolume_set(&v, ss->channels, webrtc_volume_to_pa(new_volume)); pa_cvolume_set(&v, rec_ss->channels, webrtc_volume_to_pa(new_volume));
pa_echo_canceller_set_capture_volume(ec, &v); pa_echo_canceller_set_capture_volume(ec, &v);
} }
} }
memcpy(out, out_frame.data_, ec->params.webrtc.blocksize * pa_frame_size(ss)); memcpy(out, out_frame.data_, ec->params.webrtc.blocksize * pa_frame_size(out_ss));
} }
void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) { void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {