Test submission of complete AEC-dump refactoring.

webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.h

Index: webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.h
diff --git a/webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.h b/webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.h
new file mode 100644
index 0000000000000000000000000000000000000000..4aa2fa43e10b380c3c5b54577262b0b1ca796587
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.h
@@ -0,0 +1,137 @@
+/*
+ *  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_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_IMPL_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/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 {
+
+// Optionally inherit from TaskQueue task.
+class CaptureStreamInfoImpl final : public AecDump::CaptureStreamInfo {
+ public:
+  explicit CaptureStreamInfoImpl(std::unique_ptr<audioproc::Event> event);
+  ~CaptureStreamInfoImpl() override;
+  void AddInput(const std::vector<rtc::ArrayView<const float>>& src) override;
+  void AddOutput(const std::vector<rtc::ArrayView<const float>>& src) override;
+
+  void AddInput(const AudioFrame& frame) override;
+  void AddOutput(const AudioFrame& frame) override;
+
+  void set_delay(int delay) override;
+  void set_drift(int drift) override;
+  void set_level(int level) override;
+  void set_keypress(bool keypress) override;
+
+  std::unique_ptr<audioproc::Event> GetEventMsg() {
+    return std::unique_ptr<audioproc::Event>(event_.release());
+  }
+
+ private:
+  std::unique_ptr<audioproc::Event> event_;
+};
+
+CaptureStreamInfoImpl::CaptureStreamInfoImpl(
+    std::unique_ptr<audioproc::Event> event)
+    : event_(std::move(event)) {
+  RTC_DCHECK(event_);
+  event_->set_type(audioproc::Event::STREAM);
+}
+
+CaptureStreamInfoImpl::~CaptureStreamInfoImpl() = default;
+
+void CaptureStreamInfoImpl::AddInput(
+    const std::vector<rtc::ArrayView<const float>>& src) {
+  if (!event_) {
+    return;
+  }
+  auto* stream = event_->mutable_stream();
+  for (const auto& channel_view : src) {
+    stream->add_input_channel(channel_view.begin(), channel_view.size());
+  }
+}
+
+void CaptureStreamInfoImpl::AddOutput(
+    const std::vector<rtc::ArrayView<const float>>& src) {
+  if (!event_) {
+    return;
+  }
+  auto* stream = event_->mutable_stream();
+  for (const auto& channel_view : src) {
+    stream->add_output_channel(channel_view.begin(), channel_view.size());
+  }
+}
+
+void CaptureStreamInfoImpl::AddInput(const AudioFrame& frame) {
+  if (!event_) {
+    return;
+  }
+  audioproc::Stream* stream = event_->mutable_stream();
+  const size_t data_size =
+      sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
+  stream->set_input_data(frame.data_, data_size);
+}
+
+void CaptureStreamInfoImpl::AddOutput(const AudioFrame& frame) {
+  if (!event_) {
+    return;
+  }
+  audioproc::Stream* stream = event_->mutable_stream();
+  const size_t data_size =
+      sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
+  stream->set_output_data(frame.data_, data_size);
+}
+
+void CaptureStreamInfoImpl::set_delay(int delay) {
+  if (!event_) {
+    return;
+  }
+  event_->mutable_stream()->set_delay(delay);
+}
+void CaptureStreamInfoImpl::set_drift(int drift) {
+  if (!event_) {
+    return;
+  }
+  event_->mutable_stream()->set_drift(drift);
+}
+void CaptureStreamInfoImpl::set_level(int level) {
+  if (!event_) {
+    return;
+  }
+  event_->mutable_stream()->set_level(level);
+}
+void CaptureStreamInfoImpl::set_keypress(bool keypress) {
+  if (!event_) {
+    return;
+  }
+  event_->mutable_stream()->set_keypress(keypress);
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_IMPL_H_