[pulseaudio-discuss] [PATCH 11/11] echo-cancel: Start capture at a sane value if we're doing webrtc AGC

[arun at accosted.net]

From: Arun Raghavan <git at arunraghavan.net>

This is required to make sure the capture output has sufficient energy
for the AGC to do its job.
---
 src/modules/echo-cancel/echo-cancel.h |  1 +
 src/modules/echo-cancel/webrtc.cc     | 14 +++++++++++++-
 2 files changed, 14 insertions(+), 1 deletion(-)

diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h
index 142f8ac..b570095 100644
--- a/src/modules/echo-cancel/echo-cancel.h
+++ b/src/modules/echo-cancel/echo-cancel.h
@@ -68,6 +68,7 @@ struct pa_echo_canceller_params {
             pa_sample_spec sample_spec;
             bool agc;
             bool trace;
+            bool first;
         } webrtc;
 #endif
         /* each canceller-specific structure goes here */
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index a5d5c2e..b8781f4 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -52,6 +52,7 @@ PA_C_DECL_END
 #define DEFAULT_TRACE false
 
 #define WEBRTC_AGC_MAX_VOLUME 255
+#define WEBRTC_AGC_START_VOLUME 85
 
 static const char* const valid_modargs[] = {
     "high_pass_filter",
@@ -299,6 +300,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
     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;
 
     pa_modargs_free(ma);
     return true;
@@ -356,7 +358,17 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
     apm->ProcessStream(&out_frame);
 
     if (ec->params.priv.webrtc.agc) {
-        new_volume = apm->gain_control()->stream_analog_level();
+        if (PA_UNLIKELY(ec->params.priv.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;
+            new_volume = WEBRTC_AGC_START_VOLUME;
+        } else {
+            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);
-- 
2.4.3