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 88229b4829aea7f0a4a9e68893c96f643e7944d1..f9cc98faa022846ea9e42e0f4eeba2945b8ba3ea 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -1878,11 +1878,11 @@ int AudioProcessingImpl::WriteInitMessage() { |
audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init(); |
msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz()); |
- msg->set_num_input_channels(static_cast<google::protobuf::int32>( |
+ msg->set_num_input_channels(static_cast<int32_t>( |
formats_.api_format.input_stream().num_channels())); |
- msg->set_num_output_channels(static_cast<google::protobuf::int32>( |
+ msg->set_num_output_channels(static_cast<int32_t>( |
formats_.api_format.output_stream().num_channels())); |
- msg->set_num_reverse_channels(static_cast<google::protobuf::int32>( |
+ msg->set_num_reverse_channels(static_cast<int32_t>( |
formats_.api_format.reverse_input_stream().num_channels())); |
msg->set_reverse_sample_rate( |
formats_.api_format.reverse_input_stream().sample_rate_hz()); |
@@ -1952,7 +1952,7 @@ int AudioProcessingImpl::WriteConfigMessage(bool forced) { |
} |
config.set_experiments_description(experiments_description); |
- std::string serialized_config = config.SerializeAsString(); |
+ ProtoString serialized_config = config.SerializeAsString(); |
if (!forced && |
debug_dump_.capture.last_serialized_config == serialized_config) { |
return kNoError; |