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 922997e5c7cb8bd60312344ad4a0bccebdd83671..9ece91fa9b2322f982712c4fdeb3d8a9dcb5cae0 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -1160,7 +1160,7 @@ 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 |
@@ -1178,7 +1178,7 @@ 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)); |
@@ -1514,7 +1514,7 @@ 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)); |