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echo-cancel: Use anonymous unions for echo canceller params

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Makes this part of the code just a little less verbose.
This commit is contained in:
Arun Raghavan 2016-02-17 19:47:06 +05:30
parent aa02e1654b
commit 8de7dfec14
5 changed files with 71 additions and 70 deletions
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66
src/modules/echo-cancel/webrtc.cc
View file

@ -259,8 +259,8 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
if (trace) {
webrtc::Trace::CreateTrace();
webrtc::Trace::set_level_filter(webrtc::kTraceAll);
ec->params.priv.webrtc.trace_callback = new PaWebrtcTraceCallback();
webrtc::Trace::SetTraceCallback((PaWebrtcTraceCallback *) ec->params.priv.webrtc.trace_callback);
ec->params.webrtc.trace_callback = new PaWebrtcTraceCallback();
webrtc::Trace::SetTraceCallback((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
}
pa_webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map);
@ -296,17 +296,17 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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);
ec->params.priv.webrtc.agc = false;
ec->params.webrtc.agc = false;
} else if (dgc) {
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
ec->params.priv.webrtc.agc = false;
ec->params.webrtc.agc = false;
} else {
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) != apm->kNoError) {
pa_log("Failed to initialise AGC");
goto fail;
}
ec->params.priv.webrtc.agc = true;
ec->params.webrtc.agc = true;
}
apm->gain_control()->Enable(true);
@ -315,11 +315,11 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
if (vad)
apm->voice_detection()->Enable(true);
ec->params.priv.webrtc.apm = apm;
ec->params.priv.webrtc.sample_spec = *out_ss;
ec->params.priv.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
*nframes = ec->params.priv.webrtc.blocksize / pa_frame_size(out_ss);
ec->params.priv.webrtc.first = 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.first = true;
pa_modargs_free(ma);
return true;
@ -327,9 +327,9 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
fail:
if (ma)
pa_modargs_free(ma);
if (ec->params.priv.webrtc.trace_callback) {
if (ec->params.webrtc.trace_callback) {
webrtc::Trace::ReturnTrace();
delete ((PaWebrtcTraceCallback *) ec->params.priv.webrtc.trace_callback);
delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
} if (apm)
delete apm;
@ -337,17 +337,17 @@ fail:
}
void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
webrtc::AudioFrame play_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
play_frame.num_channels_ = ss->channels;
play_frame.sample_rate_hz_ = ss->rate;
play_frame.interleaved_ = true;
play_frame.samples_per_channel_ = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
play_frame.samples_per_channel_ = ec->params.webrtc.blocksize / pa_frame_size(ss);
pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
memcpy(play_frame.data_, play, ec->params.priv.webrtc.blocksize);
memcpy(play_frame.data_, play, ec->params.webrtc.blocksize);
apm->ProcessReverseStream(&play_frame);
@ -359,21 +359,21 @@ 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) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
webrtc::AudioFrame out_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
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.priv.webrtc.blocksize / pa_frame_size(ss);
out_frame.samples_per_channel_ = ec->params.webrtc.blocksize / pa_frame_size(ss);
pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
memcpy(out_frame.data_, rec, ec->params.priv.webrtc.blocksize);
memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize);
if (ec->params.priv.webrtc.agc) {
if (ec->params.webrtc.agc) {
pa_cvolume_init(&v);
pa_echo_canceller_get_capture_volume(ec, &v);
old_volume = webrtc_volume_from_pa(pa_cvolume_avg(&v));
@ -383,13 +383,13 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
apm->set_stream_delay_ms(0);
apm->ProcessStream(&out_frame);
if (ec->params.priv.webrtc.agc) {
if (PA_UNLIKELY(ec->params.priv.webrtc.first)) {
if (ec->params.webrtc.agc) {
if (PA_UNLIKELY(ec->params.webrtc.first)) {
/* We start at a sane default volume (taken from the Chromium
* condition on the experimental AGC in audio_processing.h). This is
* needed to make sure that there's enough energy in the capture
* signal for the AGC to work */
ec->params.priv.webrtc.first = false;
ec->params.webrtc.first = false;
new_volume = WEBRTC_AGC_START_VOLUME;
} else {
new_volume = apm->gain_control()->stream_analog_level();
@ -401,14 +401,14 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
}
}
memcpy(out, out_frame.data_, ec->params.priv.webrtc.blocksize);
memcpy(out, out_frame.data_, ec->params.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;
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.priv.webrtc.blocksize / pa_frame_size(ss));
apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.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) {
@ -417,13 +417,13 @@ void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *
}
void pa_webrtc_ec_done(pa_echo_canceller *ec) {
if (ec->params.priv.webrtc.trace_callback) {
if (ec->params.webrtc.trace_callback) {
webrtc::Trace::ReturnTrace();
delete ((PaWebrtcTraceCallback *) ec->params.priv.webrtc.trace_callback);
delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
}
if (ec->params.priv.webrtc.apm) {
delete (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
ec->params.priv.webrtc.apm = NULL;
if (ec->params.webrtc.apm) {
delete (webrtc::AudioProcessing*)ec->params.webrtc.apm;
ec->params.webrtc.apm = NULL;
}
}