From: Arun Raghavan <git@xxxxxxxxxxxxxxxx> --- src/modules/echo-cancel/webrtc.cc | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc index d5d54dd..2732b38 100644 --- a/src/modules/echo-cancel/webrtc.cc +++ b/src/modules/echo-cancel/webrtc.cc @@ -355,7 +355,7 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) { play_frame.num_channels_ = ss->channels; play_frame.sample_rate_hz_ = ss->rate; - play_frame.interleaved_ = false; + play_frame.interleaved_ = true; play_frame.samples_per_channel_ = ec->params.webrtc.blocksize; pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples); @@ -373,7 +373,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out out_frame.num_channels_ = ss->channels; out_frame.sample_rate_hz_ = ss->rate; - out_frame.interleaved_ = false; + out_frame.interleaved_ = true; out_frame.samples_per_channel_ = ec->params.webrtc.blocksize; pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples); -- 2.5.0