[pulseaudio-discuss] [PATCH v3 10/24] echo-cancel: Deal with volume limit breakage in webrtc AGC
arun at accosted.net
arun at accosted.net
Sun Jan 17 23:36:24 PST 2016
From: Arun Raghavan <git at arunraghavan.net>
The AGC code no longer seems to honour the analog volume limits we set,
and internally uses 0-255 as the volume range. So we switch to use that
(keeping the old API usage as is in case this gets fixed upstream).
---
src/modules/echo-cancel/webrtc.cc | 25 +++++++++++++++++++++----
1 file changed, 21 insertions(+), 4 deletions(-)
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index 53ab7e1..c2585a8 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -51,6 +51,8 @@ PA_C_DECL_END
#define DEFAULT_INTELLIGIBILITY_ENHANCER false
#define DEFAULT_TRACE false
+#define WEBRTC_AGC_MAX_VOLUME 255
+
static const char* const valid_modargs[] = {
"high_pass_filter",
"noise_suppression",
@@ -95,6 +97,16 @@ class PaWebrtcTraceCallback : public webrtc::TraceCallback {
}
};
+static int webrtc_volume_from_pa(pa_volume_t v)
+{
+ return (v * WEBRTC_AGC_MAX_VOLUME) / PA_VOLUME_NORM;
+}
+
+static pa_volume_t webrtc_volume_to_pa(int v)
+{
+ return (v * PA_VOLUME_NORM) / WEBRTC_AGC_MAX_VOLUME;
+}
+
static void pa_webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map,
pa_sample_spec *play_ss, pa_channel_map *play_map,
pa_sample_spec *out_ss, pa_channel_map *out_map)
@@ -269,7 +281,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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) {
+ if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) != apm->kNoError) {
pa_log("Failed to initialise AGC");
goto fail;
}
@@ -321,6 +333,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
webrtc::AudioFrame out_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
pa_cvolume v;
+ int old_volume, new_volume;
out_frame.num_channels_ = ss->channels;
out_frame.sample_rate_hz_ = ss->rate;
@@ -333,15 +346,19 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
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));
+ old_volume = webrtc_volume_from_pa(pa_cvolume_avg(&v));
+ apm->gain_control()->set_stream_analog_level(old_volume);
}
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);
+ new_volume = apm->gain_control()->stream_analog_level();
+ if (old_volume != new_volume) {
+ pa_cvolume_set(&v, ss->channels, webrtc_volume_to_pa(new_volume));
+ pa_echo_canceller_set_capture_volume(ec, &v);
+ }
}
memcpy(out, out_frame.data_, ec->params.priv.webrtc.blocksize);
--
2.5.0
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