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echo-cancel: Plug in WebRTC drift compensation

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This adds the ability for echo cancellers to provide their own drift
compensation, and hooks in the appropriate bits to implement this in the
WebRTC canceller.

We do this by introducing an alternative model for the canceller. So
far, the core engine just provided a run() method which was given
blocksize-sized chunks of playback and record samples. The new model has
the engine provide play() and record() methods that can (in theory) be
called by the playback and capture threads. The latter would actually do
the processing required.

In addition to this a set_drift() method may be provided by the
implementation. PA will provide periodic samples of the drift to the
engine. These values need to be aggregated and processed over some time,
since the point values vary quite a bit (but generally fit a linear
regression reasonably accurately). At some point of time, we might move
the actual drift calculation into PA and change the semantics of this
function.

NOTE: This needs further testing before being deemed ready for wider use.
This commit is contained in:
Arun Raghavan 2011-10-07 16:28:11 +05:30
parent 8c0cca7905
commit 23ce9a4f79
3 changed files with 250 additions and 68 deletions
Download patch file Download diff file Expand all files Collapse all files

46
src/modules/echo-cancel/webrtc.cc
View file

@ -47,6 +47,7 @@ PA_C_DECL_END
#define DEFAULT_MOBILE FALSE
#define DEFAULT_ROUTING_MODE "speakerphone"
#define DEFAULT_COMFORT_NOISE TRUE
#define DEFAULT_DRIFT_COMPENSATION FALSE
static const char* const valid_modargs[] = {
"high_pass_filter",
@ -56,6 +57,7 @@ static const char* const valid_modargs[] = {
"mobile",
"routing_mode",
"comfort_noise",
"drift_compensation",
NULL
};
@ -125,7 +127,18 @@ pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
goto fail;
}
ec->params.drift_compensation = DEFAULT_DRIFT_COMPENSATION;
if (pa_modargs_get_value_boolean(ma, "drift_compensation", &ec->params.drift_compensation) < 0) {
pa_log("Failed to parse drift_compensation value");
goto fail;
}
if (mobile) {
if (ec->params.drift_compensation) {
pa_log("Can't use drift_compensation in mobile mode");
goto fail;
}
if ((rm = routing_mode_from_string(pa_modargs_get_value(ma, "routing_mode", DEFAULT_ROUTING_MODE))) < 0) {
pa_log("Failed to parse routing_mode value");
goto fail;
@ -160,7 +173,13 @@ pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
apm->high_pass_filter()->Enable(true);
if (!mobile) {
apm->echo_cancellation()->enable_drift_compensation(false);
if (ec->params.drift_compensation) {
apm->echo_cancellation()->set_device_sample_rate_hz(source_ss->rate);
apm->echo_cancellation()->enable_drift_compensation(true);
} else {
apm->echo_cancellation()->enable_drift_compensation(false);
}
apm->echo_cancellation()->Enable(true);
} else {
apm->echo_control_mobile()->set_routing_mode(static_cast<webrtc::EchoControlMobile::RoutingMode>(rm));
@ -204,9 +223,9 @@ fail:
return FALSE;
}
void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
webrtc::AudioFrame play_frame, out_frame;
webrtc::AudioFrame play_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
play_frame._audioChannel = ss->channels;
@ -214,18 +233,37 @@ void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *
play_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
memcpy(play_frame._payloadData, play, ec->params.priv.webrtc.blocksize);
apm->AnalyzeReverseStream(&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.priv.webrtc.apm;
webrtc::AudioFrame out_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
out_frame._audioChannel = ss->channels;
out_frame._frequencyInHz = ss->rate;
out_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
memcpy(out_frame._payloadData, rec, ec->params.priv.webrtc.blocksize);
apm->AnalyzeReverseStream(&play_frame);
apm->set_stream_delay_ms(0);
apm->ProcessStream(&out_frame);
memcpy(out, out_frame._payloadData, ec->params.priv.webrtc.blocksize);
}
void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.priv.webrtc.blocksize / pa_frame_size(ss));
}
void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
pa_webrtc_ec_play(ec, play);
pa_webrtc_ec_record(ec, rec, out);
}
void pa_webrtc_ec_done(pa_echo_canceller *ec) {
if (ec->params.priv.webrtc.apm) {
webrtc::AudioProcessing::Destroy((webrtc::AudioProcessing*)ec->params.priv.webrtc.apm);