Added an "enable-audio-processor" flag and WebRtcAudioProcessor class.

This CL is a break-down CL from https://codereview.chromium.org/37793005. As the first step to move WebRtc APM to Chrome, it adds a enable-audio-processor command line flag, the WebRtcAudioProcessor class and its unittest.
WebRtcAudioProcessor is not hooked up the any of the code in Chrome yet, but it will be exercised and tested by its unittest.


TBR=tommi@chromium.org
BUG=264611
TEST=content_unittest

Review URL: https://codereview.chromium.org/88513002

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@237360 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 96 insertions, 0 deletions

diff --git a/content/renderer/media/media_stream_audio_processor_options.cc b/content/renderer/media/media_stream_audio_processor_options.cc
new file mode 100644
index 0000000..add7f957
--- /dev/null
+++ b/content/renderer/media/media_stream_audio_processor_options.cc
@@ -0,0 +1,96 @@
+// Copyright 2013 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.
+
+#include "content/renderer/media/media_stream_audio_processor_options.h"
+
+#include "base/files/file_path.h"
+#include "base/logging.h"
+#include "base/path_service.h"
+#include "base/strings/utf_string_conversions.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
+#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace content {
+
+bool GetPropertyFromConstraints(const MediaConstraintsInterface* constraints,
+                                const std::string& key) {
+  bool value = false;
+  return webrtc::FindConstraint(constraints, key, &value, NULL) && value;
+}
+
+void EnableEchoCancellation(AudioProcessing* audio_processing) {
+#if defined(OS_IOS)
+  // On iOS, VPIO provides built-in EC and AGC.
+  return;
+#elif defined(OS_ANDROID)
+  // Mobile devices are using AECM.
+  int err = audio_processing->echo_control_mobile()->Enable(true);
+  err |= audio_processing->echo_control_mobile()->set_routing_mode(
+      webrtc::EchoControlMobile::kSpeakerphone);
+  CHECK_EQ(err, 0);
+#else
+  int err = audio_processing->echo_cancellation()->Enable(true);
+  err |= audio_processing->echo_cancellation()->set_suppression_level(
+      webrtc::EchoCancellation::kHighSuppression);
+
+  // Enable the metrics for AEC.
+  err |= audio_processing->echo_cancellation()->enable_metrics(true);
+  err |= audio_processing->echo_cancellation()->enable_delay_logging(true);
+  CHECK_EQ(err, 0);
+#endif
+}
+
+void EnableNoiseSuppression(AudioProcessing* audio_processing) {
+  int err = audio_processing->noise_suppression()->set_level(
+      webrtc::NoiseSuppression::kHigh);
+  err |= audio_processing->noise_suppression()->Enable(true);
+  CHECK_EQ(err, 0);
+}
+
+void EnableHighPassFilter(AudioProcessing* audio_processing) {
+  CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0);
+}
+
+// TODO(xians): stereo swapping
+void EnableTypingDetection(AudioProcessing* audio_processing) {
+  int err = audio_processing->voice_detection()->Enable(true);
+  err |= audio_processing->voice_detection()->set_likelihood(
+      webrtc::VoiceDetection::kVeryLowLikelihood);
+  CHECK_EQ(err, 0);
+}
+
+void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) {
+  webrtc::Config config;
+  config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true));
+  audio_processing->SetExtraOptions(config);
+}
+
+void StartAecDump(AudioProcessing* audio_processing) {
+  // TODO(grunell): Figure out a more suitable directory for the audio dump
+  // data.
+  base::FilePath path;
+#if defined(CHROMEOS)
+  PathService::Get(base::DIR_TEMP, &path);
+#elif defined(ANDROID)
+  path = base::FilePath(FILE_PATH_LITERAL("sdcard"));
+#else
+  PathService::Get(base::DIR_EXE, &path);
+#endif
+  base::FilePath file = path.Append(FILE_PATH_LITERAL("audio.aecdump"));
+
+#if defined(OS_WIN)
+  const std::string file_name = WideToUTF8(file.value());
+#else
+  const std::string file_name = file.value();
+#endif
+  if (audio_processing->StartDebugRecording(file_name.c_str()))
+    DLOG(ERROR) << "Fail to start AEC debug recording";
+}
+
+void StopAecDump(AudioProcessing* audio_processing) {
+  if (audio_processing->StopDebugRecording())
+    DLOG(ERROR) << "Fail to stop AEC debug recording";
+}
+
+}  // namespace content