[pulseaudio-discuss] [PATCH 03/11] echo-cancel: Add support for the webrtc intelligibility enhancer

[arun at accosted.net]

From: Arun Raghavan <git at arunraghavan.net>

Just exposing this, disabled by default. It's not used by Chromium at
the moment.
---
 src/modules/echo-cancel/webrtc.cc | 14 ++++++++++++--
 1 file changed, 12 insertions(+), 2 deletions(-)

diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index 50ce1b3..85a7180 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -47,6 +47,7 @@ PA_C_DECL_END
 #define DEFAULT_COMFORT_NOISE true
 #define DEFAULT_DRIFT_COMPENSATION false
 #define DEFAULT_EXTENDED_FILTER false
+#define DEFAULT_INTELLIGIBILITY_ENHANCER false
 
 static const char* const valid_modargs[] = {
     "high_pass_filter",
@@ -58,6 +59,7 @@ static const char* const valid_modargs[] = {
     "comfort_noise",
     "drift_compensation",
     "extended_filter",
+    "intelligibility_enhancer",
     NULL
 };
 
@@ -84,7 +86,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
     webrtc::AudioProcessing *apm = NULL;
     webrtc::ProcessingConfig pconfig;
     webrtc::Config config;
-    bool hpf, ns, agc, dgc, mobile, cn, ext_filter;
+    bool hpf, ns, agc, dgc, mobile, cn, ext_filter, intelligibility;
     int rm = -1;
     pa_modargs *ma;
 
@@ -163,8 +165,16 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
         goto fail;
     }
 
+    intelligibility = DEFAULT_INTELLIGIBILITY_ENHANCER;
+    if (pa_modargs_get_value_boolean(ma, "intelligibility_enhancer", &intelligibility) < 0) {
+        pa_log("Failed to parse intelligibility_enhancer value");
+        goto fail;
+    }
+
     if (ext_filter)
         config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
+    if (intelligibility)
+        config.Set<webrtc::Intelligibility>(new webrtc::Intelligibility(true));
 
     apm = webrtc::AudioProcessing::Create(config);
 
@@ -253,7 +263,7 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
     pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
     memcpy(play_frame.data_, play, ec->params.priv.webrtc.blocksize);
 
-    apm->AnalyzeReverseStream(&play_frame);
+    apm->ProcessReverseStream(&play_frame);
 }
 
 void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
-- 
2.4.3