| 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 56da2820c6c10332c72b21925dc31c3489a7cbbb..1f73c5984a68af29d7dd941a64f88a834fd9e72a 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| @@ -1879,11 +1879,11 @@
|
| 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<int32_t>(
|
| + msg->set_num_input_channels(static_cast<google::protobuf::int32>(
|
| formats_.api_format.input_stream().num_channels()));
|
| - msg->set_num_output_channels(static_cast<int32_t>(
|
| + msg->set_num_output_channels(static_cast<google::protobuf::int32>(
|
| formats_.api_format.output_stream().num_channels()));
|
| - msg->set_num_reverse_channels(static_cast<int32_t>(
|
| + msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
|
| formats_.api_format.reverse_input_stream().num_channels()));
|
| msg->set_reverse_sample_rate(
|
| formats_.api_format.reverse_input_stream().sample_rate_hz());
|
| @@ -1953,7 +1953,7 @@
|
| }
|
| config.set_experiments_description(experiments_description);
|
|
|
| - ProtoString serialized_config = config.SerializeAsString();
|
| + std::string serialized_config = config.SerializeAsString();
|
| if (!forced &&
|
| debug_dump_.capture.last_serialized_config == serialized_config) {
|
| return kNoError;
|
|
|