From: Arun Raghavan <git@xxxxxxxxxxxxxxxx>
Makes this part of the code just a little less verbose.
---
src/modules/echo-cancel/adrian.c | 18 +++++-----
src/modules/echo-cancel/echo-cancel.h | 3 +-
src/modules/echo-cancel/null.c | 4 +--
src/modules/echo-cancel/speex.c | 50 +++++++++++++--------------
src/modules/echo-cancel/webrtc.cc | 64 +++++++++++++++++------------------
5 files changed, 70 insertions(+), 69 deletions(-)
diff --git a/src/modules/echo-cancel/adrian.c b/src/modules/echo-cancel/adrian.c
index 60a2b66..3c47fae 100644
--- a/src/modules/echo-cancel/adrian.c
+++ b/src/modules/echo-cancel/adrian.c
@@ -78,16 +78,16 @@ bool pa_adrian_ec_init(pa_core *c, pa_echo_canceller *ec,
rate = out_ss->rate;
*nframes = (rate * frame_size_ms) / 1000;
- ec->params.priv.adrian.blocksize = (*nframes) * pa_frame_size(out_ss);
+ ec->params.adrian.blocksize = (*nframes) * pa_frame_size(out_ss);
- pa_log_debug ("Using nframes %d, blocksize %u, channels %d, rate %d", *nframes, ec->params.priv.adrian.blocksize, out_ss->channels, out_ss->rate);
+ pa_log_debug ("Using nframes %d, blocksize %u, channels %d, rate %d", *nframes, ec->params.adrian.blocksize, out_ss->channels, out_ss->rate);
/* For now we only support SSE */
if (c->cpu_info.cpu_type == PA_CPU_X86 && (c->cpu_info.flags.x86 & PA_CPU_X86_SSE))
have_vector = 1;
- ec->params.priv.adrian.aec = AEC_init(rate, have_vector);
- if (!ec->params.priv.adrian.aec)
+ ec->params.adrian.aec = AEC_init(rate, have_vector);
+ if (!ec->params.adrian.aec)
goto fail;
pa_modargs_free(ma);
@@ -102,17 +102,17 @@ fail:
void pa_adrian_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
unsigned int i;
- for (i = 0; i < ec->params.priv.adrian.blocksize; i += 2) {
+ for (i = 0; i < ec->params.adrian.blocksize; i += 2) {
/* We know it's S16NE mono data */
int r = *(int16_t *)(rec + i);
int p = *(int16_t *)(play + i);
- *(int16_t *)(out + i) = (int16_t) AEC_doAEC(ec->params.priv.adrian.aec, r, p);
+ *(int16_t *)(out + i) = (int16_t) AEC_doAEC(ec->params.adrian.aec, r, p);
}
}
void pa_adrian_ec_done(pa_echo_canceller *ec) {
- if (ec->params.priv.adrian.aec) {
- AEC_done(ec->params.priv.adrian.aec);
- ec->params.priv.adrian.aec = NULL;
+ if (ec->params.adrian.aec) {
+ AEC_done(ec->params.adrian.aec);
+ ec->params.adrian.aec = NULL;
}
}
diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h
index b570095..37f99c0 100644
--- a/src/modules/echo-cancel/echo-cancel.h
+++ b/src/modules/echo-cancel/echo-cancel.h
@@ -69,10 +69,11 @@ struct pa_echo_canceller_params {
bool agc;
bool trace;
bool first;
+ unsigned int agc_start_volume;
} webrtc;
#endif
/* each canceller-specific structure goes here */
- } priv;
+ };
/* Set this if canceller can do drift compensation. Also see set_drift()
* below */
diff --git a/src/modules/echo-cancel/null.c b/src/modules/echo-cancel/null.c
index 673b14f..c8ecf27 100644
--- a/src/modules/echo-cancel/null.c
+++ b/src/modules/echo-cancel/null.c
@@ -34,7 +34,7 @@ bool pa_null_ec_init(pa_core *c, pa_echo_canceller *ec,
char strss_sink[PA_SAMPLE_SPEC_SNPRINT_MAX];
*nframes = 256;
- ec->params.priv.null.out_ss = *out_ss;
+ ec->params.null.out_ss = *out_ss;
*rec_ss = *out_ss;
*rec_map = *out_map;
@@ -49,7 +49,7 @@ bool pa_null_ec_init(pa_core *c, pa_echo_canceller *ec,
void pa_null_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
/* The null implementation simply copies the recorded buffer to the output
buffer and ignores the play buffer. */
- memcpy(out, rec, 256 * pa_frame_size(&ec->params.priv.null.out_ss));
+ memcpy(out, rec, 256 * pa_frame_size(&ec->params.null.out_ss));
}
void pa_null_ec_done(pa_echo_canceller *ec) {
diff --git a/src/modules/echo-cancel/speex.c b/src/modules/echo-cancel/speex.c
index 11e53b3..08c1027 100644
--- a/src/modules/echo-cancel/speex.c
+++ b/src/modules/echo-cancel/speex.c
@@ -111,26 +111,26 @@ static bool pa_speex_ec_preprocessor_init(pa_echo_canceller *ec, pa_sample_spec
goto fail;
}
- ec->params.priv.speex.pp_state = speex_preprocess_state_init(nframes, out_ss->rate);
+ ec->params.speex.pp_state = speex_preprocess_state_init(nframes, out_ss->rate);
tmp = agc;
- speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_AGC, &tmp);
+ speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_AGC, &tmp);
tmp = denoise;
- speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp);
+ speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp);
if (echo_suppress) {
if (echo_suppress_attenuation)
- speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS,
+ speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS,
&echo_suppress_attenuation);
if (echo_suppress_attenuation_active) {
- speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE,
+ speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE,
&echo_suppress_attenuation_active);
}
- speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_STATE,
- ec->params.priv.speex.state);
+ speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_STATE,
+ ec->params.speex.state);
}
pa_log_info("Loaded speex preprocessor with params: agc=%s, denoise=%s, echo_suppress=%s", pa_yes_no(agc),
@@ -176,12 +176,12 @@ bool pa_speex_ec_init(pa_core *c, pa_echo_canceller *ec,
*nframes = pa_echo_canceller_blocksize_power2(rate, frame_size_ms);
pa_log_debug ("Using nframes %d, channels %d, rate %d", *nframes, out_ss->channels, out_ss->rate);
- ec->params.priv.speex.state = speex_echo_state_init_mc(*nframes, (rate * filter_size_ms) / 1000, out_ss->channels, out_ss->channels);
+ ec->params.speex.state = speex_echo_state_init_mc(*nframes, (rate * filter_size_ms) / 1000, out_ss->channels, out_ss->channels);
- if (!ec->params.priv.speex.state)
+ if (!ec->params.speex.state)
goto fail;
- speex_echo_ctl(ec->params.priv.speex.state, SPEEX_ECHO_SET_SAMPLING_RATE, &rate);
+ speex_echo_ctl(ec->params.speex.state, SPEEX_ECHO_SET_SAMPLING_RATE, &rate);
if (!pa_speex_ec_preprocessor_init(ec, out_ss, *nframes, ma))
goto fail;
@@ -192,34 +192,34 @@ bool pa_speex_ec_init(pa_core *c, pa_echo_canceller *ec,
fail:
if (ma)
pa_modargs_free(ma);
- if (ec->params.priv.speex.pp_state) {
- speex_preprocess_state_destroy(ec->params.priv.speex.pp_state);
- ec->params.priv.speex.pp_state = NULL;
+ if (ec->params.speex.pp_state) {
+ speex_preprocess_state_destroy(ec->params.speex.pp_state);
+ ec->params.speex.pp_state = NULL;
}
- if (ec->params.priv.speex.state) {
- speex_echo_state_destroy(ec->params.priv.speex.state);
- ec->params.priv.speex.state = NULL;
+ if (ec->params.speex.state) {
+ speex_echo_state_destroy(ec->params.speex.state);
+ ec->params.speex.state = NULL;
}
return false;
}
void pa_speex_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
- speex_echo_cancellation(ec->params.priv.speex.state, (const spx_int16_t *) rec, (const spx_int16_t *) play,
+ speex_echo_cancellation(ec->params.speex.state, (const spx_int16_t *) rec, (const spx_int16_t *) play,
(spx_int16_t *) out);
/* preprecessor is run after AEC. This is not a mistake! */
- if (ec->params.priv.speex.pp_state)
- speex_preprocess_run(ec->params.priv.speex.pp_state, (spx_int16_t *) out);
+ if (ec->params.speex.pp_state)
+ speex_preprocess_run(ec->params.speex.pp_state, (spx_int16_t *) out);
}
void pa_speex_ec_done(pa_echo_canceller *ec) {
- if (ec->params.priv.speex.pp_state) {
- speex_preprocess_state_destroy(ec->params.priv.speex.pp_state);
- ec->params.priv.speex.pp_state = NULL;
+ if (ec->params.speex.pp_state) {
+ speex_preprocess_state_destroy(ec->params.speex.pp_state);
+ ec->params.speex.pp_state = NULL;
}
- if (ec->params.priv.speex.state) {
- speex_echo_state_destroy(ec->params.priv.speex.state);
- ec->params.priv.speex.state = NULL;
+ if (ec->params.speex.state) {
+ speex_echo_state_destroy(ec->params.speex.state);
+ ec->params.speex.state = NULL;
}
}
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index 7c1501d..395d31e 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -249,13 +249,13 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
if (experimental_agc)
config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, WEBRTC_AGC_START_VOLUME));
- ec->params.priv.webrtc.trace = DEFAULT_TRACE;
- if (pa_modargs_get_value_boolean(ma, "trace", &ec->params.priv.webrtc.trace) < 0) {
+ ec->params.webrtc.trace = DEFAULT_TRACE;
+ if (pa_modargs_get_value_boolean(ma, "trace", &ec->params.webrtc.trace) < 0) {
pa_log("Failed to parse trace value");
goto fail;
}
- if (ec->params.priv.webrtc.trace) {
+ if (ec->params.webrtc.trace) {
webrtc::Trace::CreateTrace();
webrtc::Trace::set_level_filter(webrtc::kTraceAll);
webrtc::Trace::SetTraceCallback(new PaWebrtcTraceCallback());
@@ -294,17 +294,17 @@ 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);
@@ -313,11 +313,11 @@ 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;
@@ -325,7 +325,7 @@ 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)
+ if (ec->params.webrtc.trace)
webrtc::Trace::ReturnTrace();
if (apm)
delete apm;
@@ -334,37 +334,37 @@ 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);
}
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));
@@ -374,13 +374,13 @@ 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();
@@ -392,14 +392,14 @@ 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) {
@@ -408,11 +408,11 @@ 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)
+ if (ec->params.webrtc.trace)
webrtc::Trace::ReturnTrace();
- 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;
}
}
--
2.5.0
