AecDump interface

webrtc/modules/audio_processing/include/aec_dump.h

+/*
+ *  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/base/array_view.h"
+
+namespace audioproc {
+class Event;
+}  // namespace audioproc
+
+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&) = delete;
+  InternalAPMConfig& operator=(const InternalAPMConfig&&) = delete;
+
+  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 AecDump {
+ public:
+  // A capture stream frame is logged before and after processing in
+  // the same protobuf message. To facilitate that, a CaptureStreamInfo
+  // instance is first filled with Input, then Output.
+  //
+  // To log an input/output pair, first call
+  // AecDump::GetCaptureStreamInfo.  Add the input and output to
+  // it. Then call AecDump::WriteCaptureStreamMessage.
+  class CaptureStreamInfo {
+   public:
+    virtual ~CaptureStreamInfo() = default;
+    virtual void AddInput(
+        const rtc::ArrayView<rtc::ArrayView<const float>>& src) = 0;
+    virtual void AddOutput(
+        const rtc::ArrayView<rtc::ArrayView<const float>>& src) = 0;
+
+    virtual void AddInput(const AudioFrame& frame) = 0;
+    virtual void AddOutput(const AudioFrame& frame) = 0;
+
+    virtual void set_delay(int delay) = 0;
+    virtual void set_drift(int drift) = 0;
+    virtual void set_level(int level) = 0;
+    virtual void set_keypress(bool keypress) = 0;
+  };
+
+  virtual ~AecDump() = default;
+
+  virtual std::unique_ptr<CaptureStreamInfo> GetCaptureStreamInfo() = 0;
+
+  // 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.
+  virtual void WriteInitMessage(
+      const InternalAPMStreamsConfig& streams_config) = 0;
+
+  virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0;
+
+  virtual void WriteRenderStreamMessage(
+      const rtc::ArrayView<rtc::ArrayView<const float>>& src) = 0;
+
+  virtual void WriteCaptureStreamMessage(
+      std::unique_ptr<CaptureStreamInfo> stream_info) = 0;
+
+  // If not |forced|, only writes the current config if it is
+  // different from the last saved one; if |forced|, writes the config
+  // regardless of the last saved.
+  virtual void WriteConfig(const InternalAPMConfig& config, bool forced) = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_