| 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));
|
|
|