Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_processing/include/aec_dump.h

@@ -0,0 +1,141 @@
+/*
+ *  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_INCLUDE_AEC_DUMP_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "webrtc/api/array_view.h"
+
+namespace webrtc {
+
+class AudioFrame;
+
+// Struct for passing current config from APM without having to
+// include protobuf headers.
+struct InternalAPMConfig {
+  InternalAPMConfig();
+  InternalAPMConfig(const InternalAPMConfig&);
+  InternalAPMConfig(InternalAPMConfig&&);
+
+  InternalAPMConfig& operator=(const InternalAPMConfig&);
+  InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
+
+  bool operator==(const InternalAPMConfig& other);
+
+  bool aec_enabled = false;
+  bool aec_delay_agnostic_enabled = false;
+  bool aec_drift_compensation_enabled = false;
+  bool aec_extended_filter_enabled = false;
+  int aec_suppression_level = 0;
+  bool aecm_enabled = false;
+  bool aecm_comfort_noise_enabled = false;
+  int aecm_routing_mode = 0;
+  bool agc_enabled = false;
+  int agc_mode = 0;
+  bool agc_limiter_enabled = false;
+  bool hpf_enabled = false;
+  bool ns_enabled = false;
+  int ns_level = 0;
+  bool transient_suppression_enabled = false;
+  bool intelligibility_enhancer_enabled = false;
+  bool noise_robust_agc_enabled = false;
+  std::string experiments_description = "";
+};
+
+struct InternalAPMStreamsConfig {
+  int input_sample_rate = 0;
+  int output_sample_rate = 0;
+  int render_input_sample_rate = 0;
+  int render_output_sample_rate = 0;
+
+  size_t input_num_channels = 0;
+  size_t output_num_channels = 0;
+  size_t render_input_num_channels = 0;
+  size_t render_output_num_channels = 0;
+};
+
+// Class to pass audio data in float** format. This is to avoid
+// dependence on AudioBuffer, and avoid problems associated with
+// rtc::ArrayView<rtc::ArrayView>.
+class FloatAudioFrame {
+ public:
+  // |num_channels| and |channel_size| describe the float**
+  // |audio_samples|. |audio_samples| is assumed to point to a
+  // two-dimensional |num_channels * channel_size| array of floats.
+  FloatAudioFrame(const float* const* audio_samples,
+                  size_t num_channels,
+                  size_t channel_size)
+      : audio_samples_(audio_samples),
+        num_channels_(num_channels),
+        channel_size_(channel_size) {}
+
+  FloatAudioFrame() = delete;
+
+  size_t num_channels() const { return num_channels_; }
+
+  rtc::ArrayView<const float> channel(size_t idx) const {
+    RTC_DCHECK_LE(0, idx);
+    RTC_DCHECK_LE(idx, num_channels_);
+    return rtc::ArrayView<const float>(audio_samples_[idx], channel_size_);
+  }
+
+ private:
+  const float* const* audio_samples_;
+  size_t num_channels_;
+  size_t channel_size_;
+};
+
+// An interface for recording configuration and input/output streams
+// of the Audio Processing Module. The recordings are called
+// 'aec-dumps' and are stored in a protobuf format defined in
+// debug.proto.
+// The Write* methods are always safe to call concurrently or
+// otherwise for all implementing subclasses. The intended mode of
+// operation is to create a protobuf object from the input, and send
+// it away to be written to file asynchronously.
+class AecDump {
+ public:
+  struct AudioProcessingState {
+    int delay;
+    int drift;
+    int level;
+    bool keypress;
+  };
+
+  virtual ~AecDump() = default;
+
+  // Logs Event::Type INIT message.
+  virtual void WriteInitMessage(
+      const InternalAPMStreamsConfig& streams_config) = 0;
+
+  // Logs Event::Type STREAM message. To log an input/output pair,
+  // call the AddCapture* and AddAudioProcessingState methods followed
+  // by a WriteCaptureStreamMessage call.
+  virtual void AddCaptureStreamInput(const FloatAudioFrame& src) = 0;
+  virtual void AddCaptureStreamOutput(const FloatAudioFrame& src) = 0;
+  virtual void AddCaptureStreamInput(const AudioFrame& frame) = 0;
+  virtual void AddCaptureStreamOutput(const AudioFrame& frame) = 0;
+  virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
+  virtual void WriteCaptureStreamMessage() = 0;
+
+  // Logs Event::Type REVERSE_STREAM message.
+  virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0;
+  virtual void WriteRenderStreamMessage(const FloatAudioFrame& src) = 0;
+
+  // Logs Event::Type CONFIG message.
+  virtual void WriteConfig(const InternalAPMConfig& config) = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_