Set default audio constraints for mediastreams with audio.

content/renderer/media/media_stream_dependency_factory.cc

 // Copyright (c) 2012 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_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 36 matching lines @@
 #if defined(GOOGLE_TV)
 #include "content/renderer/media/rtc_video_decoder_factory_tv.h"
 #endif
 
 namespace content {
 
 // The constraint key for the PeerConnection constructor for enabling diagnostic
 // WebRTC logging. It's a Google specific key, hence the "goog" prefix.
 const char kWebRtcLoggingConstraint[] = "googLog";
 
-// Constant constraint keys which disables all audio constraints.
-// Only used in combination with WebAudio sources.
+// Constant constraint keys which enables default audio constraints on
+// mediastreams with audio.
 struct {
   const char* key;
   const char* value;
-} const kWebAudioConstraints[] = {
-    {webrtc::MediaConstraintsInterface::kEchoCancellation,
-     webrtc::MediaConstraintsInterface::kValueTrue},
-    {webrtc::MediaConstraintsInterface::kAutoGainControl,
-     webrtc::MediaConstraintsInterface::kValueTrue},
-    {webrtc::MediaConstraintsInterface::kNoiseSuppression,
-     webrtc::MediaConstraintsInterface::kValueTrue},
-    {webrtc::MediaConstraintsInterface::kHighpassFilter,
-     webrtc::MediaConstraintsInterface::kValueTrue},
+} const kDefaultAudioConstraints[] = {
+  { webrtc::MediaConstraintsInterface::kEchoCancellation,
+    webrtc::MediaConstraintsInterface::kValueTrue },
+  { webrtc::MediaConstraintsInterface::kAutoGainControl,
+    webrtc::MediaConstraintsInterface::kValueTrue },
+  { webrtc::MediaConstraintsInterface::kNoiseSuppression,
+    webrtc::MediaConstraintsInterface::kValueTrue },
+  { webrtc::MediaConstraintsInterface::kHighpassFilter,
+    webrtc::MediaConstraintsInterface::kValueTrue },
 };
 
-void ApplyFixedWebAudioConstraints(RTCMediaConstraints* constraints) {
-  for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kWebAudioConstraints); ++i) {
-    constraints->AddMandatory(kWebAudioConstraints[i].key,
-        kWebAudioConstraints[i].value, false);
+void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) {
+  const webrtc::MediaConstraintsInterface::Constraints& mandatory =
+      constraints->GetMandatory();
+  std::string string_value;
+  for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) {
+    if (!mandatory.FindFirst(kDefaultAudioConstraints[i].key, &string_value)) {
+      constraints->AddMandatory(kDefaultAudioConstraints[i].key,
+          kDefaultAudioConstraints[i].value, false);
+    } else {
+      DVLOG(1) << "Constraint " << kDefaultAudioConstraints[i].key
+               << " already set to " << string_value;
+    }
   }
 }
 
 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
  public:
   P2PPortAllocatorFactory(
       P2PSocketDispatcher* socket_dispatcher,
       talk_base::NetworkManager* network_manager,
       talk_base::PacketSocketFactory* socket_factory,
       WebKit::WebFrame* web_frame)
@@ skipping 194 matching lines @@
     source_data->SetVideoSource(
         CreateLocalVideoSource(source_data->device_info().session_id,
                                is_screencast,
                                &native_video_constraints).get());
     source_observer->AddSource(source_data->video_source());
   }
 
   // Do additional source initialization if the audio source is a valid
   // microphone or tab audio.
   RTCMediaConstraints native_audio_constraints(audio_constraints);
+  ApplyFixedAudioConstraints(&native_audio_constraints);
   WebKit::WebVector<WebKit::WebMediaStreamTrack> audio_tracks;
   web_stream->audioTracks(audio_tracks);
   const CommandLine& command_line = *CommandLine::ForCurrentProcess();
   if (command_line.HasSwitch(switches::kEnableWebRtcAecRecordings)) {
     native_audio_constraints.AddOptional(
         RTCMediaConstraints::kInternalAecDump, "true");
   }
   for (size_t i = 0; i < audio_tracks.size(); ++i) {
     const WebKit::WebMediaStreamSource& source = audio_tracks[i].source();
     MediaStreamSourceExtraData* source_data =
@@ skipping 314 matching lines @@
   scoped_refptr<WebAudioCapturerSource>
       webaudio_capturer_source(new WebAudioCapturerSource(capturer.get()));
   MediaStreamSourceExtraData* source_data =
       new content::MediaStreamSourceExtraData(webaudio_capturer_source.get());
 
   // Create a LocalAudioSource object which holds audio options.
   // Use audio constraints where all values are true, i.e., enable
   // echo cancellation, automatic gain control, noise suppression and
   // high-pass filter. SetLocalAudioSource() affects core audio parts in
   // third_party/Libjingle.
-  ApplyFixedWebAudioConstraints(constraints);
+  ApplyFixedAudioConstraints(constraints);
   source_data->SetLocalAudioSource(CreateLocalAudioSource(constraints).get());
   source->setExtraData(source_data);
 
   // Replace the default source with WebAudio as source instead.
   source->addAudioConsumer(webaudio_capturer_source.get());
 
   return capturer;
 }
 
 scoped_refptr<webrtc::VideoTrackInterface>
@@ skipping 190 matching lines @@
   }
 
   // Add the capturer to the WebRtcAudioDeviceImpl if it is a new capturer.
   if (is_new_capturer)
     GetWebRtcAudioDevice()->AddAudioCapturer(capturer);
 
   return capturer;
 }
 
 }  // namespace content