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https://gitlab.freedesktop.org/pulseaudio/pulseaudio.git
synced 2025-11-02 09:01:46 -05:00
echo-cancel: Improve webrtc canceller error handling a bit
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parent
5baecd37c3
commit
222a98846c
1 changed files with 8 additions and 4 deletions
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@ -286,7 +286,10 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse input stream */
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webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse output stream */
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};
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apm->Initialize(pconfig);
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if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
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pa_log("Error initialising audio processing module");
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goto fail;
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}
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if (hpf)
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apm->high_pass_filter()->Enable(true);
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@ -315,7 +318,8 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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ec->params.webrtc.agc = false;
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} else {
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apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
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if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) != apm->kNoError) {
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if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) !=
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webrtc::AudioProcessing::kNoError) {
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pa_log("Failed to initialise AGC");
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goto fail;
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}
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@ -363,7 +367,7 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
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pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
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memcpy(play_frame.data_, play, ec->params.webrtc.blocksize * pa_frame_size(ss));
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apm->ProcessReverseStream(&play_frame);
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pa_assert_se(apm->ProcessReverseStream(&play_frame) == webrtc::AudioProcessing::kNoError);
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/* FIXME: If ProcessReverseStream() makes any changes to the audio, such as
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* applying intelligibility enhancement, those changes don't have any
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@ -395,7 +399,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
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}
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apm->set_stream_delay_ms(0);
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apm->ProcessStream(&out_frame);
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pa_assert_se(apm->ProcessStream(&out_frame) == webrtc::AudioProcessing::kNoError);
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if (ec->params.webrtc.agc) {
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if (PA_UNLIKELY(ec->params.webrtc.first)) {
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