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| author | Arun Raghavan <git@arunraghavan.net> | 2016-02-17 19:47:09 +0530 |
|---|---|---|
| committer | Arun Raghavan <git@arunraghavan.net> | 2016-02-25 09:09:13 +0530 |
| commit | 222a98846c8c0b0c2b1b6adccf248b6a181f634e (patch) | |
| tree | f13da819656d6557c288c6eb9e1f145b5bffc7a7 | |
| parent | 5baecd37c3e17f1a5f6a3ef2d38f2da20db498d7 (diff) | |
| download | pulseaudio-222a98846c8c0b0c2b1b6adccf248b6a181f634e.tar.gz | |
echo-cancel: Improve webrtc canceller error handling a bit
| -rw-r--r-- | src/modules/echo-cancel/webrtc.cc | 12 |
1 files changed, 8 insertions, 4 deletions
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc index 0bf31dc24..1df9b6173 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 +367,7 @@ 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.webrtc.blocksize * pa_frame_size(ss)); - apm->ProcessReverseStream(&play_frame); + pa_assert_se(apm->ProcessReverseStream(&play_frame) == webrtc::AudioProcessing::kNoError); /* FIXME: If ProcessReverseStream() makes any changes to the audio, such as * applying intelligibility enhancement, those changes don't have any @@ -395,7 +399,7 @@ 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)) { |