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echo-cancel: Fix webrtc canceller when rec channels != play channels

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The calculations around how many samples were sent to the canceller
engine was not updated when we started supporting different channel
counts for playback and capture.
This commit is contained in:
Arun Raghavan 2016-02-17 19:47:08 +05:30
parent 08afc36ae4
commit 5baecd37c3
2 changed files with 14 additions and 14 deletions
Download patch file Download diff file Expand all files Collapse all files

4
src/modules/echo-cancel/echo-cancel.h
View file

@ -64,8 +64,8 @@ struct pa_echo_canceller_params {
/* This is a void* so that we don't have to convert this whole file /* This is a void* so that we don't have to convert this whole file
* 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;
uint32_t blocksize; unsigned int blocksize; /* in frames */
pa_sample_spec sample_spec; pa_sample_spec rec_ss, play_ss;
void *trace_callback; void *trace_callback;
bool agc; bool agc;
bool first; bool first;

24
src/modules/echo-cancel/webrtc.cc
View file

@ -329,9 +329,10 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
apm->voice_detection()->Enable(true); apm->voice_detection()->Enable(true);
ec->params.webrtc.apm = apm; ec->params.webrtc.apm = apm;
ec->params.webrtc.sample_spec = *out_ss; ec->params.webrtc.rec_ss = *rec_ss;
ec->params.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC; ec->params.webrtc.play_ss = *play_ss;
*nframes = ec->params.webrtc.blocksize / pa_frame_size(out_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; ec->params.webrtc.first = true;
pa_modargs_free(ma); pa_modargs_free(ma);
@ -352,15 +353,15 @@ fail:
void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) { void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm; webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
webrtc::AudioFrame play_frame; 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.num_channels_ = ss->channels;
play_frame.sample_rate_hz_ = ss->rate; play_frame.sample_rate_hz_ = ss->rate;
play_frame.interleaved_ = true; 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); 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); apm->ProcessReverseStream(&play_frame);
@ -374,17 +375,17 @@ 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.sample_spec; const pa_sample_spec *ss = &ec->params.webrtc.rec_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_ = ss->channels;
out_frame.sample_rate_hz_ = ss->rate; out_frame.sample_rate_hz_ = ss->rate;
out_frame.interleaved_ = true; 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); 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) { if (ec->params.webrtc.agc) {
pa_cvolume_init(&v); pa_cvolume_init(&v);
@ -414,14 +415,13 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
} }
} }
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) { void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm; 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) { void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {