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echo-cancel: Hook up WebRTC analog gain control

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This commit is contained in:
Arun Raghavan 2011-11-04 16:38:30 +05:30
parent 3d2f2424eb
commit fb02d50fe3
2 changed files with 25 additions and 7 deletions
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1
src/modules/echo-cancel/echo-cancel.h
View file

@ -58,6 +58,7 @@ struct pa_echo_canceller_params {
void *apm;
uint32_t blocksize;
pa_sample_spec sample_spec;
pa_bool_t agc;
} webrtc;
#endif
/* each canceller-specific structure goes here */

31
src/modules/echo-cancel/webrtc.cc
View file

@ -110,7 +110,7 @@ pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
goto fail;
}
dgc = DEFAULT_DIGITAL_GAIN_CONTROL;
dgc = agc ? FALSE : DEFAULT_DIGITAL_GAIN_CONTROL;
if (pa_modargs_get_value_boolean(ma, "digital_gain_control", &dgc) < 0) {
pa_log("Failed to parse digital_gain_control value");
goto fail;
@ -193,15 +193,20 @@ pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
}
if (agc || dgc) {
if (mobile && rm <= webrtc::EchoControlMobile::kEarpiece)
if (mobile && rm <= webrtc::EchoControlMobile::kEarpiece) {
/* Maybe this should be a knob, but we've got a lot of knobs already */
apm->gain_control()->set_mode(webrtc::GainControl::kFixedDigital);
else if (dgc)
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
else {
/* FIXME: Hook up for analog AGC */
pa_log("Analog gain control isn't implemented yet -- using ditital gain control.");
ec->params.priv.webrtc.agc = FALSE;
} else if (dgc) {
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
ec->params.priv.webrtc.agc = FALSE;
} else {
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
if (apm->gain_control()->set_analog_level_limits(0, PA_VOLUME_NORM-1) != apm->kNoError) {
pa_log("Failed to initialise AGC");
goto fail;
}
ec->params.priv.webrtc.agc = TRUE;
}
apm->gain_control()->Enable(true);
@ -242,15 +247,27 @@ 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;
pa_cvolume v;
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);
if (ec->params.priv.webrtc.agc) {
pa_cvolume_init(&v);
pa_echo_canceller_get_capture_volume(ec, &v);
apm->gain_control()->set_stream_analog_level(pa_cvolume_avg(&v));
}
apm->set_stream_delay_ms(0);
apm->ProcessStream(&out_frame);
if (ec->params.priv.webrtc.agc) {
pa_cvolume_set(&v, ss->channels, apm->gain_control()->stream_analog_level());
pa_echo_canceller_set_capture_volume(ec, &v);
}
memcpy(out, out_frame._payloadData, ec->params.priv.webrtc.blocksize);
}