Index: webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc |
diff --git a/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc b/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc |
index 2d7affcf4d5e0ab402fcc1a74e374cb0f38d168c..5a49685494da10d66b57d6fdbf83288c4e971a6a 100644 |
--- a/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc |
+++ b/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc |
@@ -46,7 +46,7 @@ void CaptureStreamInfo::AddInput(const AudioFrame& frame) { |
auto* stream = task_->GetEvent()->mutable_stream(); |
const size_t data_size = |
sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_; |
- stream->set_input_data(frame.data_, data_size); |
+ stream->set_input_data(frame.data(), data_size); |
} |
void CaptureStreamInfo::AddOutput(const AudioFrame& frame) { |
@@ -54,7 +54,7 @@ void CaptureStreamInfo::AddOutput(const AudioFrame& frame) { |
auto* stream = task_->GetEvent()->mutable_stream(); |
const size_t data_size = |
sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_; |
- stream->set_output_data(frame.data_, data_size); |
+ stream->set_output_data(frame.data(), data_size); |
} |
void CaptureStreamInfo::AddAudioProcessingState( |