1 files changed, 85 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
new file mode 100644
index 0000000000..fac3712b7a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
@@ -0,0 +1,85 @@
+/*
+ *  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 MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec_dump/capture_stream_info.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.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 "modules/audio_processing/debug.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+
+// Task-queue based implementation of AecDump. It is thread safe by
+// relying on locks in TaskQueue.
+class AecDumpImpl : public AecDump {
+ public:
+  // `max_log_size_bytes` - maximum number of bytes to write to the debug file,
+  // `max_log_size_bytes == -1` means the log size will be unlimited.
+  AecDumpImpl(FileWrapper debug_file,
+              int64_t max_log_size_bytes,
+              rtc::TaskQueue* worker_queue);
+  AecDumpImpl(const AecDumpImpl&) = delete;
+  AecDumpImpl& operator=(const AecDumpImpl&) = delete;
+  ~AecDumpImpl() override;
+
+  void WriteInitMessage(const ProcessingConfig& api_format,
+                        int64_t time_now_ms) override;
+  void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
+  void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;
+  void AddCaptureStreamInput(const int16_t* const data,
+                             int num_channels,
+                             int samples_per_channel) override;
+  void AddCaptureStreamOutput(const int16_t* const data,
+                              int num_channels,
+                              int samples_per_channel) override;
+  void AddAudioProcessingState(const AudioProcessingState& state) override;
+  void WriteCaptureStreamMessage() override;
+
+  void WriteRenderStreamMessage(const int16_t* const data,
+                                int num_channels,
+                                int samples_per_channel) override;
+  void WriteRenderStreamMessage(
+      const AudioFrameView<const float>& src) override;
+
+  void WriteConfig(const InternalAPMConfig& config) override;
+
+  void WriteRuntimeSetting(
+      const AudioProcessing::RuntimeSetting& runtime_setting) override;
+
+ private:
+  void PostWriteToFileTask(std::unique_ptr<audioproc::Event> event);
+
+  FileWrapper debug_file_;
+  int64_t num_bytes_left_for_log_ = 0;
+  rtc::RaceChecker race_checker_;
+  rtc::TaskQueue* worker_queue_;
+  CaptureStreamInfo capture_stream_info_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_