From: Arun Raghavan <[email protected]>

---
 src/modules/echo-cancel/webrtc.cc | 13 +++++++++----
 1 file changed, 9 insertions(+), 4 deletions(-)

diff --git a/src/modules/echo-cancel/webrtc.cc 
b/src/modules/echo-cancel/webrtc.cc
index ffd60c2..d918656 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -286,7 +286,10 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
         webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* 
reverse input stream */
         webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* 
reverse output stream */
     };
-    apm->Initialize(pconfig);
+    if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
+        pa_log("Error initialising audio processing module");
+        goto fail;
+    }
 
     if (hpf)
         apm->high_pass_filter()->Enable(true);
@@ -315,7 +318,8 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
             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) {
+            if (apm->gain_control()->set_analog_level_limits(0, 
WEBRTC_AGC_MAX_VOLUME) !=
+                    webrtc::AudioProcessing::kNoError) {
                 pa_log("Failed to initialise AGC");
                 goto fail;
             }
@@ -363,7 +367,8 @@ 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.webrtc.blocksize * 
pa_frame_size(ss));
 
-    apm->ProcessReverseStream(&play_frame);
+    if (apm->ProcessReverseStream(&play_frame) != 
webrtc::AudioProcessing::kNoError)
+        pa_log("Failed to process playback stream");
 }
 
 void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t 
*out) {
@@ -389,7 +394,7 @@ 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);
+    pa_assert_se(apm->ProcessStream(&out_frame) == 
webrtc::AudioProcessing::kNoError);
 
     if (ec->params.webrtc.agc) {
         if (PA_UNLIKELY(ec->params.webrtc.first)) {
-- 
2.5.0

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