APM move: Add interfaces for Audio Processing in Audio Service

Adds some mojo interfaces for configuring audio processing, intended for
both WebRTC and system-level echo cancellation, though the latter will
be less configurable.

There are two parts to this change:
1. AudioProcessingSettings in media/audio, used to configure audio
   processing when done in the audio service. At one point, the similar
   AudioProcessingParameters in content will be replaced by this one.
2. AudioProcessorControls in media/webrtc; an interface used to control
   auxiliary audio processor functionality, specifically getting stats
   and starting/stopping AEC dumps.

For an outline of the project this CL is part of, see: https://docs.google.com/document/d/1u4POff_ts_1LE3WDLA_wDDFnUswdlsuHL5DsiTE0a3U/edit?usp=sharing
It's accessible to everyone @chromium.org.

Bug: 851959
Tbr: tommi@webrtc.org
Cq-Include-Trybots: luci.chromium.try:android_optional_gpu_tests_rel;luci.chromium.try:linux_optional_gpu_tests_rel;luci.chromium.try:mac_optional_gpu_tests_rel;luci.chromium.try:win_optional_gpu_tests_rel
Change-Id: I0c8c934b35b93aa728a0d1b60ae76063e925ee6c
Reviewed-on: https://chromium-review.googlesource.com/1167848
Commit-Queue: Tommi <tommi@chromium.org>
Reviewed-by: Tommi <tommi@chromium.org>
Reviewed-by: Dale Curtis <dalecurtis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#583239}

media/DEPS

@@ -25,6 +25,7 @@ include_rules = [
   "+third_party/widevine/cdm/widevine_cdm_common.h",
   "-ipc",
   "-media/blink",
+  "-media/webrtc",
 ]
 
 specific_include_rules = {

media/audio/BUILD.gn

@@ -123,6 +123,7 @@ source_set("audio") {
     "audio_output_stream_sink.h",
     "audio_power_monitor.cc",
     "audio_power_monitor.h",
+    "audio_processing.h",
     "audio_source_diverter.h",
     "audio_sync_reader.cc",
     "audio_sync_reader.h",

media/audio/audio_processing.h

+// Copyright 2018 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef MEDIA_AUDIO_AUDIO_PROCESSING_H_
+#define MEDIA_AUDIO_AUDIO_PROCESSING_H_
+
+#include "base/files/file.h"
+#include "base/time/time.h"
+#include "base/unguessable_token.h"
+#include "media/base/media_export.h"
+
+namespace media {
+
+enum class AutomaticGainControlType { kDisabled, kDefault, kExperimental };
+enum class EchoCancellationType { kDisabled, kAec2, kAec3, kSystemAec };
+enum class NoiseSuppressionType { kDisabled, kDefault, kExperimental };
+
+struct MEDIA_EXPORT AudioProcessingSettings {
+  EchoCancellationType echo_cancellation = EchoCancellationType::kDisabled;
+  NoiseSuppressionType noise_suppression = NoiseSuppressionType::kDisabled;
+  AutomaticGainControlType automatic_gain_control =
+      AutomaticGainControlType::kDisabled;
+  bool high_pass_filter = false;
+  bool typing_detection = false;
+  bool stereo_mirroring = false;
+
+  bool operator==(const AudioProcessingSettings& b) const {
+    return echo_cancellation == b.echo_cancellation &&
+           noise_suppression == b.noise_suppression &&
+           automatic_gain_control == b.automatic_gain_control &&
+           high_pass_filter == b.high_pass_filter &&
+           typing_detection == b.typing_detection &&
+           stereo_mirroring == b.stereo_mirroring;
+  }
+};
+
+}  // namespace media
+
+#endif  // MEDIA_AUDIO_AUDIO_PROCESSING_H_

media/webrtc/BUILD.gn

+# Copyright 2018 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+import("//media/media_options.gni")
+import("//third_party/webrtc/webrtc.gni")
+
+source_set("webrtc") {
+  sources = [
+    "audio_processor_controls.h",
+  ]
+  deps = [
+    "//base",
+    "//media",
+    "//third_party/webrtc/api:libjingle_peerconnection_api",
+    "//third_party/webrtc/modules/audio_processing",
+    "//third_party/webrtc/modules/audio_processing:audio_processing_statistics",
+    "//third_party/webrtc_overrides:init_webrtc",
+  ]
+}

media/webrtc/DEPS

+include_rules = [
+   "+third_party/webrtc/api",
+   "+third_party/webrtc/modules/audio_processing",
+]

media/webrtc/OWNERS

+olka@chromium.org
+maxmorin@chromium.org
+dalecurtis@chromium.org
+miu@chromium.org
+ossu@chromium.org
+
+# COMPONENT: Blink>WebRTC>Audio

media/webrtc/audio_processor_controls.h

+// Copyright 2018 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef MEDIA_WEBRTC_AUDIO_PROCESSOR_CONTROLS_H_
+#define MEDIA_WEBRTC_AUDIO_PROCESSOR_CONTROLS_H_
+
+#include "base/callback.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
+
+namespace media {
+
+class MEDIA_EXPORT AudioProcessorControls {
+ public:
+  using GetStatsCB = base::OnceCallback<void(
+      const webrtc::AudioProcessorInterface::AudioProcessorStatistics& stats)>;
+
+  // Request the latest stats from the audio processor. Stats are returned
+  // asynchronously through |callback|.
+  virtual void GetStats(GetStatsCB callback) = 0;
+
+  // Begin dumping echo cancellation data into |file|.
+  virtual void StartEchoCancellationDump(base::File file) = 0;
+
+  // Stop any ongoin dump of echo cancellation data.
+  virtual void StopEchoCancellationDump() = 0;
+
+ protected:
+  virtual ~AudioProcessorControls() = default;
+};
+
+}  // namespace media
+
+#endif  // MEDIA_WEBRTC_AUDIO_PROCESSOR_CONTROLS_H_