| 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 {
|
|
|