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Changes based on the review comments.

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* Rename WebRTCSession to WebRtcSession.
* Add comments to the signal.
Review URL: http://webrtc-codereview.appspot.com/114009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@402 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org 2011-08-19 00:14:23 +00:00
parent bfc63ae83f
commit 765c918677
8 changed files with 107 additions and 155 deletions
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6
third_party_mods/libjingle/source/talk/app/webrtc/peerconnection_impl.cc
View File

@ -285,10 +285,10 @@ bool PeerConnectionImpl::SignalingMessage_s(const std::string& msg) {
return ret;
}
WebRTCSession* PeerConnectionImpl::CreateMediaSession(
WebRtcSession* PeerConnectionImpl::CreateMediaSession(
const std::string& id, const std::string& dir) {
ASSERT(port_allocator_ != NULL);
WebRTCSession* session = new WebRTCSession(id, dir,
WebRtcSession* session = new WebRtcSession(id, dir,
port_allocator_, channel_manager_.get(),
signaling_thread_.get());
@ -318,7 +318,7 @@ WebRTCSession* PeerConnectionImpl::CreateMediaSession(
return session;
}
void PeerConnectionImpl::SendRemoveSignal(WebRTCSession* session) {
void PeerConnectionImpl::SendRemoveSignal(WebRtcSession* session) {
if (event_callback_) {
std::string message;
if (GetJSONSignalingMessage(session->remote_description(),

8
third_party_mods/libjingle/source/talk/app/webrtc/peerconnection_impl.h
View File

@ -53,7 +53,7 @@ namespace webrtc {
class AudioDeviceModule;
class ExternalRenderer;
class WebRTCSession;
class WebRtcSession;
class PeerConnectionImpl : public PeerConnection,
public talk_base::MessageHandler,
@ -114,8 +114,8 @@ class PeerConnectionImpl : public PeerConnection,
bool ParseConfigString(const std::string& config,
talk_base::SocketAddress* stun_addr);
void WrapChromiumThread();
void SendRemoveSignal(WebRTCSession* session);
WebRTCSession* CreateMediaSession(const std::string& id,
void SendRemoveSignal(WebRtcSession* session);
WebRtcSession* CreateMediaSession(const std::string& id,
const std::string& dir);
virtual void OnMessage(talk_base::Message* message);
@ -161,7 +161,7 @@ class PeerConnectionImpl : public PeerConnection,
ServiceType service_type_;
PeerConnectionObserver* event_callback_;
talk_base::scoped_ptr<WebRTCSession> session_;
talk_base::scoped_ptr<WebRtcSession> session_;
// TODO(ronghua): There's no such concept as "incoming" and "outgoing"
// according to the spec. This will be removed in the new PeerConnection.
bool incoming_;

8
third_party_mods/libjingle/source/talk/app/webrtc/peerconnection_impl_dev.cc
View File

@ -36,12 +36,12 @@ bool PeerConnectionImpl::Init() {
return true;
}
WebRTCSession* PeerConnectionImpl::CreateSession() {
// TODO(ronghuawu): when we have the new WebRTCSession we don't need these
WebRtcSession* PeerConnectionImpl::CreateSession() {
// TODO(ronghuawu): when we have the new WebRtcSession we don't need these
std::string id = "";
std::string direction = "";
WebRTCSession* session =
new WebRTCSession(id, direction, port_allocator_,
WebRtcSession* session =
new WebRtcSession(id, direction, port_allocator_,
channel_manager_,
// TODO(ronghuawu): implement PeerConnectionImplCallbacks
// this,

6
third_party_mods/libjingle/source/talk/app/webrtc/peerconnection_impl_dev.h
View File

@ -42,7 +42,7 @@ class Message;
namespace webrtc {
class WebRTCSession;
class WebRtcSession;
class PeerConnectionImpl : public PeerConnection {
public:
@ -100,7 +100,7 @@ class PeerConnectionImpl : public PeerConnection {
}
private:
WebRTCSession* CreateSession();
WebRtcSession* CreateSession();
virtual void OnMessage(talk_base::Message* msg);
void AddStream_s(LocalStream* stream);
void RemoveStream_s(LocalStream* stream);
@ -110,7 +110,7 @@ private:
bool initialized_;
ReadyState ready_state_;
PeerConnectionObserver* observer_;
talk_base::scoped_ptr<WebRTCSession> session_;
talk_base::scoped_ptr<WebRtcSession> session_;
talk_base::scoped_ptr<talk_base::Thread> signaling_thread_;
cricket::ChannelManager* channel_manager_;
cricket::PortAllocator* port_allocator_;

2
third_party_mods/libjingle/source/talk/app/webrtc/peerconnectiontransport.h
View File

@ -53,7 +53,7 @@ class PeerConnectionTransport : public talk_base::MessageHandler,
void ConnectChannels();
// methods to handle transport channels. These methods are relayed from
// WebRTCSessionImpl which implements cricket::BaseSession methods
// WebRtcSession which implements cricket::BaseSession methods
cricket::TransportChannel* CreateChannel(const std::string& channel_name,
const std::string& content_type);
cricket::TransportChannel* GetChannel(const std::string& channel_name,

116
third_party_mods/libjingle/source/talk/app/webrtc/webrtcsession.cc
View File

@ -64,10 +64,10 @@ typedef std::vector<StreamInfo*> StreamMap; // not really a map (vector)
static const char kVideoStream[] = "video_rtp";
static const char kAudioStream[] = "rtp";
const char WebRTCSession::kOutgoingDirection[] = "s";
const char WebRTCSession::kIncomingDirection[] = "r";
const char WebRtcSession::kOutgoingDirection[] = "s";
const char WebRtcSession::kIncomingDirection[] = "r";
WebRTCSession::WebRTCSession(
WebRtcSession::WebRtcSession(
const std::string& id,
const std::string& direction,
cricket::PortAllocator* allocator,
@ -76,7 +76,7 @@ WebRTCSession::WebRTCSession(
: BaseSession(signaling_thread),
transport_(NULL),
channel_manager_(channelmgr),
all_transports_writable_(false),
transports_writable_(false),
muted_(false),
camera_muted_(false),
setup_timeout_(kCallSetupTimeout),
@ -87,7 +87,7 @@ WebRTCSession::WebRTCSession(
BaseSession::sid_ = id;
}
WebRTCSession::~WebRTCSession() {
WebRtcSession::~WebRtcSession() {
RemoveAllStreams();
if (state_ != STATE_RECEIVEDTERMINATE) {
Terminate();
@ -95,7 +95,7 @@ WebRTCSession::~WebRTCSession() {
signaling_thread_->Send(this, MSG_WEBRTC_DELETE_TRANSPORT, NULL);
}
bool WebRTCSession::Initiate() {
bool WebRtcSession::Initiate() {
signaling_thread_->Send(this, MSG_WEBRTC_CREATE_TRANSPORT, NULL);
if (transport_ == NULL) {
return false;
@ -104,24 +104,24 @@ bool WebRTCSession::Initiate() {
// start transports
transport_->SignalRequestSignaling.connect(
this, &WebRTCSession::OnRequestSignaling);
this, &WebRtcSession::OnRequestSignaling);
transport_->SignalCandidatesReady.connect(
this, &WebRTCSession::OnCandidatesReady);
this, &WebRtcSession::OnCandidatesReady);
transport_->SignalWritableState.connect(
this, &WebRTCSession::OnWritableState);
this, &WebRtcSession::OnWritableState);
// Limit the amount of time that setting up a call may take.
StartTransportTimeout(kCallSetupTimeout);
return true;
}
cricket::Transport* WebRTCSession::CreateTransport() {
cricket::Transport* WebRtcSession::CreateTransport() {
ASSERT(signaling_thread()->IsCurrent());
return new cricket::P2PTransport(
talk_base::Thread::Current(),
channel_manager_->worker_thread(), port_allocator());
}
bool WebRTCSession::CreateVoiceChannel(const std::string& stream_id) {
bool WebRtcSession::CreateVoiceChannel(const std::string& stream_id) {
StreamInfo* stream_info = new StreamInfo(stream_id);
stream_info->video = false;
streams_.push_back(stream_info);
@ -138,7 +138,7 @@ bool WebRTCSession::CreateVoiceChannel(const std::string& stream_id) {
return true;
}
bool WebRTCSession::CreateVideoChannel(const std::string& stream_id) {
bool WebRtcSession::CreateVideoChannel(const std::string& stream_id) {
StreamInfo* stream_info = new StreamInfo(stream_id);
stream_info->video = true;
streams_.push_back(stream_info);
@ -155,7 +155,7 @@ bool WebRTCSession::CreateVideoChannel(const std::string& stream_id) {
return true;
}
cricket::TransportChannel* WebRTCSession::CreateChannel(
cricket::TransportChannel* WebRtcSession::CreateChannel(
const std::string& content_name,
const std::string& name) {
if (!transport_) {
@ -184,7 +184,7 @@ cricket::TransportChannel* WebRTCSession::CreateChannel(
return transport_channel;
}
cricket::TransportChannel* WebRTCSession::GetChannel(
cricket::TransportChannel* WebRtcSession::GetChannel(
const std::string& content_name, const std::string& name) {
if (!transport_)
return NULL;
@ -198,7 +198,7 @@ cricket::TransportChannel* WebRTCSession::GetChannel(
return NULL;
}
void WebRTCSession::DestroyChannel(
void WebRtcSession::DestroyChannel(
const std::string& content_name, const std::string& name) {
if (!transport_)
return;
@ -214,7 +214,7 @@ void WebRTCSession::DestroyChannel(
}
}
void WebRTCSession::OnMessage(talk_base::Message* message) {
void WebRtcSession::OnMessage(talk_base::Message* message) {
switch (message->message_id) {
case MSG_CANDIDATE_TIMEOUT:
if (transport_->writable()) {
@ -239,7 +239,7 @@ void WebRTCSession::OnMessage(talk_base::Message* message) {
}
}
bool WebRTCSession::Connect() {
bool WebRtcSession::Connect() {
if (streams_.empty()) {
// nothing to initiate
return false;
@ -259,7 +259,7 @@ bool WebRTCSession::Connect() {
return true;
}
bool WebRTCSession::SetVideoRenderer(const std::string& stream_id,
bool WebRtcSession::SetVideoRenderer(const std::string& stream_id,
cricket::VideoRenderer* renderer) {
bool ret = false;
StreamMap::iterator iter;
@ -277,12 +277,12 @@ bool WebRTCSession::SetVideoRenderer(const std::string& stream_id,
return ret;
}
bool WebRTCSession::SetVideoCapture(bool capture) {
bool WebRtcSession::SetVideoCapture(bool capture) {
channel_manager_->SetVideoCapture(capture);
return true;
}
bool WebRTCSession::RemoveStream(const std::string& stream_id) {
bool WebRtcSession::RemoveStream(const std::string& stream_id) {
bool ret = false;
StreamMap::iterator iter;
for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
@ -313,7 +313,7 @@ bool WebRTCSession::RemoveStream(const std::string& stream_id) {
return ret;
}
void WebRTCSession::DisableLocalCandidate(const std::string& name) {
void WebRtcSession::DisableLocalCandidate(const std::string& name) {
for (size_t i = 0; i < local_candidates_.size(); ++i) {
if (local_candidates_[i].name().compare(name) == 0) {
talk_base::SocketAddress address(local_candidates_[i].address().ip(), 0);
@ -322,7 +322,7 @@ void WebRTCSession::DisableLocalCandidate(const std::string& name) {
}
}
void WebRTCSession::EnableAllStreams() {
void WebRtcSession::EnableAllStreams() {
StreamMap::const_iterator i;
for (i = streams_.begin(); i != streams_.end(); ++i) {
cricket::BaseChannel* channel = (*i)->channel;
@ -331,7 +331,7 @@ void WebRTCSession::EnableAllStreams() {
}
}
void WebRTCSession::RemoveAllStreams() {
void WebRtcSession::RemoveAllStreams() {
// signaling_thread_->Post(this, MSG_RTC_REMOVEALLSTREAMS);
// First build a list of streams to remove and then remove them.
// The reason we do this is that if we remove the streams inside the
@ -353,7 +353,7 @@ void WebRTCSession::RemoveAllStreams() {
SignalRemoveStreamMessage(this);
}
bool WebRTCSession::HasStream(const std::string& stream_id) const {
bool WebRtcSession::HasStream(const std::string& stream_id) const {
StreamMap::const_iterator iter;
for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
StreamInfo* sinfo = (*iter);
@ -364,7 +364,7 @@ bool WebRTCSession::HasStream(const std::string& stream_id) const {
return false;
}
bool WebRTCSession::HasStream(bool video) const {
bool WebRtcSession::HasStream(bool video) const {
StreamMap::const_iterator iter;
for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
StreamInfo* sinfo = (*iter);
@ -375,27 +375,27 @@ bool WebRTCSession::HasStream(bool video) const {
return false;
}
bool WebRTCSession::HasAudioStream() const {
bool WebRtcSession::HasAudioStream() const {
return HasStream(false);
}
bool WebRTCSession::HasVideoStream() const {
bool WebRtcSession::HasVideoStream() const {
return HasStream(true);
}
talk_base::Thread* WebRTCSession::worker_thread() {
talk_base::Thread* WebRtcSession::worker_thread() {
return channel_manager_->worker_thread();
}
void WebRTCSession::OnRequestSignaling(cricket::Transport* transport) {
void WebRtcSession::OnRequestSignaling(cricket::Transport* transport) {
transport->OnSignalingReady();
}
void WebRTCSession::OnWritableState(cricket::Transport* transport) {
void WebRtcSession::OnWritableState(cricket::Transport* transport) {
ASSERT(transport == transport_);
const bool all_transports_writable = transport_->writable();
if (all_transports_writable) {
if (all_transports_writable != all_transports_writable_) {
const bool transports_writable = transport_->writable();
if (transports_writable) {
if (transports_writable != transports_writable_) {
signaling_thread_->Clear(this, MSG_CANDIDATE_TIMEOUT);
} else {
// At one point all channels were writable and we had full connectivity,
@ -403,22 +403,22 @@ void WebRTCSession::OnWritableState(cricket::Transport* transport) {
// doesn't come back.
StartTransportTimeout(kCallLostTimeout);
}
all_transports_writable_ = all_transports_writable;
transports_writable_ = transports_writable;
}
NotifyTransportState();
return;
}
void WebRTCSession::StartTransportTimeout(int timeout) {
void WebRtcSession::StartTransportTimeout(int timeout) {
talk_base::Thread::Current()->PostDelayed(timeout, this,
MSG_CANDIDATE_TIMEOUT,
NULL);
}
void WebRTCSession::NotifyTransportState() {
void WebRtcSession::NotifyTransportState() {
}
bool WebRTCSession::OnInitiateMessage(
bool WebRtcSession::OnInitiateMessage(
cricket::SessionDescription* offer,
const std::vector<cricket::Candidate>& candidates) {
if (!offer) {
@ -480,7 +480,7 @@ bool WebRTCSession::OnInitiateMessage(
return true;
}
bool WebRTCSession::OnRemoteDescription(
bool WebRtcSession::OnRemoteDescription(
cricket::SessionDescription* desc,
const std::vector<cricket::Candidate>& candidates) {
@ -519,7 +519,7 @@ bool WebRTCSession::OnRemoteDescription(
return true;
}
void WebRTCSession::ProcessTerminateAccept(cricket::SessionDescription* desc) {
void WebRtcSession::ProcessTerminateAccept(cricket::SessionDescription* desc) {
const cricket::ContentInfo* video_content = GetFirstVideoContent(desc);
if (video_content) {
SignalRemoveStream(video_content->name, true);
@ -531,7 +531,7 @@ void WebRTCSession::ProcessTerminateAccept(cricket::SessionDescription* desc) {
}
}
bool WebRTCSession::CheckForStreamDeleteMessage(
bool WebRtcSession::CheckForStreamDeleteMessage(
const std::vector<cricket::Candidate>& candidates) {
for (size_t i = 0; i < candidates.size(); ++i) {
if (candidates[i].address().port() == 0) {
@ -541,7 +541,7 @@ bool WebRTCSession::CheckForStreamDeleteMessage(
return false;
}
bool WebRTCSession::OnStreamDeleteMessage(
bool WebRtcSession::OnStreamDeleteMessage(
const cricket::SessionDescription* desc,
const std::vector<cricket::Candidate>& candidates) {
// This will be called when session is in connected state
@ -570,7 +570,7 @@ bool WebRTCSession::OnStreamDeleteMessage(
return true;
}
void WebRTCSession::RemoveStreamOnRequest(
void WebRtcSession::RemoveStreamOnRequest(
const cricket::Candidate& candidate) {
// 1. Get Transport corresponding to candidate name
// 2. Get StreamInfo for the transport found in step 1
@ -606,7 +606,7 @@ void WebRTCSession::RemoveStreamOnRequest(
}
}
cricket::SessionDescription* WebRTCSession::CreateOffer() {
cricket::SessionDescription* WebRtcSession::CreateOffer() {
cricket::SessionDescription* offer = new cricket::SessionDescription();
StreamMap::iterator iter;
for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
@ -641,7 +641,7 @@ cricket::SessionDescription* WebRTCSession::CreateOffer() {
return offer;
}
cricket::SessionDescription* WebRTCSession::CreateAnswer(
cricket::SessionDescription* WebRtcSession::CreateAnswer(
const cricket::SessionDescription* offer) {
cricket::SessionDescription* answer = new cricket::SessionDescription();
@ -699,35 +699,11 @@ cricket::SessionDescription* WebRTCSession::CreateAnswer(
return answer;
}
void WebRTCSession::OnMute(bool mute) {
StreamMap::iterator iter;
for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
if (!(*iter)->video) {
cricket::VoiceChannel* voice_channel =
static_cast<cricket::VoiceChannel*>((*iter)->channel);
ASSERT(voice_channel != NULL);
voice_channel->Mute(mute);
}
}
}
void WebRTCSession::OnCameraMute(bool mute) {
StreamMap::iterator iter;
for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
if ((*iter)->video) {
cricket::VideoChannel* video_channel =
static_cast<cricket::VideoChannel*>((*iter)->channel);
ASSERT(video_channel != NULL);
video_channel->Mute(mute);
}
}
}
void WebRTCSession::SetError(Error error) {
void WebRtcSession::SetError(Error error) {
BaseSession::SetError(error);
}
void WebRTCSession::OnCandidatesReady(
void WebRtcSession::OnCandidatesReady(
cricket::Transport* transport,
const std::vector<cricket::Candidate>& candidates) {
std::vector<cricket::Candidate>::const_iterator iter;

33
third_party_mods/libjingle/source/talk/app/webrtc/webrtcsession.h
View File

@ -74,15 +74,15 @@ struct StreamInfo {
typedef std::vector<cricket::AudioCodec> AudioCodecs;
typedef std::vector<cricket::VideoCodec> VideoCodecs;
class WebRTCSession : public cricket::BaseSession {
class WebRtcSession : public cricket::BaseSession {
public:
WebRTCSession(const std::string& id,
WebRtcSession(const std::string& id,
const std::string& direction,
cricket::PortAllocator* allocator,
cricket::ChannelManager* channelmgr,
talk_base::Thread* signaling_thread);
~WebRTCSession();
~WebRtcSession();
bool Initiate();
bool Connect();
@ -90,8 +90,6 @@ class WebRTCSession : public cricket::BaseSession {
const std::vector<cricket::Candidate>& candidates);
bool OnInitiateMessage(cricket::SessionDescription* sdp,
const std::vector<cricket::Candidate>& candidates);
void OnMute(bool mute);
void OnCameraMute(bool mute);
bool CreateVoiceChannel(const std::string& stream_id);
bool CreateVideoChannel(const std::string& stream_id);
bool RemoveStream(const std::string& stream_id);
@ -110,14 +108,31 @@ class WebRTCSession : public cricket::BaseSession {
bool SetVideoRenderer(const std::string& stream_id,
cricket::VideoRenderer* renderer);
sigslot::signal1<WebRTCSession*> SignalRemoveStreamMessage;
// This signal occurs when all the streams have been removed.
// It is triggered by a successful call to the RemoveAllStream or
// the OnRemoteDescription with stream deleted signaling message with the
// candidates port equal to 0.
sigslot::signal1<WebRtcSession*> SignalRemoveStreamMessage;
// This signal indicates a stream has been added properly.
// It is triggered by a successful call to the OnInitiateMessage or
// the OnRemoteDescription and if it's going to the STATE_RECEIVEDACCEPT.
sigslot::signal2<const std::string&, bool> SignalAddStream;
// This signal occurs when one stream is removed with the signaling
// message from the remote peer with the candidates port equal to 0.
sigslot::signal2<const std::string&, bool> SignalRemoveStream;
// This signal occurs when audio/video channel has been created for the
// new added stream.
sigslot::signal2<const std::string&, bool> SignalRtcMediaChannelCreated;
// Triggered when the local candidate is ready
// This signal occurs when the local candidate is ready
sigslot::signal2<const cricket::SessionDescription*,
const std::vector<cricket::Candidate>&> SignalLocalDescription;
// This callback will trigger if setting up a call times out.
// This signal triggers when setting up or resuming a call has not been
// successful before a certain time out.
sigslot::signal0<> SignalFailedCall;
bool muted() const { return muted_; }
@ -194,7 +209,7 @@ class WebRTCSession : public cricket::BaseSession {
cricket::ChannelManager* channel_manager_;
std::vector<StreamInfo*> streams_;
TransportChannelMap transport_channels_;
bool all_transports_writable_;
bool transports_writable_;
bool muted_;
bool camera_muted_;
int setup_timeout_;

83
third_party_mods/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc
View File

@ -168,7 +168,7 @@ class OnSignalImpl
enum CallbackId {
kNone,
kOnAddStream,
kOnRemoveStream2,
kOnRemoveStream,
kOnRtcMediaChannelCreated,
kOnLocalDescription,
kOnFailedCall,
@ -190,8 +190,8 @@ class OnSignalImpl
last_stream_id_ = stream_id;
last_was_video_ = video;
}
void OnRemoveStream2(const std::string& stream_id, bool video) {
callback_ids_.push_back(kOnRemoveStream2);
void OnRemoveStream(const std::string& stream_id, bool video) {
callback_ids_.push_back(kOnRemoveStream);
last_stream_id_ = stream_id;
last_was_video_ = video;
}
@ -310,14 +310,14 @@ typedef ReturnBoolPassArgument<
std::pair<std::string, cricket::VideoRenderer*> >
ReturnBoolPassStringVideoRenderer;
class WebRTCSessionExtendedForTest : public webrtc::WebRTCSession {
class WebRtcSessionExtendedForTest : public webrtc::WebRtcSession {
public:
WebRTCSessionExtendedForTest(const std::string& id,
WebRtcSessionExtendedForTest(const std::string& id,
const std::string& direction,
cricket::PortAllocator* allocator,
cricket::ChannelManager* channelmgr,
talk_base::Thread* signaling_thread)
: WebRTCSession(id, direction, allocator, channelmgr, signaling_thread),
: WebRtcSession(id, direction, allocator, channelmgr, signaling_thread),
worker_thread_(channelmgr->worker_thread()) {
}
private:
@ -330,7 +330,7 @@ class WebRTCSessionExtendedForTest : public webrtc::WebRTCSession {
talk_base::Thread* worker_thread_;
};
class WebRTCSessionTest : public OnSignalImpl,
class WebRtcSessionTest : public OnSignalImpl,
public talk_base::MessageHandler {
public:
enum FunctionCallId {
@ -338,8 +338,6 @@ class WebRTCSessionTest : public OnSignalImpl,
kCallConnect,
kCallOnRemoteDescription,
kCallOnInitiateMessage,
kCallOnMute,
kCallOnCameraMute,
kCallMuted,
kCallCameraMuted,
kCallCreateVoiceChannel,
@ -356,9 +354,9 @@ class WebRTCSessionTest : public OnSignalImpl,
enum {kInit = kCallLocalCandidates + 1};
enum {kTerminate = kInit + 1};
static WebRTCSessionTest* CreateWebRTCSessionTest(bool receiving) {
WebRTCSessionTest* return_value =
new WebRTCSessionTest();
static WebRtcSessionTest* CreateWebRtcSessionTest(bool receiving) {
WebRtcSessionTest* return_value =
new WebRtcSessionTest();
if (return_value == NULL) {
return NULL;
}
@ -436,16 +434,16 @@ class WebRTCSessionTest : public OnSignalImpl,
talk_base::CreateRandomString(8, &id_);
session_ = new webrtc::WebRTCSession(
session_ = new webrtc::WebRtcSession(
id_, DirectionAsString() , allocator_,
channel_manager_,
signaling_thread_);
session_->SignalAddStream.connect(
static_cast<OnSignalImpl*> (this),
&OnSignalImpl::OnAddStream);
session_->SignalRemoveStream2.connect(
session_->SignalRemoveStream.connect(
static_cast<OnSignalImpl*> (this),
&OnSignalImpl::OnRemoveStream2);
&OnSignalImpl::OnRemoveStream);
session_->SignalRtcMediaChannelCreated.connect(
static_cast<OnSignalImpl*> (this),
&OnSignalImpl::OnRtcMediaChannelCreated);
@ -466,7 +464,7 @@ class WebRTCSessionTest : public OnSignalImpl,
delete allocator_;
}
~WebRTCSessionTest() {
~WebRtcSessionTest() {
if (signaling_thread_ != NULL) {
signaling_thread_->Send(this, kTerminate, NULL);
signaling_thread_->Stop();
@ -509,28 +507,6 @@ class WebRTCSessionTest : public OnSignalImpl,
return return_value.return_value_;
}
void CallOnMute(bool mute) {
PassBool return_value(mute);
signaling_thread_->Send(this, kCallOnMute, &return_value);
}
void CallOnCameraMute(bool mute) {
PassBool return_value(mute);
signaling_thread_->Send(this, kCallOnCameraMute, &return_value);
}
bool CallMuted() {
ReturnBool return_value;
signaling_thread_->Send(this, kCallMuted, &return_value);
return return_value.return_value_;
}
bool CallCameraMuted() {
ReturnBool return_value;
signaling_thread_->Send(this, kCallCameraMuted, &return_value);
return return_value.return_value_;
}
bool CallCreateVoiceChannel(const std::string& stream_id) {
ReturnBoolPassString return_value(stream_id);
signaling_thread_->Send(this, kCallCreateVoiceChannel, &return_value);
@ -641,16 +617,6 @@ class WebRTCSessionTest : public OnSignalImpl,
return_value->return_value_ = true;
}
void OnMute_s(talk_base::Message* message) {
PassBool* return_value = reinterpret_cast<PassBool*>(message->pdata);
session_->OnMute(return_value->argument());
}
void OnCameraMute_s(talk_base::Message* message) {
PassBool* return_value = reinterpret_cast<PassBool*>(message->pdata);
session_->OnCameraMute(return_value->argument());
}
void Muted_s(talk_base::Message* message) {
ReturnBool* return_value = reinterpret_cast<ReturnBool*>(message->pdata);
return_value->return_value_ = session_->muted();
@ -747,12 +713,6 @@ class WebRTCSessionTest : public OnSignalImpl,
case kCallOnInitiateMessage:
OnInitiateMessage_s(message);
return;
case kCallOnMute:
OnMute_s(message);
return;
case kCallOnCameraMute:
OnCameraMute_s(message);
return;
case kCallMuted:
Muted_s(message);
return;
@ -802,7 +762,7 @@ class WebRTCSessionTest : public OnSignalImpl,
}
private:
WebRTCSessionTest()
WebRtcSessionTest()
: session_(NULL),
id_(),
receiving_(false),
@ -812,7 +772,7 @@ class WebRTCSessionTest : public OnSignalImpl,
signaling_thread_(NULL) {
}
webrtc::WebRTCSession* session_;
webrtc::WebRtcSession* session_;
std::string id_;
bool receiving_;
@ -824,7 +784,7 @@ class WebRTCSessionTest : public OnSignalImpl,
talk_base::Thread* signaling_thread_;
};
bool CallbackReceived(WebRTCSessionTest* session, int timeout) {
bool CallbackReceived(WebRtcSessionTest* session, int timeout) {
talk_base::Thread::SleepMs(timeout);
const OnSignalImpl::CallbackId peek_id =
session->PeekOldestCallback();
@ -837,8 +797,8 @@ void SleepMs(int timeout_ms) {
TEST(WebRtcSessionTest, InitializationReceiveSanity) {
const bool kReceiving = true;
talk_base::scoped_ptr<WebRTCSessionTest> my_session;
my_session.reset(WebRTCSessionTest::CreateWebRTCSessionTest(kReceiving));
talk_base::scoped_ptr<WebRtcSessionTest> my_session;
my_session.reset(WebRtcSessionTest::CreateWebRtcSessionTest(kReceiving));
ASSERT_TRUE(my_session.get() != NULL);
ASSERT_TRUE(my_session->CallInitiate());
@ -854,10 +814,11 @@ TEST(WebRtcSessionTest, InitializationReceiveSanity) {
my_session->PopOldestCallback());
}
// TODO(ronghuawu): Add tests for video calls and incoming calls.
TEST(WebRtcSessionTest, SendCallSetUp) {
const bool kReceiving = false;
talk_base::scoped_ptr<WebRTCSessionTest> my_session;
my_session.reset(WebRTCSessionTest::CreateWebRTCSessionTest(kReceiving));
talk_base::scoped_ptr<WebRtcSessionTest> my_session;
my_session.reset(WebRtcSessionTest::CreateWebRtcSessionTest(kReceiving));
ASSERT_TRUE(my_session.get() != NULL);
ASSERT_TRUE(my_session->CallInitiate());