Index: content/renderer/media/webrtc_audio_capturer.cc |
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc |
index aa3ecf8627f84ca4165d481afd14b50388faf291..3362e1dae2fe882d69a433aae397b4db1da59762 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.cc |
+++ b/content/renderer/media/webrtc_audio_capturer.cc |
@@ -453,11 +453,15 @@ bool WebRtcAudioCapturer::GetPairedOutputParameters( |
int* session_id, |
int* output_sample_rate, |
int* output_frames_per_buffer) const { |
+ // Don't set output parameters unless all of them are valid. |
+ if (session_id_ <= 0 || !output_sample_rate_ || !output_frames_per_buffer_) |
+ return false; |
+ |
*session_id = session_id_; |
*output_sample_rate = output_sample_rate_; |
*output_frames_per_buffer = output_frames_per_buffer_; |
- return session_id_ > 0 && output_sample_rate_ > 0 && |
- output_frames_per_buffer_> 0; |
+ |
+ return true; |
} |
int WebRtcAudioCapturer::GetBufferSize(int sample_rate) const { |