AecDump implementation.

webrtc/modules/audio_processing/aec_dump/capture_stream_info.h

Index: webrtc/modules/audio_processing/aec_dump/capture_stream_info.h
diff --git a/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h b/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h
new file mode 100644
index 0000000000000000000000000000000000000000..36b860cea4b0ee2a74fa0bd5be60dabbd044375d
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+++ b/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/ignore_wundef.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_processing/aec_dump/write_to_file_task.h"
+#include "webrtc/modules/audio_processing/include/aec_dump.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+// Files generated at build-time by the protobuf compiler.
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/modules/audio_processing/debug.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+
+class CaptureStreamInfo {
+ public:
+  explicit CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task);
+  ~CaptureStreamInfo();
+  void AddInput(const FloatAudioFrame& src);
+  void AddOutput(const FloatAudioFrame& src);
+
+  void AddInput(const AudioFrame& frame);
+  void AddOutput(const AudioFrame& frame);
+
+  void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
+
+  std::unique_ptr<WriteToFileTask> GetTask() {
+    RTC_DCHECK(task_);
+    return std::move(task_);
+  }
+
+  void SetTask(std::unique_ptr<WriteToFileTask> task) {
+    RTC_DCHECK(!task_);
+    RTC_DCHECK(task);
+    task_ = std::move(task);
+    task_->GetEvent()->set_type(audioproc::Event::STREAM);
+  }
+
+ private:
+  std::unique_ptr<WriteToFileTask> task_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_