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 4c9188f7682df48404c8d70ce31e6ec2dc944ad0..9d05fbfb0c92be76fc787b0876cffc3462662b05 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -1077,7 +1077,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) { |
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); |
const size_t data_size = |
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
- msg->set_input_data(frame->data_, data_size); |
+ msg->set_input_data(frame->data(), data_size); |
} |
#endif |
@@ -1091,7 +1091,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) { |
audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); |
const size_t data_size = |
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
- msg->set_output_data(frame->data_, data_size); |
+ msg->set_output_data(frame->data(), data_size); |
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), |
&debug_dump_.num_bytes_left_for_log_, |
&crit_debug_, &debug_dump_.capture)); |
@@ -1409,7 +1409,7 @@ int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) { |
debug_dump_.render.event_msg->mutable_reverse_stream(); |
const size_t data_size = |
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
- msg->set_data(frame->data_, data_size); |
+ msg->set_data(frame->data(), data_size); |
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), |
&debug_dump_.num_bytes_left_for_log_, |
&crit_debug_, &debug_dump_.render)); |