On Mon, 2016-01-18 at 13:06 +0530, arun at accosted.net wrote:
> From: Arun Raghavan <git at arunraghavan.net>
>
> ---
> Â src/modules/echo-cancel/webrtc.cc | 14 ++++++++++----
> Â 1 file changed, 10 insertions(+), 4 deletions(-)
>
> diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
> index 2732b38..5741f45 100644
> --- a/src/modules/echo-cancel/webrtc.cc
> +++ b/src/modules/echo-cancel/webrtc.cc
> @@ -284,7 +284,10 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
> Â Â Â Â Â Â Â Â Â webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse input stream */
> Â Â Â Â Â Â Â Â Â webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse output stream */
> Â Â Â Â Â };
> -Â Â Â Â apm->Initialize(pconfig);
> +Â Â Â Â if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
> +Â Â Â Â Â Â Â Â pa_log("Error initialising audio processing module");
> +Â Â Â Â Â Â Â Â goto fail;
> +Â Â Â Â }
> Â
> Â Â Â Â Â if (hpf)
> Â Â Â Â Â Â Â Â Â apm->high_pass_filter()->Enable(true);
> @@ -313,7 +316,8 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
> Â Â Â Â Â Â Â Â Â Â Â Â Â ec->params.webrtc.agc = false;
> Â Â Â Â Â Â Â Â Â } else {
> Â Â Â Â Â Â Â Â Â Â Â Â Â apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
> -Â Â Â Â Â Â Â Â Â Â Â Â if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) != apm->kNoError) {
> +Â Â Â Â Â Â Â Â Â Â Â Â if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) !=
> +Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â webrtc::AudioProcessing::kNoError) {
> Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â pa_log("Failed to initialise AGC");
> Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â goto fail;
> Â Â Â Â Â Â Â Â Â Â Â Â Â }
> @@ -361,7 +365,8 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
> Â Â Â Â Â pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
> Â Â Â Â Â memcpy(play_frame.data_, play, ec->params.webrtc.blocksize * pa_frame_size(ss));
> Â
> -Â Â Â Â apm->ProcessReverseStream(&play_frame);
> +Â Â Â Â if (apm->ProcessReverseStream(&play_frame) != webrtc::AudioProcessing::kNoError)
> +Â Â Â Â Â Â Â Â pa_log("Failed to process playback stream");
> Â }
> Â
> Â void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
> @@ -387,7 +392,8 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
> Â Â Â Â Â }
> Â
> Â Â Â Â Â apm->set_stream_delay_ms(0);
> -Â Â Â Â apm->ProcessStream(&out_frame);
> +Â Â Â Â if (apm->ProcessStream(&out_frame) != webrtc::AudioProcessing::kNoError)
> +Â Â Â Â Â Â Â Â pa_log("Failed to process capture stream");
I don't like error messages that can spam the syslog. Would a hard
failure be acceptable here (that is, unload the module, or change to a
failure state where the module stays around but does no processing)?
--Â
Tanu
