Do not provide the input level if the APM is not enabled in Chrome.

NOTRY=true
BUG=365616
TEST=content_unittests --gtest_filter=*GetSignalLevel*"

Review URL: https://codereview.chromium.org/247183002

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@265564 0039d316-1c4b-4281-b951-d872f2087c98

5 files changed, 34 insertions, 10 deletions

diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc
index aaa1138..9532eb3 100644
--- a/content/renderer/media/media_stream_audio_processor.cc
+++ b/content/renderer/media/media_stream_audio_processor.cc
@@ -155,6 +155,13 @@ class MediaStreamAudioProcessor::MediaStreamAudioConverter
   scoped_ptr<media::AudioFifo> fifo_;
 };
 
+bool MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() {
+  const std::string group_name =
+      base::FieldTrialList::FindFullName("MediaStreamAudioTrackProcessing");
+  return group_name == "Enabled" || CommandLine::ForCurrentProcess()->HasSwitch(
+      switches::kEnableAudioTrackProcessing);
+}
+
 MediaStreamAudioProcessor::MediaStreamAudioProcessor(
     const blink::WebMediaConstraints& constraints,
     int effects,
@@ -508,11 +515,4 @@ void MediaStreamAudioProcessor::StopAudioProcessing() {
   audio_processing_.reset();
 }
 
-bool MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() const {
-  const std::string group_name =
-      base::FieldTrialList::FindFullName("MediaStreamAudioTrackProcessing");
-  return group_name == "Enabled" || CommandLine::ForCurrentProcess()->HasSwitch(
-      switches::kEnableAudioTrackProcessing);
-}
-
 }  // namespace content
diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h
index 0c80514..73bf234 100644
--- a/content/renderer/media/media_stream_audio_processor.h
+++ b/content/renderer/media/media_stream_audio_processor.h
@@ -47,6 +47,10 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
     NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
     NON_EXPORTED_BASE(public AudioProcessorInterface) {
  public:
+  // Returns true if |kEnableAudioTrackProcessing| is on or if the
+  // |MediaStreamAudioTrackProcessing| finch experiment is enabled.
+  static bool IsAudioTrackProcessingEnabled();
+
   // |playout_data_source| is used to register this class as a sink to the
   // WebRtc playout data for processing AEC. If clients do not enable AEC,
   // |playout_data_source| won't be used.
@@ -85,8 +89,6 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
                              int* new_volume,
                              int16** out);
 
-  bool IsAudioTrackProcessingEnabled() const;
-
   // The audio format of the input to the processor.
   const media::AudioParameters& InputFormat() const;
 
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
index b1bde99..95958a8 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
@@ -86,6 +86,14 @@ void WebRtcLocalAudioTrackAdapter::RemoveSink(
 
 bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
   base::AutoLock auto_lock(lock_);
+  // It is required to provide the signal level after audio processing. In
+  // case the audio processing is not enabled for the track, we return
+  // false here in order not to overwrite the value from WebRTC.
+  // TODO(xians): Remove this after we turn on the APM in Chrome by default.
+  // http://crbug/365672 .
+  if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
+    return false;
+
   *level = signal_level_;
   return true;
 }
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
index 4fe1eaf..d798b31 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
@@ -2,6 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+#include "base/command_line.h"
+#include "content/public/common/content_switches.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "testing/gmock/include/gmock/gmock.h"
@@ -79,4 +81,16 @@ TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
 }
 
+TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
+  webrtc::AudioTrackInterface* webrtc_track =
+      static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
+  int signal_level = 0;
+  EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
+
+  // Enable the audio processing in the audio track.
+  CommandLine::ForCurrentProcess()->AppendSwitch(
+      switches::kEnableAudioTrackProcessing);
+  EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
+}
+
 }  // namespace content
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 5169c27..4b14af7 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -485,7 +485,7 @@ void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
     // Note that, we turn off the audio processing in PeerConnection if the
     // processor has already processed the data.
     need_audio_processing = need_audio_processing_ ?
-        !audio_processor_->IsAudioTrackProcessingEnabled() : false;
+        !MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() : false;
   }
 
   DCHECK(audio_processor_->InputFormat().IsValid());