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-rw-r--r--content/renderer/media/webrtc_audio_capturer.cc7
1 files changed, 5 insertions, 2 deletions
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 3299987..45034e8 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -579,10 +579,13 @@ int WebRtcAudioCapturer::GetBufferSize(int sample_rate) const {
// Use the native hardware buffer size in non peer connection mode when the
// platform is using a native buffer size smaller than the PeerConnection
- // buffer size.
+ // buffer size and audio processing is off.
int hardware_buffer_size = device_info_.device.input.frames_per_buffer;
if (!peer_connection_mode_ && hardware_buffer_size &&
- hardware_buffer_size <= peer_connection_buffer_size) {
+ hardware_buffer_size <= peer_connection_buffer_size &&
+ !audio_processor_->has_audio_processing()) {
+ DVLOG(1) << "WebRtcAudioCapturer is using hardware buffer size "
+ << hardware_buffer_size;
return hardware_buffer_size;
}