Index: webrtc/modules/audio_processing/audio_processing_impl.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc |
index f3856128bb6bbd83544c1901e25a5790d2633e95..92765a2a3c2e972db1b3b0f3ae1864ab913df10f 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -297,11 +297,11 @@ int AudioProcessingImpl::InitializeLocked() { |
formats_.rev_proc_format.num_channels(), |
rev_audio_buffer_out_num_frames)); |
if (rev_conversion_needed()) { |
- render_.render_converter = AudioConverter::Create( |
+ render_.render_converter = rtc::UniqueToScoped(AudioConverter::Create( |
formats_.api_format.reverse_input_stream().num_channels(), |
formats_.api_format.reverse_input_stream().num_frames(), |
formats_.api_format.reverse_output_stream().num_channels(), |
- formats_.api_format.reverse_output_stream().num_frames()); |
+ formats_.api_format.reverse_output_stream().num_frames())); |
} else { |
render_.render_converter.reset(nullptr); |
} |