Remove old-style constraints from Chrome internals

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 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/webrtc/peer_connection_dependency_factory.h"
 
 #include <stddef.h>
 
 #include <utility>
 #include <vector>
@@ skipping 14 matching lines @@
 #include "content/public/common/content_switches.h"
 #include "content/public/common/feature_h264_with_openh264_ffmpeg.h"
 #include "content/public/common/features.h"
 #include "content/public/common/renderer_preferences.h"
 #include "content/public/common/webrtc_ip_handling_policy.h"
 #include "content/public/renderer/content_renderer_client.h"
 #include "content/renderer/media/media_stream.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/media_stream_audio_source.h"
+#include "content/renderer/media/media_stream_constraints_util.h"
 #include "content/renderer/media/media_stream_video_source.h"
 #include "content/renderer/media/media_stream_video_track.h"
 #include "content/renderer/media/peer_connection_identity_store.h"
-#include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 #include "content/renderer/media/rtc_video_decoder_factory.h"
 #include "content/renderer/media/rtc_video_encoder_factory.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc/media_stream_remote_audio_track.h"
 #include "content/renderer/media/webrtc/stun_field_trial.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
@@ skipping 14 matching lines @@
 #include "media/renderers/gpu_video_accelerator_factories.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebURL.h"
 #include "third_party/WebKit/public/web/WebDocument.h"
 #include "third_party/WebKit/public/web/WebFrame.h"
 #include "third_party/webrtc/api/mediaconstraintsinterface.h"
 #include "third_party/webrtc/base/ssladapter.h"
+#include "third_party/webrtc/media/base/mediachannel.h"
 #include "third_party/webrtc/modules/video_coding/codecs/h264/include/h264.h"
 
 #if defined(OS_ANDROID)
 #include "media/base/android/media_codec_util.h"
 #endif
 
 namespace content {
 
 namespace {
 
@@ skipping 10 matching lines @@
     return DEFAULT_PUBLIC_AND_PRIVATE_INTERFACES;
   if (preference == kWebRTCIPHandlingDefaultPublicInterfaceOnly)
     return DEFAULT_PUBLIC_INTERFACE_ONLY;
   if (preference == kWebRTCIPHandlingDisableNonProxiedUdp)
     return DISABLE_NON_PROXIED_UDP;
   return DEFAULT;
 }
 
 }  // namespace
 
-// Map of corresponding media constraints and platform effects.
-struct {
-  const char* constraint;
-  const media::AudioParameters::PlatformEffectsMask effect;
-} const kConstraintEffectMap[] = {
-  { webrtc::MediaConstraintsInterface::kGoogEchoCancellation,
-    media::AudioParameters::ECHO_CANCELLER },
-};
-
-// If any platform effects are available, check them against the constraints.
-// Disable effects to match false constraints, but if a constraint is true, set
-// the constraint to false to later disable the software effect.
-//
-// This function may modify both |constraints| and |effects|.
-void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
-                                    int* effects) {
-  if (*effects != media::AudioParameters::NO_EFFECTS) {
-    for (size_t i = 0; i < arraysize(kConstraintEffectMap); ++i) {
-      bool value;
-      size_t is_mandatory = 0;
-      if (!webrtc::FindConstraint(constraints,
-                                  kConstraintEffectMap[i].constraint,
-                                  &value,
-                                  &is_mandatory) || !value) {
-        // If the constraint is false, or does not exist, disable the platform
-        // effect.
-        *effects &= ~kConstraintEffectMap[i].effect;
-        DVLOG(1) << "Disabling platform effect: "
-                 << kConstraintEffectMap[i].effect;
-      } else if (*effects & kConstraintEffectMap[i].effect) {
-        // If the constraint is true, leave the platform effect enabled, and
-        // set the constraint to false to later disable the software effect.
-        if (is_mandatory) {
-          constraints->AddMandatory(kConstraintEffectMap[i].constraint,
-              webrtc::MediaConstraintsInterface::kValueFalse, true);
-        } else {
-          constraints->AddOptional(kConstraintEffectMap[i].constraint,
-              webrtc::MediaConstraintsInterface::kValueFalse, true);
-        }
-        DVLOG(1) << "Disabling constraint: "
-                 << kConstraintEffectMap[i].constraint;
-      }
-    }
-  }
-}
-
 PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
     P2PSocketDispatcher* p2p_socket_dispatcher)
     : network_manager_(NULL),
       p2p_socket_dispatcher_(p2p_socket_dispatcher),
       signaling_thread_(NULL),
       worker_thread_(NULL),
       chrome_signaling_thread_("Chrome_libJingle_Signaling"),
       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
   TryScheduleStunProbeTrial();
 }
@@ skipping 15 matching lines @@
 }
 
 bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
     int render_frame_id,
     const blink::WebMediaConstraints& audio_constraints,
     MediaStreamAudioSource* source_data) {
   DVLOG(1) << "InitializeMediaStreamAudioSources()";
 
   // Do additional source initialization if the audio source is a valid
   // microphone or tab audio.
-  RTCMediaConstraints native_audio_constraints(audio_constraints);
-  MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints);
 
   StreamDeviceInfo device_info = source_data->device_info();
-  RTCMediaConstraints constraints = native_audio_constraints;
-  // May modify both |constraints| and |effects|.
-  HarmonizeConstraintsAndEffects(&constraints,
-                                 &device_info.device.input.effects);
+
+  cricket::AudioOptions options;
+  // Apply relevant constraints.
+  options.echo_cancellation = ConstraintToOptional(
+      audio_constraints, &blink::WebMediaTrackConstraintSet::echoCancellation);
+  options.delay_agnostic_aec = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googDAEchoCancellation);
+  options.auto_gain_control = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googAutoGainControl);
+  options.experimental_agc = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googExperimentalAutoGainControl);
+  options.noise_suppression = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googNoiseSuppression);
+  options.experimental_ns = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googExperimentalNoiseSuppression);
+  options.highpass_filter = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googHighpassFilter);
+  options.typing_detection = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googTypingNoiseDetection);
+  options.stereo_swapping = ConstraintToOptional(
+      audio_constraints,
+      &blink::WebMediaTrackConstraintSet::googAudioMirroring);
+
+  MediaAudioConstraints::ApplyFixedAudioConstraints(&options);
+
+  if (device_info.device.input.effects &
+      media::AudioParameters::ECHO_CANCELLER) {
+    // TODO(hta): Figure out if we should be looking at echoCancellation.
+    // Previous code had googEchoCancellation only.
+    const blink::BooleanConstraint& echoCancellation =
+        audio_constraints.basic().googEchoCancellation;
+    if (echoCancellation.hasExact() && !echoCancellation.exact()) {
+      device_info.device.input.effects &=
+          ~media::AudioParameters::ECHO_CANCELLER;
+    }
+    options.echo_cancellation = rtc::Optional<bool>(false);
+  }
 
   scoped_ptr<WebRtcAudioCapturer> capturer = CreateAudioCapturer(
       render_frame_id, device_info, audio_constraints, source_data);
   if (!capturer.get()) {
     const std::string log_string =
         "PCDF::InitializeMediaStreamAudioSource: fails to create capturer";
     WebRtcLogMessage(log_string);
     DVLOG(1) << log_string;
     // TODO(xians): Don't we need to check if source_observer is observing
     // something? If not, then it looks like we have a leak here.
     // OTOH, if it _is_ observing something, then the callback might
     // be called multiple times which is likely also a bug.
     return false;
   }
   source_data->SetAudioCapturer(std::move(capturer));
 
   // Creates a LocalAudioSource object which holds audio options.
   // TODO(xians): The option should apply to the track instead of the source.
   // TODO(perkj): Move audio constraints parsing to Chrome.
   // Currently there are a few constraints that are parsed by libjingle and
   // the state is set to ended if parsing fails.
   scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
-      CreateLocalAudioSource(&constraints).get());
+      CreateLocalAudioSource(options).get());
   if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
     return false;
   }
   source_data->SetLocalAudioSource(rtc_source.get());
   return true;
 }
 
 WebRtcVideoCapturerAdapter*
 PeerConnectionDependencyFactory::CreateVideoCapturer(
     bool is_screeencast) {
   // We need to make sure the libjingle thread wrappers have been created
   // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
   // since the base class of WebRtcVideoCapturerAdapter is a
   // cricket::VideoCapturer and it uses the libjingle thread wrappers.
   if (!GetPcFactory().get())
     return NULL;
   return new WebRtcVideoCapturerAdapter(is_screeencast);
 }
 
 scoped_refptr<webrtc::VideoTrackSourceInterface>
 PeerConnectionDependencyFactory::CreateVideoSource(
-    cricket::VideoCapturer* capturer,
-    const blink::WebMediaConstraints& constraints) {
-  RTCMediaConstraints webrtc_constraints(constraints);
+    cricket::VideoCapturer* capturer) {
   scoped_refptr<webrtc::VideoTrackSourceInterface> source =
-      GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
+      GetPcFactory()->CreateVideoSource(capturer).get();
   return source;
 }
 
 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
 PeerConnectionDependencyFactory::GetPcFactory() {
   if (!pc_factory_.get())
     CreatePeerConnectionFactory();
   CHECK(pc_factory_.get());
   return pc_factory_;
 }
@@ skipping 131 matching lines @@
   event->Signal();
 }
 
 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
   return pc_factory_.get() != NULL;
 }
 
 scoped_refptr<webrtc::PeerConnectionInterface>
 PeerConnectionDependencyFactory::CreatePeerConnection(
     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
-    const webrtc::MediaConstraintsInterface* constraints,
     blink::WebFrame* web_frame,
     webrtc::PeerConnectionObserver* observer) {
   CHECK(web_frame);
   CHECK(observer);
   if (!GetPcFactory().get())
     return NULL;
 
   rtc::scoped_ptr<PeerConnectionIdentityStore> identity_store(
       new PeerConnectionIdentityStore(
           base::ThreadTaskRunnerHandle::Get(),
@@ skipping 102 matching lines @@
     }
     network_manager.reset(filtering_network_manager);
   } else {
     network_manager.reset(new EmptyNetworkManager(network_manager_));
   }
   rtc::scoped_ptr<P2PPortAllocator> port_allocator(new P2PPortAllocator(
       p2p_socket_dispatcher_, std::move(network_manager), socket_factory_.get(),
       port_config, requesting_origin, chrome_worker_thread_.task_runner()));
 
   return GetPcFactory()
-      ->CreatePeerConnection(config, constraints, std::move(port_allocator),
+      ->CreatePeerConnection(config, std::move(port_allocator),
                              std::move(identity_store), observer)
       .get();
 }
 
 scoped_refptr<webrtc::MediaStreamInterface>
 PeerConnectionDependencyFactory::CreateLocalMediaStream(
     const std::string& label) {
   return GetPcFactory()->CreateLocalMediaStream(label).get();
 }
 
 scoped_refptr<webrtc::AudioSourceInterface>
 PeerConnectionDependencyFactory::CreateLocalAudioSource(
-    const webrtc::MediaConstraintsInterface* constraints) {
+    const cricket::AudioOptions& options) {
   scoped_refptr<webrtc::AudioSourceInterface> source =
-      GetPcFactory()->CreateAudioSource(constraints).get();
+      GetPcFactory()->CreateAudioSource(options).get();
   return source;
 }
 
 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
     const blink::WebMediaStreamTrack& track) {
   blink::WebMediaStreamSource source = track.source();
   DCHECK_EQ(source.getType(), blink::WebMediaStreamSource::TypeAudio);
   MediaStreamAudioSource* source_data = MediaStreamAudioSource::From(source);
 
   if (!source_data) {
@@ skipping 56 matching lines @@
 void PeerConnectionDependencyFactory::CreateWebAudioSource(
     blink::WebMediaStreamSource* source) {
   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
 
   MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
   source_data->SetWebAudioCapturer(
       make_scoped_ptr(new WebAudioCapturerSource(source)));
 
   // Create a LocalAudioSource object which holds audio options.
   // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
-  source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
+  cricket::AudioOptions options;
+  source_data->SetLocalAudioSource(CreateLocalAudioSource(options).get());
   source->setExtraData(source_data);
 }
 
 scoped_refptr<webrtc::VideoTrackInterface>
 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
     const std::string& id,
     webrtc::VideoTrackSourceInterface* source) {
   return GetPcFactory()->CreateVideoTrack(id, source).get();
 }
 
@@ skipping 166 matching lines @@
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
   if (audio_device_.get())
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content