Update talk to 58127566 together with
https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/localaudiosource.cc b/talk/app/webrtc/localaudiosource.cc
index 2cd472a..3dc5c6c 100644
--- a/talk/app/webrtc/localaudiosource.cc
+++ b/talk/app/webrtc/localaudiosource.cc
@@ -54,8 +54,6 @@
const char MediaConstraintsInterface::kTypingNoiseDetection[] =
"googTypingNoiseDetection";
const char MediaConstraintsInterface::kAudioMirroring[] = "googAudioMirroring";
-// TODO(perkj): Remove kInternalAecDump once its not used by Chrome.
-const char MediaConstraintsInterface::kInternalAecDump[] = "deprecatedAecDump";
namespace {
@@ -129,8 +127,6 @@
return;
}
options_.SetAll(audio_options);
- if (options.enable_aec_dump)
- options_.aec_dump.Set(true);
source_state_ = kLive;
}
diff --git a/talk/app/webrtc/mediaconstraintsinterface.h b/talk/app/webrtc/mediaconstraintsinterface.h
index ba6b09b..5cf2184 100644
--- a/talk/app/webrtc/mediaconstraintsinterface.h
+++ b/talk/app/webrtc/mediaconstraintsinterface.h
@@ -117,13 +117,6 @@
// stripped by Chrome before passed down to Libjingle.
static const char kInternalConstraintPrefix[];
- // These constraints are for internal use only, representing Chrome command
- // line flags. So they are prefixed with "internal" so JS values will be
- // removed.
- // Used by a local audio source.
- // TODO(perkj): Remove once Chrome use PeerConnectionFactory::SetOptions.
- static const char kInternalAecDump[]; // internalAecDump
-
protected:
// Dtor protected as objects shouldn't be deleted via this interface
virtual ~MediaConstraintsInterface() {}
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index e8b8f63..ee15b5d 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -105,12 +105,21 @@
scoped_refptr<webrtc::VideoSourceInterface> source;
};
+struct StartAecDumpParams : public talk_base::MessageData {
+ explicit StartAecDumpParams(FILE* aec_dump_file)
+ : aec_dump_file(aec_dump_file) {
+ }
+ FILE* aec_dump_file;
+ bool result;
+};
+
enum {
MSG_INIT_FACTORY = 1,
MSG_TERMINATE_FACTORY,
MSG_CREATE_PEERCONNECTION,
MSG_CREATE_AUDIOSOURCE,
MSG_CREATE_VIDEOSOURCE,
+ MSG_START_AEC_DUMP,
};
} // namespace
@@ -223,6 +232,12 @@
pdata->source = CreateVideoSource_s(pdata->capturer, pdata->constraints);
break;
}
+ case MSG_START_AEC_DUMP: {
+ StartAecDumpParams* pdata =
+ static_cast<StartAecDumpParams*>(msg->pdata);
+ pdata->result = StartAecDump_s(pdata->aec_dump_file);
+ break;
+ }
}
}
@@ -274,6 +289,10 @@
return VideoSourceProxy::Create(signaling_thread_, source);
}
+bool PeerConnectionFactory::StartAecDump_s(FILE* file) {
+ return channel_manager_->StartAecDump(file);
+}
+
scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::IceServers& configuration,
@@ -361,6 +380,12 @@
return AudioTrackProxy::Create(signaling_thread_, track);
}
+bool PeerConnectionFactory::StartAecDump(FILE* file) {
+ StartAecDumpParams params(file);
+ signaling_thread_->Send(this, MSG_START_AEC_DUMP, ¶ms);
+ return params.result;
+}
+
cricket::ChannelManager* PeerConnectionFactory::channel_manager() {
return channel_manager_.get();
}
diff --git a/talk/app/webrtc/peerconnectionfactory.h b/talk/app/webrtc/peerconnectionfactory.h
index dff885d..63d37f0 100644
--- a/talk/app/webrtc/peerconnectionfactory.h
+++ b/talk/app/webrtc/peerconnectionfactory.h
@@ -78,6 +78,8 @@
CreateAudioTrack(const std::string& id,
AudioSourceInterface* audio_source);
+ virtual bool StartAecDump(FILE* file);
+
virtual cricket::ChannelManager* channel_manager();
virtual talk_base::Thread* signaling_thread();
virtual talk_base::Thread* worker_thread();
@@ -93,7 +95,6 @@
cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
virtual ~PeerConnectionFactory();
-
private:
bool Initialize_s();
void Terminate_s();
@@ -108,6 +109,8 @@
PortAllocatorFactoryInterface* allocator_factory,
DTLSIdentityServiceInterface* dtls_identity_service,
PeerConnectionObserver* observer);
+ bool StartAecDump_s(FILE* file);
+
// Implements talk_base::MessageHandler.
void OnMessage(talk_base::Message* msg);
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index a127dad..01f1e1c 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -393,11 +393,9 @@
class Options {
public:
Options() :
- enable_aec_dump(false),
disable_encryption(false),
disable_sctp_data_channels(false) {
}
- bool enable_aec_dump;
bool disable_encryption;
bool disable_sctp_data_channels;
};
@@ -442,6 +440,12 @@
CreateAudioTrack(const std::string& label,
AudioSourceInterface* source) = 0;
+ // Starts AEC dump using existing file. Takes ownership of |file| and passes
+ // it on to VoiceEngine (via other objects) immediately, which will take
+ // the ownerhip.
+ // TODO(grunell): Remove when Chromium has started to use AEC in each source.
+ virtual bool StartAecDump(FILE* file) = 0;
+
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.
