Update to libjingle 0.6.1.
- Add dummydevicemanager.
- Remove underscores from the files names for app/webrtc folder.
- Remove PeerConnection OnLocalStreamInitialized callback.
- Fix webrtcjson.cc numeric locale formatting issue.
- Don't start playout until the local content has been set.
Review URL: http://webrtc-codereview.appspot.com/149003/
git-svn-id: http://libjingle.googlecode.com/svn/trunk@85 dd674b97-3498-5ee5-1854-bdd07cd0ff33
diff --git a/CHANGELOG b/CHANGELOG
index d5c6d59..f846076 100644
--- a/CHANGELOG
+++ b/CHANGELOG
@@ -1,5 +1,12 @@
Libjingle
+0.6.1 - Sep 15, 2011
+ - Add dummydevicemanager.
+ - Remove underscores from the files names for app/webrtc folder.
+ - Remove PeerConnection OnLocalStreamInitialized callback.
+ - Fix webrtcjson.cc numeric locale formatting issue.
+ - Don't start playout until the local content has been set.
+
0.6.0 - Sep 13, 2011
- Add pub sub support
- Add unit tests
diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h
index 15c34b4..62c2504 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -51,14 +51,6 @@
// serialized signaling message
virtual void OnSignalingMessage(const std::string& msg) = 0;
- // Triggered when a local stream has been added and initialized
- // TODO: This callback was added intense to give the client a
- // chance to set the capture device. It doesn't map to any JS callback. Once
- // we figure out a better way to set video capture device, we should remove
- // this callback.
- virtual void OnLocalStreamInitialized(const std::string& stream_id,
- bool video) = 0;
-
// Triggered when a remote peer accepts a media connection.
virtual void OnAddStream(const std::string& stream_id, bool video) = 0;
@@ -88,8 +80,7 @@
virtual bool SignalingMessage(const std::string& msg) = 0;
// Asynchronously adds a local stream device to the peer
- // connection. The operation is complete when
- // PeerConnectionObserver::OnLocalStreamInitialized is called.
+ // connection.
virtual bool AddStream(const std::string& stream_id, bool video) = 0;
// Asynchronously removes a local stream device from the peer
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index 0930bde..1b40c14 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -27,7 +27,7 @@
#include "talk/app/webrtc/peerconnectionfactory.h"
-#include "talk/app/webrtc/peerconnection_proxy.h"
+#include "talk/app/webrtc/peerconnectionproxy.h"
#include "talk/base/logging.h"
#include "talk/p2p/client/basicportallocator.h"
#include "talk/session/phone/channelmanager.h"
@@ -37,7 +37,7 @@
PeerConnectionFactory::PeerConnectionFactory(
cricket::PortAllocator* port_allocator,
cricket::MediaEngineInterface* media_engine,
- cricket::DeviceManager* device_manager,
+ cricket::DeviceManagerInterface* device_manager,
talk_base::Thread* worker_thread)
: initialized_(false),
port_allocator_(port_allocator),
@@ -58,7 +58,7 @@
}
bool PeerConnectionFactory::Initialize() {
- ASSERT(channel_manager_.get());
+ ASSERT(channel_manager_.get() != NULL);
initialized_ = channel_manager_->Init();
return initialized_;
}
diff --git a/talk/app/webrtc/peerconnection_impl.cc b/talk/app/webrtc/peerconnectionimpl.cc
similarity index 93%
rename from talk/app/webrtc/peerconnection_impl.cc
rename to talk/app/webrtc/peerconnectionimpl.cc
index f16f3f2..28e4685 100644
--- a/talk/app/webrtc/peerconnection_impl.cc
+++ b/talk/app/webrtc/peerconnectionimpl.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/app/webrtc/peerconnection_impl.h"
+#include "talk/app/webrtc/peerconnectionimpl.h"
-#include "talk/app/webrtc/webrtc_json.h"
+#include "talk/app/webrtc/webrtcjson.h"
#include "talk/app/webrtc/webrtcsession.h"
#include "talk/base/basicpacketsocketfactory.h"
#include "talk/base/helpers.h"
@@ -108,9 +108,6 @@
session->SignalRemoveStream.connect(
this,
&PeerConnectionImpl::OnRemoveStream);
- session->SignalRtcMediaChannelCreated.connect(
- this,
- &PeerConnectionImpl::OnRtcMediaChannelCreated);
session->SignalLocalDescription.connect(
this,
&PeerConnectionImpl::OnLocalDescription);
@@ -210,13 +207,6 @@
}
}
-void PeerConnectionImpl::OnRtcMediaChannelCreated(const std::string& stream_id,
- bool video) {
- if (event_callback_) {
- event_callback_->OnLocalStreamInitialized(stream_id, video);
- }
-}
-
PeerConnectionImpl::ReadyState PeerConnectionImpl::GetReadyState() {
ReadyState ready_state;
cricket::BaseSession::State state = session_->state();
diff --git a/talk/app/webrtc/peerconnection_impl.h b/talk/app/webrtc/peerconnectionimpl.h
similarity index 93%
rename from talk/app/webrtc/peerconnection_impl.h
rename to talk/app/webrtc/peerconnectionimpl.h
index 4c004ec..6ff2f25 100644
--- a/talk/app/webrtc/peerconnection_impl.h
+++ b/talk/app/webrtc/peerconnectionimpl.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_APP_WEBRTC_PEERCONNECTION_IMPL_H_
-#define TALK_APP_WEBRTC_PEERCONNECTION_IMPL_H_
+#ifndef TALK_APP_WEBRTC_PEERCONNECTIONIMPL_H_
+#define TALK_APP_WEBRTC_PEERCONNECTIONIMPL_H_
#include <string>
#include <vector>
@@ -80,8 +80,6 @@
const cricket::SessionDescription* desc,
const std::vector<cricket::Candidate>& candidates);
void OnFailedCall();
- void OnRtcMediaChannelCreated(const std::string& stream_id,
- bool video);
bool Init();
private:
@@ -96,4 +94,4 @@
} // namespace webrtc
-#endif // TALK_APP_WEBRTC_PEERCONNECTION_IMPL_H_
+#endif // TALK_APP_WEBRTC_PEERCONNECTIONIMPL_H_
diff --git a/talk/app/webrtc/peerconnection_proxy.cc b/talk/app/webrtc/peerconnectionproxy.cc
similarity index 98%
rename from talk/app/webrtc/peerconnection_proxy.cc
rename to talk/app/webrtc/peerconnectionproxy.cc
index 6591151..fca3ad4 100644
--- a/talk/app/webrtc/peerconnection_proxy.cc
+++ b/talk/app/webrtc/peerconnectionproxy.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/app/webrtc/peerconnection_proxy.h"
+#include "talk/app/webrtc/peerconnectionproxy.h"
-#include "talk/app/webrtc/peerconnection_impl.h"
+#include "talk/app/webrtc/peerconnectionimpl.h"
#include "talk/base/logging.h"
namespace webrtc {
diff --git a/talk/app/webrtc/peerconnection_proxy.h b/talk/app/webrtc/peerconnectionproxy.h
similarity index 95%
rename from talk/app/webrtc/peerconnection_proxy.h
rename to talk/app/webrtc/peerconnectionproxy.h
index 6e7fa6c..e83b4ec 100644
--- a/talk/app/webrtc/peerconnection_proxy.h
+++ b/talk/app/webrtc/peerconnectionproxy.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_APP_WEBRTC_PEERCONNECTION_PROXY_H_
-#define TALK_APP_WEBRTC_PEERCONNECTION_PROXY_H_
+#ifndef TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_
+#define TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_
#include <string>
@@ -78,4 +78,4 @@
};
}
-#endif // TALK_APP_WEBRTC_PEERCONNECTION_PROXY_H_
+#endif // TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_
diff --git a/talk/app/webrtc/webrtc.scons b/talk/app/webrtc/webrtc.scons
index 2394e69..271d413 100644
--- a/talk/app/webrtc/webrtc.scons
+++ b/talk/app/webrtc/webrtc.scons
@@ -8,11 +8,10 @@
env,
name = 'webrtc',
srcs = [
- # TODO: remove underscores from filenames
- 'peerconnection_impl.cc',
- 'peerconnection_proxy.cc',
+ 'peerconnectionimpl.cc',
+ 'peerconnectionproxy.cc',
'peerconnectionfactory.cc',
- 'webrtc_json.cc',
+ 'webrtcjson.cc',
'webrtcsession.cc',
],
)
@@ -26,13 +25,14 @@
'webrtcsession_unittest.cc',
],
libs = [
- 'srtp',
'base',
+ 'expat',
'jpeg',
'json',
'webrtc',
'p2p',
'phone',
+ 'srtp',
'xmpp',
'xmllite',
'yuvscaler'
@@ -47,6 +47,7 @@
'IOKit',
'QTKit',
],
+ win_link_flags = [('', '/nodefaultlib:libcmt')[env.Bit('debug')]],
lin_libs = [
'rt',
'dl',
diff --git a/talk/app/webrtc/webrtc_json.cc b/talk/app/webrtc/webrtcjson.cc
similarity index 89%
rename from talk/app/webrtc/webrtc_json.cc
rename to talk/app/webrtc/webrtcjson.cc
index d0e0256..1662e23 100644
--- a/talk/app/webrtc/webrtc_json.cc
+++ b/talk/app/webrtc/webrtcjson.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/app/webrtc/webrtc_json.h"
+#include "talk/app/webrtc/webrtcjson.h"
#include <stdio.h>
#include <string>
@@ -33,40 +33,13 @@
#include "talk/base/json.h"
#include "talk/base/logging.h"
#include "talk/base/stringutils.h"
-#include "talk/session/phone/mediasessionclient.h"
#include "talk/session/phone/codec.h"
+#include "talk/session/phone/mediasessionclient.h"
namespace webrtc {
static const int kIceComponent = 1;
static const int kIceFoundation = 1;
-bool GetStunServer(const Json::Value& value, StunServiceDetails* stunServer) {
- if (value.type() != Json::objectValue && value.type() != Json::nullValue) {
- LOG(LS_WARNING) << "Failed to parse stun values";
- return false;
- }
-
- Json::Value stun;
- if (GetValueFromJsonObject(value, "stun_service", &stun)) {
- if (stun.type() == Json::objectValue) {
- if (!GetStringFromJsonObject(stun, "host", &stunServer->host) ||
- !GetStringFromJsonObject(stun, "service", &stunServer->service) ||
- !GetStringFromJsonObject(stun, "protocol", &stunServer->protocol)) {
- LOG(LS_WARNING) << "Failed to parse JSON value: "
- << value.toStyledString();
- return false;
- }
- } else {
- LOG(LS_WARNING) << "Failed to find the stun_service member.";
- return false;
- }
- } else {
- LOG(LS_WARNING) << "Wrong ValueType. Expect Json::objectValue).";
- return false;
- }
- return true;
-}
-
bool GetJsonSignalingMessage(
const cricket::SessionDescription* sdp,
const std::vector<cricket::Candidate>& candidates,
@@ -90,8 +63,9 @@
Json::Value signal;
Append(&signal, "media", media);
- // now serialize
+ // Now serialize.
*signaling_message = Serialize(signal);
+
return true;
}
@@ -191,7 +165,7 @@
Append(&candidate, "foundation", kIceFoundation);
Append(&candidate, "generation", iter->generation());
Append(&candidate, "proto", iter->protocol());
- Append(&candidate, "priority", iter->preference());
+ Append(&candidate, "priority", iter->preference_str());
Append(&candidate, "ip", iter->address().IPAsString());
Append(&candidate, "port", iter->address().PortAsString());
Append(&candidate, "type", iter->type());
@@ -344,10 +318,10 @@
return false;
cand.set_protocol(proto);
- double priority;
- if (!GetDoubleFromJsonObject(*iter, "priority", &priority))
+ std::string priority;
+ if (!GetStringFromJsonObject(*iter, "priority", &priority))
return false;
- cand.set_preference_str(talk_base::ToString(priority));
+ cand.set_preference_str(priority);
std::string str;
talk_base::SocketAddress addr;
@@ -390,7 +364,7 @@
std::vector<Json::Value> ReadValues(
const Json::Value& value, const std::string& key) {
std::vector<Json::Value> objects;
- for (size_t i = 0; i < value[key].size(); ++i) {
+ for (Json::Value::ArrayIndex i = 0; i < value[key].size(); ++i) {
objects.push_back(value[key][i]);
}
return objects;
@@ -412,7 +386,6 @@
return value[key].asDouble();
}
-// Add values
void Append(Json::Value* object, const std::string& key, bool value) {
(*object)[key] = Json::Value(value);
}
@@ -420,19 +393,16 @@
void Append(Json::Value* object, const std::string& key, char * value) {
(*object)[key] = Json::Value(value);
}
-void Append(Json::Value* object, const std::string& key, double value) {
- (*object)[key] = Json::Value(value);
-}
-void Append(Json::Value* object, const std::string& key, float value) {
- (*object)[key] = Json::Value(value);
-}
+
void Append(Json::Value* object, const std::string& key, int value) {
(*object)[key] = Json::Value(value);
}
+
void Append(Json::Value* object, const std::string& key,
const std::string& value) {
(*object)[key] = Json::Value(value);
}
+
void Append(Json::Value* object, const std::string& key, uint32 value) {
(*object)[key] = Json::Value(value);
}
diff --git a/talk/app/webrtc/webrtc_json.h b/talk/app/webrtc/webrtcjson.h
similarity index 91%
rename from talk/app/webrtc/webrtc_json.h
rename to talk/app/webrtc/webrtcjson.h
index b57adf0..906326d 100644
--- a/talk/app/webrtc/webrtc_json.h
+++ b/talk/app/webrtc/webrtcjson.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_APP_WEBRTC_WEBRTC_JSON_H_
-#define TALK_APP_WEBRTC_WEBRTC_JSON_H_
+#ifndef TALK_APP_WEBRTC_WEBRTCJSON_H_
+#define TALK_APP_WEBRTC_WEBRTCJSON_H_
#include <string>
@@ -35,8 +35,8 @@
#else
#include "third_party/jsoncpp/json.h"
#endif
-#include "talk/session/phone/codec.h"
#include "talk/p2p/base/candidate.h"
+#include "talk/session/phone/codec.h"
namespace Json {
class Value;
@@ -49,12 +49,6 @@
class SessionDescription;
}
-struct StunServiceDetails {
- std::string host;
- std::string service;
- std::string protocol;
-};
-
namespace webrtc {
std::vector<Json::Value> ReadValues(const Json::Value& value,
@@ -98,12 +92,8 @@
double ReadDouble(const Json::Value& value, const std::string& key);
uint32 ReadUInt(const Json::Value& value, const std::string& key);
-// Add values
void Append(Json::Value* object, const std::string& key, bool value);
-
void Append(Json::Value* object, const std::string& key, char * value);
-void Append(Json::Value* object, const std::string& key, double value);
-void Append(Json::Value* object, const std::string& key, float value);
void Append(Json::Value* object, const std::string& key, int value);
void Append(Json::Value* object, const std::string& key,
const std::string& value);
@@ -115,4 +105,4 @@
const std::vector<Json::Value>& values);
}
-#endif // TALK_APP_WEBRTC_WEBRTC_JSON_H_
+#endif // TALK_APP_WEBRTC_WEBRTCJSON_H_
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index d8046c4..1048828 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -132,9 +132,6 @@
ASSERT(voice_channel != NULL);
stream_info->channel = voice_channel;
- if (!incoming()) {
- SignalRtcMediaChannelCreated(stream_id, false);
- }
return true;
}
@@ -149,9 +146,6 @@
ASSERT(video_channel != NULL);
stream_info->channel = video_channel;
- if (!incoming()) {
- SignalRtcMediaChannelCreated(stream_id, true);
- }
return true;
}
@@ -457,17 +451,33 @@
if (!incoming()) {
// Trigger OnAddStream callback at the initiator
const cricket::ContentInfo* video_content = GetFirstVideoContent(desc);
- if (video_content) {
- SignalAddStream(video_content->name, true);
+ if (video_content && !SendSignalAddStream(true)) {
+ LOG(LERROR) << "Video stream unexpected in answer.";
+ return false;
} else {
const cricket::ContentInfo* audio_content = GetFirstAudioContent(desc);
- if (audio_content)
- SignalAddStream(audio_content->name, false);
+ if (audio_content && !SendSignalAddStream(false)) {
+ LOG(LERROR) << "Audio stream unexpected in answer.";
+ return false;
+ }
}
}
return true;
}
+// Send the SignalAddStream with the stream_id based on the content type.
+bool WebRtcSession::SendSignalAddStream(bool video) {
+ StreamMap::const_iterator iter;
+ for (iter = streams_.begin(); iter != streams_.end(); ++iter) {
+ StreamInfo* sinfo = (*iter);
+ if (sinfo->video == video) {
+ SignalAddStream(sinfo->stream_id, video);
+ return true;
+ }
+ }
+ return false;
+}
+
cricket::SessionDescription* WebRtcSession::CreateOffer() {
cricket::SessionDescription* offer = new cricket::SessionDescription();
StreamMap::iterator iter;
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index bbddb07..11e389e 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -106,10 +106,6 @@
// 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;
-
// This signal occurs when the local candidate is ready
sigslot::signal2<const cricket::SessionDescription*,
const std::vector<cricket::Candidate>&> SignalLocalDescription;
@@ -186,6 +182,7 @@
bool SetVideoCapture(bool capture);
void EnableAllStreams();
+ bool SendSignalAddStream(bool video);
cricket::Transport* transport_;
cricket::ChannelManager* channel_manager_;
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 648a5d6..7a07614 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -50,7 +50,6 @@
kNone,
kOnAddStream,
kOnRemoveStream,
- kOnRtcMediaChannelCreated,
kOnLocalDescription,
kOnFailedCall,
};
@@ -84,12 +83,6 @@
last_stream_id_ = stream_id;
last_was_video_ = video;
}
- void OnRtcMediaChannelCreated(const std::string& stream_id,
- bool video) {
- callback_ids_.push_back(kOnRtcMediaChannelCreated);
- last_stream_id_ = stream_id;
- last_was_video_ = video;
- }
void OnLocalDescription(
const cricket::SessionDescription* desc,
const std::vector<cricket::Candidate>& candidates) {
@@ -183,8 +176,6 @@
session_->SignalAddStream.connect(this, &WebRtcSessionTest::OnAddStream);
session_->SignalRemoveStream.connect(this,
&WebRtcSessionTest::OnRemoveStream);
- session_->SignalRtcMediaChannelCreated.connect(this,
- &WebRtcSessionTest::OnRtcMediaChannelCreated);
session_->SignalLocalDescription.connect(this,
&WebRtcSessionTest::OnLocalDescription);
session_->SignalFailedCall.connect(this, &WebRtcSessionTest::OnFailedCall);
@@ -234,9 +225,6 @@
if (!session_->CreateVoiceChannel(stream_id)) {
return false;
}
- if (!WaitForCallback(kOnRtcMediaChannelCreated, 1000)) {
- return false;
- }
return true;
}
@@ -244,9 +232,6 @@
if (!session_->CreateVideoChannel(stream_id)) {
return false;
}
- if (!WaitForCallback(kOnRtcMediaChannelCreated, 1000)) {
- return false;
- }
return true;
}
diff --git a/talk/base/asynchttprequest.cc b/talk/base/asynchttprequest.cc
index 1e05d82..68f6100 100644
--- a/talk/base/asynchttprequest.cc
+++ b/talk/base/asynchttprequest.cc
@@ -29,7 +29,10 @@
namespace talk_base {
-enum { MSG_TIMEOUT = SignalThread::ST_MSG_FIRST_AVAILABLE };
+enum {
+ MSG_TIMEOUT = SignalThread::ST_MSG_FIRST_AVAILABLE,
+ MSG_LAUNCH_REQUEST
+};
static const int kDefaultHTTPTimeout = 30 * 1000; // 30 sec
///////////////////////////////////////////////////////////////////////////////
@@ -37,10 +40,16 @@
///////////////////////////////////////////////////////////////////////////////
AsyncHttpRequest::AsyncHttpRequest(const std::string &user_agent)
- : firewall_(NULL), port_(80), secure_(false),
- timeout_(kDefaultHTTPTimeout), fail_redirect_(false),
+ : start_delay_(0),
+ firewall_(NULL),
+ port_(80),
+ secure_(false),
+ timeout_(kDefaultHTTPTimeout),
+ fail_redirect_(false),
factory_(Thread::Current()->socketserver(), user_agent),
- pool_(&factory_), client_(user_agent.c_str(), &pool_), error_(HE_NONE) {
+ pool_(&factory_),
+ client_(user_agent.c_str(), &pool_),
+ error_(HE_NONE) {
client_.SignalHttpClientComplete.connect(this,
&AsyncHttpRequest::OnComplete);
}
@@ -49,22 +58,11 @@
}
void AsyncHttpRequest::OnWorkStart() {
- factory_.SetProxy(proxy_);
- if (secure_)
- factory_.UseSSL(host_.c_str());
-
- bool transparent_proxy = (port_ == 80) &&
- ((proxy_.type == PROXY_HTTPS) || (proxy_.type == PROXY_UNKNOWN));
- if (transparent_proxy) {
- client_.set_proxy(proxy_);
+ if (start_delay_ <= 0) {
+ LaunchRequest();
+ } else {
+ Thread::Current()->PostDelayed(start_delay_, this, MSG_LAUNCH_REQUEST);
}
- client_.set_fail_redirect(fail_redirect_);
- client_.set_server(SocketAddress(host_, port_));
-
- LOG(LS_INFO) << "HttpRequest start: " << host_ + client_.request().path;
-
- Thread::Current()->PostDelayed(timeout_, this, MSG_TIMEOUT);
- client_.start();
}
void AsyncHttpRequest::OnWorkStop() {
@@ -91,14 +89,19 @@
}
void AsyncHttpRequest::OnMessage(Message* message) {
- if (message->message_id != MSG_TIMEOUT) {
+ switch (message->message_id) {
+ case MSG_TIMEOUT:
+ LOG(LS_INFO) << "HttpRequest timed out";
+ client_.reset();
+ worker()->Quit();
+ break;
+ case MSG_LAUNCH_REQUEST:
+ LaunchRequest();
+ break;
+ default:
SignalThread::OnMessage(message);
- return;
+ break;
}
-
- LOG(LS_INFO) << "HttpRequest timed out";
- client_.reset();
- worker()->Quit();
}
void AsyncHttpRequest::DoWork() {
@@ -108,4 +111,23 @@
Thread::Current()->ProcessMessages(kForever);
}
+void AsyncHttpRequest::LaunchRequest() {
+ factory_.SetProxy(proxy_);
+ if (secure_)
+ factory_.UseSSL(host_.c_str());
+
+ bool transparent_proxy = (port_ == 80) &&
+ ((proxy_.type == PROXY_HTTPS) || (proxy_.type == PROXY_UNKNOWN));
+ if (transparent_proxy) {
+ client_.set_proxy(proxy_);
+ }
+ client_.set_fail_redirect(fail_redirect_);
+ client_.set_server(SocketAddress(host_, port_));
+
+ LOG(LS_INFO) << "HttpRequest start: " << host_ + client_.request().path;
+
+ Thread::Current()->PostDelayed(timeout_, this, MSG_TIMEOUT);
+ client_.start();
+}
+
} // namespace talk_base
diff --git a/talk/base/asynchttprequest.h b/talk/base/asynchttprequest.h
index f5d7f46..a1e8aa0 100644
--- a/talk/base/asynchttprequest.h
+++ b/talk/base/asynchttprequest.h
@@ -50,6 +50,11 @@
explicit AsyncHttpRequest(const std::string &user_agent);
~AsyncHttpRequest();
+ // If start_delay is less than or equal to zero, this starts immediately.
+ // Start_delay defaults to zero.
+ int start_delay() const { return start_delay_; }
+ void set_start_delay(int delay) { start_delay_ = delay; }
+
void set_proxy(const ProxyInfo& proxy) {
proxy_ = proxy;
}
@@ -93,6 +98,9 @@
virtual void DoWork();
private:
+ void LaunchRequest();
+
+ int start_delay_;
ProxyInfo proxy_;
FirewallManager* firewall_;
std::string host_;
diff --git a/talk/base/asynchttprequest_unittest.cc b/talk/base/asynchttprequest_unittest.cc
index e8db1e0..b7a8a03 100644
--- a/talk/base/asynchttprequest_unittest.cc
+++ b/talk/base/asynchttprequest_unittest.cc
@@ -65,10 +65,13 @@
public sigslot::has_slots<> {
public:
AsyncHttpRequestTest()
- : done_(false), server_(Thread::Current(), kServerAddr) {
+ : started_(false),
+ done_(false),
+ server_(Thread::Current(), kServerAddr) {
server_.SignalHttpRequest.connect(this, &AsyncHttpRequestTest::OnRequest);
}
+ bool started() const { return started_; }
bool done() const { return done_; }
AsyncHttpRequest* CreateGetRequest(const std::string& host, int port,
@@ -105,6 +108,8 @@
protected:
void OnRequest(HttpServer* server, HttpServerTransaction* t) {
+ started_ = true;
+
if (t->request.path == kServerGetPath) {
t->response.set_success("text/plain", new MemoryStream(kServerResponse));
} else if (t->request.path == kServerPostPath) {
@@ -131,6 +136,7 @@
}
private:
+ bool started_;
bool done_;
TestHttpServer server_;
};
@@ -139,7 +145,9 @@
AsyncHttpRequest* req = CreateGetRequest(
kServerHostnameAddr.hostname(), server().address().port(),
kServerGetPath);
+ EXPECT_FALSE(started());
req->Start();
+ EXPECT_TRUE_WAIT(started(), 100); // Should have started by now.
EXPECT_TRUE_WAIT(done(), 5000);
std::string response;
EXPECT_EQ(200U, req->response().scode);
@@ -188,4 +196,23 @@
req->Destroy(true);
}
+TEST_F(AsyncHttpRequestTest, TestGetSuccessDelay) {
+ AsyncHttpRequest* req = CreateGetRequest(
+ kServerHostnameAddr.hostname(), server().address().port(),
+ kServerGetPath);
+ req->set_start_delay(10); // Delay 10ms.
+ req->Start();
+ Thread::SleepMs(5);
+ EXPECT_FALSE(started()); // Should not have started immediately.
+ EXPECT_TRUE_WAIT(started(), 200); // Should have started by now.
+ EXPECT_TRUE_WAIT(done(), 5000);
+ std::string response;
+ EXPECT_EQ(200U, req->response().scode);
+ ASSERT_TRUE(req->response().document.get() != NULL);
+ req->response().document->Rewind();
+ req->response().document->ReadLine(&response);
+ EXPECT_EQ(kServerResponse, response);
+ req->Release();
+}
+
} // namespace talk_base
diff --git a/talk/base/json.h b/talk/base/json.h
index 5685503..d8cee35 100644
--- a/talk/base/json.h
+++ b/talk/base/json.h
@@ -31,7 +31,11 @@
#include <string>
#include <vector>
+#ifdef JSONCPP_RELATIVE_PATH
+#include "json/json.h"
+#else
#include "third_party/jsoncpp/json.h"
+#endif
// TODO: Move to talk_base namespace
diff --git a/talk/base/testechoserver.h b/talk/base/testechoserver.h
index 0b645d1..2c65167 100644
--- a/talk/base/testechoserver.h
+++ b/talk/base/testechoserver.h
@@ -1,5 +1,29 @@
-// Copyright 2008 Google Inc. All Rights Reserved.
-
+/*
+ * libjingle
+ * Copyright 2004--2011, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
#ifndef TALK_BASE_TESTECHOSERVER_H_
#define TALK_BASE_TESTECHOSERVER_H_
diff --git a/talk/libjingle.scons b/talk/libjingle.scons
index 5e029b0..358f817 100644
--- a/talk/libjingle.scons
+++ b/talk/libjingle.scons
@@ -90,6 +90,10 @@
"pthread",
"rt",
],
+ 'mac_libs': [
+ "-lcrypto",
+ "-lssl",
+ ],
},
mac_srcs = [
"base/macasyncsocket.cc",
@@ -225,13 +229,13 @@
"xmllite/xmlparser.cc",
"xmllite/xmlprinter.cc",
"xmpp/constants.cc",
+ "xmpp/hangoutpubsubclient.cc",
"xmpp/iqtask.cc",
"xmpp/jid.cc",
- "xmpp/pubsubtasks.cc",
- "xmpp/pubsubclient.cc",
- "xmpp/hangoutpubsubclient.cc",
"xmpp/mucroomconfigtask.cc",
"xmpp/mucroomlookuptask.cc",
+ "xmpp/pubsubclient.cc",
+ "xmpp/pubsubtasks.cc",
"xmpp/ratelimitmanager.cc",
"xmpp/receivetask.cc",
"xmpp/saslmechanism.cc",
diff --git a/talk/session/phone/channel.cc b/talk/session/phone/channel.cc
index 32ff5c1..f41c2ad 100644
--- a/talk/session/phone/channel.cc
+++ b/talk/session/phone/channel.cc
@@ -87,7 +87,8 @@
enabled_(false),
writable_(false),
was_ever_writable_(false),
- has_codec_(false),
+ has_local_content_(false),
+ has_remote_content_(false),
muted_(false) {
ASSERT(worker_thread_ == talk_base::Thread::Current());
media_channel_->SetInterface(this);
@@ -750,16 +751,16 @@
}
void VoiceChannel::ChangeState() {
- // render incoming data if we are the active call
- // we receive data on the default channel and multiplexed streams
- bool recv = enabled();
+ // Render incoming data if we're the active call, and we have the local
+ // content. We receive data on the default channel and multiplexed streams.
+ bool recv = enabled() && has_local_content();
if (!media_channel()->SetPlayout(recv)) {
SendLastMediaError();
}
-
- // Send outgoing data if we are the active call and we know their codec, and
- // we have had some form of connectivity.
- bool send = enabled() && has_codec() && was_ever_writable();
+
+ // Send outgoing data if we're the active call, we have the remote content,
+ // and we have had some form of connectivity.
+ bool send = enabled() && has_remote_content() && was_ever_writable();
SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING;
if (!media_channel()->SetSend(send_flag)) {
LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel";
@@ -807,6 +808,12 @@
ret = media_channel()->SetRecvRtpHeaderExtensions(
audio->rtp_header_extensions());
}
+ if (ret) {
+ set_has_local_content(true);
+ ChangeState();
+ } else {
+ LOG(LS_WARNING) << "Failed to set local voice description";
+ }
return ret;
}
@@ -847,8 +854,10 @@
// update state
if (ret) {
- set_has_codec(true);
+ set_has_remote_content(true);
ChangeState();
+ } else {
+ LOG(LS_WARNING) << "Failed to set remote voice description";
}
return ret;
}
@@ -1056,17 +1065,17 @@
}
void VideoChannel::ChangeState() {
- // render incoming data if we are the active call
- // we receive data on the default channel and multiplexed streams
- bool recv = enabled();
+ // Render incoming data if we're the active call, and we have the local
+ // content. We receive data on the default channel and multiplexed streams.
+ bool recv = enabled() && has_local_content();
if (!media_channel()->SetRender(recv)) {
LOG(LS_ERROR) << "Failed to SetRender on video channel";
// TODO: Report error back to server.
}
- // Send outgoing data if we are the active call and we know their codec, and
- // we have had some form of connectivity.
- bool send = enabled() && has_codec() && was_ever_writable();
+ // Send outgoing data if we're the active call, we have the remote content,
+ // and we have had some form of connectivity.
+ bool send = enabled() && has_remote_content() && was_ever_writable();
if (!media_channel()->SetSend(send)) {
LOG(LS_ERROR) << "Failed to SetSend on video channel";
// TODO: Report error back to server.
@@ -1127,6 +1136,12 @@
ret = media_channel()->SetRecvRtpHeaderExtensions(
video->rtp_header_extensions());
}
+ if (ret) {
+ set_has_local_content(true);
+ ChangeState();
+ } else {
+ LOG(LS_WARNING) << "Failed to set local video description";
+ }
return ret;
}
@@ -1161,8 +1176,10 @@
video->rtp_header_extensions());
}
if (ret) {
- set_has_codec(true);
+ set_has_remote_content(true);
ChangeState();
+ } else {
+ LOG(LS_WARNING) << "Failed to set remote video description";
}
return ret;
}
diff --git a/talk/session/phone/channel.h b/talk/session/phone/channel.h
index 157999d..a35ed86 100644
--- a/talk/session/phone/channel.h
+++ b/talk/session/phone/channel.h
@@ -164,8 +164,10 @@
void set_rtcp_transport_channel(TransportChannel* transport);
bool writable() const { return writable_; }
bool was_ever_writable() const { return was_ever_writable_; }
- bool has_codec() const { return has_codec_; }
- void set_has_codec(bool has_codec) { has_codec_ = has_codec; }
+ bool has_local_content() const { return has_local_content_; }
+ bool has_remote_content() const { return has_remote_content_; }
+ void set_has_local_content(bool has) { has_local_content_ = has; }
+ void set_has_remote_content(bool has) { has_remote_content_ = has; }
bool muted() const { return muted_; }
talk_base::Thread* signaling_thread() { return session_->signaling_thread(); }
SrtpFilter* srtp_filter() { return &srtp_filter_; }
@@ -276,7 +278,8 @@
bool enabled_;
bool writable_;
bool was_ever_writable_;
- bool has_codec_;
+ bool has_local_content_;
+ bool has_remote_content_;
bool muted_;
};
diff --git a/talk/session/phone/dummydevicemanager.h b/talk/session/phone/dummydevicemanager.h
new file mode 100644
index 0000000..18bc63b
--- /dev/null
+++ b/talk/session/phone/dummydevicemanager.h
@@ -0,0 +1,51 @@
+/*
+ * libjingle
+ * Copyright 2011, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_SESSION_PHONE_DUMMYDEVICEMANAGER_H_
+#define TALK_SESSION_PHONE_DUMMYDEVICEMANAGER_H_
+
+#include <vector>
+
+#include "talk/session/phone/fakedevicemanager.h"
+#include "talk/session/phone/mediacommon.h"
+
+namespace cricket {
+
+class DummyDeviceManager : public FakeDeviceManager {
+ public:
+ DummyDeviceManager() {
+ std::vector<std::string> devices;
+ devices.push_back(DeviceManagerInterface::kDefaultDeviceName);
+ SetAudioInputDevices(devices);
+ SetAudioOutputDevices(devices);
+ SetVideoCaptureDevices(devices);
+ }
+};
+
+} // namespace cricket
+
+#endif // TALK_SESSION_PHONE_DUMMYDEVICEMANAGER_H_
diff --git a/talk/session/phone/fakedevicemanager.h b/talk/session/phone/fakedevicemanager.h
new file mode 100644
index 0000000..c096007
--- /dev/null
+++ b/talk/session/phone/fakedevicemanager.h
@@ -0,0 +1,161 @@
+/*
+ * libjingle
+ * Copyright 2008--2011, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_SESSION_PHONE_FAKEDEVICEMANAGER_H_
+#define TALK_SESSION_PHONE_FAKEDEVICEMANAGER_H_
+
+#include <string>
+#include <vector>
+
+#include "talk/session/phone/devicemanager.h"
+#include "talk/session/phone/mediacommon.h"
+
+namespace cricket {
+
+class FakeDeviceManager : public DeviceManagerInterface {
+ public:
+ FakeDeviceManager() {}
+ virtual bool Init() {
+ return true;
+ }
+ virtual void Terminate() {
+ }
+ virtual int GetCapabilities() {
+ std::vector<Device> devices;
+ int caps = VIDEO_RECV;
+ if (!input_devices_.empty()) {
+ caps |= AUDIO_SEND;
+ }
+ if (!output_devices_.empty()) {
+ caps |= AUDIO_RECV;
+ }
+ if (!vidcap_devices_.empty()) {
+ caps |= VIDEO_SEND;
+ }
+ return caps;
+ }
+ virtual bool GetAudioInputDevices(std::vector<Device>* devs) {
+ *devs = input_devices_;
+ return true;
+ }
+ virtual bool GetAudioOutputDevices(std::vector<Device>* devs) {
+ *devs = output_devices_;
+ return true;
+ }
+ virtual bool GetAudioInputDevice(const std::string& name, Device* out) {
+ return GetAudioDevice(true, name, out);
+ }
+ virtual bool GetAudioOutputDevice(const std::string& name, Device* out) {
+ return GetAudioDevice(false, name, out);
+ }
+ virtual bool GetVideoCaptureDevices(std::vector<Device>* devs) {
+ *devs = vidcap_devices_;
+ return true;
+ }
+
+ virtual bool GetDefaultVideoCaptureDevice(Device* device) {
+ if (vidcap_devices_.empty()) {
+ return false;
+ }
+ *device = vidcap_devices_[0];
+ return true;
+ }
+
+#ifdef OSX
+ bool QtKitToSgDevice(const std::string& qtkit_name, Device* out) {
+ out->name = qtkit_name;
+ out->id = "sg:" + qtkit_name;
+ return true;
+ }
+#endif
+
+ void SetAudioInputDevices(const std::vector<std::string>& devices) {
+ input_devices_.clear();
+ for (size_t i = 0; i < devices.size(); ++i) {
+ input_devices_.push_back(Device(devices[i], i));
+ }
+ SignalDevicesChange();
+ }
+ void SetAudioOutputDevices(const std::vector<std::string>& devices) {
+ output_devices_.clear();
+ for (size_t i = 0; i < devices.size(); ++i) {
+ output_devices_.push_back(Device(devices[i], i));
+ }
+ SignalDevicesChange();
+ }
+ void SetVideoCaptureDevices(const std::vector<std::string>& devices) {
+ vidcap_devices_.clear();
+ for (size_t i = 0; i < devices.size(); ++i) {
+ vidcap_devices_.push_back(Device(devices[i], i));
+ }
+ SignalDevicesChange();
+ }
+ virtual bool GetVideoCaptureDevice(const std::string& name,
+ Device* out) {
+ if (vidcap_devices_.empty())
+ return false;
+
+ // If the name is empty, return the default device.
+ if (name.empty() || name == kDefaultDeviceName) {
+ *out = vidcap_devices_[0];
+ return true;
+ }
+
+ return FindDeviceByName(vidcap_devices_, name, out);
+ }
+ bool GetAudioDevice(bool is_input, const std::string& name,
+ Device* out) {
+ // If the name is empty, return the default device.
+ if (name.empty() || name == kDefaultDeviceName) {
+ *out = Device(name, -1);
+ return true;
+ }
+
+ return FindDeviceByName((is_input ? input_devices_ : output_devices_),
+ name, out);
+ }
+ static bool FindDeviceByName(const std::vector<Device>& devices,
+ const std::string& name,
+ Device* out) {
+ for (std::vector<Device>::const_iterator it = devices.begin();
+ it != devices.end(); ++it) {
+ if (name == it->name) {
+ *out = *it;
+ return true;
+ }
+ }
+ return false;
+ }
+ private:
+ std::vector<Device> input_devices_;
+ std::vector<Device> output_devices_;
+ std::vector<Device> vidcap_devices_;
+};
+
+} // namespace cricket
+
+#endif // TALK_SESSION_PHONE_FAKEDEVICEMANAGER_H_
diff --git a/talk/session/phone/videocommon.h b/talk/session/phone/videocommon.h
index 27a2e90..995e800 100644
--- a/talk/session/phone/videocommon.h
+++ b/talk/session/phone/videocommon.h
@@ -65,6 +65,7 @@
FOURCC_YUY2 = FOURCC('Y', 'U', 'Y', '2'),
FOURCC_UYVY = FOURCC('U', 'Y', 'V', 'Y'),
FOURCC_M420 = FOURCC('M', '4', '2', '0'),
+ FOURCC_Q420 = FOURCC('Q', '4', '2', '0'),
FOURCC_24BG = FOURCC('2', '4', 'B', 'G'),
FOURCC_ABGR = FOURCC('A', 'B', 'G', 'R'),
FOURCC_BGRA = FOURCC('B', 'G', 'R', 'A'),
diff --git a/talk/session/phone/webrtcvideoengine.h b/talk/session/phone/webrtcvideoengine.h
index dc69a83..bb6b9c8 100644
--- a/talk/session/phone/webrtcvideoengine.h
+++ b/talk/session/phone/webrtcvideoengine.h
@@ -36,7 +36,7 @@
#include "talk/session/phone/channel.h"
#include "talk/session/phone/webrtccommon.h"
#ifdef WEBRTC_RELATIVE_PATH
-#include "voice_engine/main/interface/vie_base.h"
+#include "video_engine/main/interface/vie_base.h"
#else
#include "third_party/webrtc/files/include/vie_base.h"
#endif // WEBRTC_RELATIVE_PATH
diff --git a/talk/session/phone/webrtcvoiceengine.cc b/talk/session/phone/webrtcvoiceengine.cc
index 561b025..9135fb3 100644
--- a/talk/session/phone/webrtcvoiceengine.cc
+++ b/talk/session/phone/webrtcvoiceengine.cc
@@ -243,7 +243,7 @@
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
for (int i = 0; i < ncodecs; ++i) {
webrtc::CodecInst voe_codec;
- if (voe_wrapper_->codec()->GetCodec(i, voe_codec) >= 0) {
+ if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
// Skip uncompressed formats.
if (_stricmp(voe_codec.plname, kL16CodecName) == 0) {
continue;
@@ -729,7 +729,7 @@
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
for (int i = 0; i < ncodecs; ++i) {
webrtc::CodecInst voe_codec;
- if (voe_wrapper_->codec()->GetCodec(i, voe_codec) >= 0) {
+ if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels, 0);
// Allow arbitrary rates for ISAC to be specified.
@@ -836,11 +836,12 @@
static const char* kTracesToIgnore[] = {
"\tfailed to GetReportBlockInformation",
"GetRecCodec() failed to get received codec",
- "GetRemoteRTCPData() failed to retrieve sender info for remote side",
- "GetRTPStatistics() failed to measure RTT since no RTP packets have been received yet", // NOLINT
+ "GetRemoteRTCPData() failed to measure statistics dueto lack of received RTP and/or RTCP packets", // NOLINT
+ "GetRemoteRTCPData() failed to retrieve sender info for remoteside",
+ "GetRTPStatistics() failed to measure RTT since noRTP packets have been received yet", // NOLINT
"GetRTPStatistics() failed to read RTP statistics from the RTP/RTCP module",
- "GetRTPStatistics() failed to retrieve RTT from the RTP/RTCP module",
- "RTCPReceiver::SenderInfoReceived No received SR",
+ "GetRTPStatistics() failed to retrieve RTT fromthe RTP/RTCP module",
+ "webrtc::RTCPReceiver::SenderInfoReceived No received SR",
"StatisticsRTP() no statisitics availble",
NULL
};
@@ -1033,14 +1034,7 @@
SetSendSsrc(talk_base::CreateRandomNonZeroId());
// Reset all recv codecs; they will be enabled via SetRecvCodecs.
- int ncodecs = engine->voe()->codec()->NumOfCodecs();
- for (int i = 0; i < ncodecs; ++i) {
- webrtc::CodecInst voe_codec;
- if (engine->voe()->codec()->GetCodec(i, voe_codec) >= 0) {
- voe_codec.pltype = -1;
- engine->voe()->codec()->SetRecPayloadType(voe_channel(), voe_codec);
- }
- }
+ ResetRecvCodecs(voe_channel());
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
@@ -1437,6 +1431,24 @@
return false;
}
+ // Use the same recv payload types as our default channel.
+ ResetRecvCodecs(channel);
+ int ncodecs = engine()->voe()->codec()->NumOfCodecs();
+ for (int i = 0; i < ncodecs; ++i) {
+ webrtc::CodecInst voe_codec;
+ if (engine()->voe()->codec()->GetCodec(i, voe_codec) != -1) {
+ voe_codec.rate = 0; // Needed to make GetRecPayloadType work for ISAC
+ if (engine()->voe()->codec()->GetRecPayloadType(
+ voe_channel(), voe_codec) != -1) {
+ if (engine()->voe()->codec()->SetRecPayloadType(
+ channel, voe_codec) == -1) {
+ LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+ return false;
+ }
+ }
+ }
+ }
+
if (mux_channels_.empty() && playout_) {
// This is the first stream in a multi user meeting. We can now
// disable playback of the default stream. This since the default
@@ -1950,6 +1962,22 @@
return true;
}
+bool WebRtcVoiceMediaChannel::ResetRecvCodecs(int channel) {
+ int ncodecs = engine()->voe()->codec()->NumOfCodecs();
+ for (int i = 0; i < ncodecs; ++i) {
+ webrtc::CodecInst voe_codec;
+ if (engine()->voe()->codec()->GetCodec(i, voe_codec) != -1) {
+ voe_codec.pltype = -1;
+ if (engine()->voe()->codec()->SetRecPayloadType(
+ channel, voe_codec) == -1) {
+ LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+ return false;
+ }
+ }
+ }
+ return true;
+}
+
bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) {
if (playout) {
LOG(LS_INFO) << "Starting playout for channel #" << channel;
diff --git a/talk/session/phone/webrtcvoiceengine.h b/talk/session/phone/webrtcvoiceengine.h
index 2f2ab13..66c9785 100644
--- a/talk/session/phone/webrtcvoiceengine.h
+++ b/talk/session/phone/webrtcvoiceengine.h
@@ -298,6 +298,7 @@
const std::vector<AudioCodec>& all_codecs,
webrtc::CodecInst* send_codec);
bool EnableRtcp(int channel);
+ bool ResetRecvCodecs(int channel);
bool SetPlayout(int channel, bool playout);
static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
static Error WebRtcErrorToChannelError(int err_code);
diff --git a/talk/session/phone/webrtcvoiceengine_unittest.cc b/talk/session/phone/webrtcvoiceengine_unittest.cc
index 768f1fd..f4e437f 100644
--- a/talk/session/phone/webrtcvoiceengine_unittest.cc
+++ b/talk/session/phone/webrtcvoiceengine_unittest.cc
@@ -244,6 +244,32 @@
EXPECT_FALSE(channel_->SetRecvCodecs(codecs));
}
+// Test that changes to recv codecs are applied to all streams.
+TEST_F(WebRtcVoiceEngineTest, SetRecvCodecsWithMultipleStreams) {
+ EXPECT_TRUE(SetupEngine());
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kIsacCodec);
+ codecs.push_back(kPcmuCodec);
+ codecs.push_back(kTelephoneEventCodec);
+ codecs[0].id = 106; // collide with existing telephone-event
+ codecs[2].id = 126;
+ EXPECT_TRUE(channel_->SetRecvCodecs(codecs));
+ EXPECT_TRUE(channel_->AddStream(2));
+ int channel_num2 = voe_.GetLastChannel();
+ webrtc::CodecInst gcodec;
+ talk_base::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname), "ISAC");
+ gcodec.plfreq = 16000;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+ EXPECT_EQ(106, gcodec.pltype);
+ EXPECT_STREQ("ISAC", gcodec.plname);
+ talk_base::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname),
+ "telephone-event");
+ gcodec.plfreq = 8000;
+ EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+ EXPECT_EQ(126, gcodec.pltype);
+ EXPECT_STREQ("telephone-event", gcodec.plname);
+}
+
// Test that we apply codecs properly.
TEST_F(WebRtcVoiceEngineTest, SetSendCodecs) {
EXPECT_TRUE(SetupEngine());
@@ -584,6 +610,64 @@
EXPECT_FALSE(voe_.GetPlayout(channel_num));
}
+// Test that we can add and remove streams, and do proper send/playout.
+// We can receive on multiple streams, but will only send on one.
+TEST_F(WebRtcVoiceEngineTest, SendAndPlayoutWithMultipleStreams) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num1 = voe_.GetLastChannel();
+
+ // Start playout on the default channel.
+ EXPECT_TRUE(channel_->SetPlayout(true));
+ EXPECT_TRUE(voe_.GetPlayout(channel_num1));
+
+ // Adding another stream should disable playout on the default channel.
+ EXPECT_TRUE(channel_->AddStream(2));
+ int channel_num2 = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kPcmuCodec);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
+ EXPECT_TRUE(voe_.GetSend(channel_num1));
+ EXPECT_FALSE(voe_.GetSend(channel_num2));
+
+ // Make sure only the new channel is played out.
+ EXPECT_FALSE(voe_.GetPlayout(channel_num1));
+ EXPECT_TRUE(voe_.GetPlayout(channel_num2));
+
+ // Adding yet another stream should have stream 2 and 3 enabled for playout.
+ EXPECT_TRUE(channel_->AddStream(3));
+ int channel_num3 = voe_.GetLastChannel();
+ EXPECT_FALSE(voe_.GetPlayout(channel_num1));
+ EXPECT_TRUE(voe_.GetPlayout(channel_num2));
+ EXPECT_TRUE(voe_.GetPlayout(channel_num3));
+ EXPECT_FALSE(voe_.GetSend(channel_num3));
+
+ // Stop sending.
+ EXPECT_TRUE(channel_->SetSend(cricket::SEND_NOTHING));
+ EXPECT_FALSE(voe_.GetSend(channel_num1));
+ EXPECT_FALSE(voe_.GetSend(channel_num2));
+ EXPECT_FALSE(voe_.GetSend(channel_num3));
+
+ // Stop playout.
+ EXPECT_TRUE(channel_->SetPlayout(false));
+ EXPECT_FALSE(voe_.GetPlayout(channel_num1));
+ EXPECT_FALSE(voe_.GetPlayout(channel_num2));
+ EXPECT_FALSE(voe_.GetPlayout(channel_num3));
+
+ // Restart playout and make sure the default channel still is not played out.
+ EXPECT_TRUE(channel_->SetPlayout(true));
+ EXPECT_FALSE(voe_.GetPlayout(channel_num1));
+ EXPECT_TRUE(voe_.GetPlayout(channel_num2));
+ EXPECT_TRUE(voe_.GetPlayout(channel_num3));
+
+ // Now remove the new streams and verify that the default channel is
+ // played out again.
+ EXPECT_TRUE(channel_->RemoveStream(3));
+ EXPECT_TRUE(channel_->RemoveStream(2));
+
+ EXPECT_TRUE(voe_.GetPlayout(channel_num1));
+}
+
// Test that we can set the devices to use.
TEST_F(WebRtcVoiceEngineTest, SetDevices) {
EXPECT_TRUE(SetupEngine());
@@ -758,73 +842,6 @@
EXPECT_EQ(0x99U, send_ssrc);
}
-// Test that we can properly receive packets.
-TEST_F(WebRtcVoiceEngineTest, Recv) {
- EXPECT_TRUE(SetupEngine());
- int channel_num = voe_.GetLastChannel();
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
- EXPECT_TRUE(voe_.CheckPacket(channel_num, kPcmuFrame,
- sizeof(kPcmuFrame)));
-}
-
-// Test that we can add and remove streams, and do proper send/playout.
-// We can receive on multiple streams, but will only send on one.
-TEST_F(WebRtcVoiceEngineTest, SendAndPlayoutWithMultipleStreams) {
- EXPECT_TRUE(SetupEngine());
- int channel_num1 = voe_.GetLastChannel();
-
- // Start playout on the default channel.
- EXPECT_TRUE(channel_->SetPlayout(true));
- EXPECT_TRUE(voe_.GetPlayout(channel_num1));
-
- // Adding another stream should disable playout on the default channel.
- EXPECT_TRUE(channel_->AddStream(2));
- int channel_num2 = voe_.GetLastChannel();
- std::vector<cricket::AudioCodec> codecs;
- codecs.push_back(kPcmuCodec);
- EXPECT_TRUE(channel_->SetSendCodecs(codecs));
- EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
- EXPECT_TRUE(voe_.GetSend(channel_num1));
- EXPECT_FALSE(voe_.GetSend(channel_num2));
-
- // Make sure only the new channel is played out.
- EXPECT_FALSE(voe_.GetPlayout(channel_num1));
- EXPECT_TRUE(voe_.GetPlayout(channel_num2));
-
- // Adding yet another stream should have stream 2 and 3 enabled for playout.
- EXPECT_TRUE(channel_->AddStream(3));
- int channel_num3 = voe_.GetLastChannel();
- EXPECT_FALSE(voe_.GetPlayout(channel_num1));
- EXPECT_TRUE(voe_.GetPlayout(channel_num2));
- EXPECT_TRUE(voe_.GetPlayout(channel_num3));
- EXPECT_FALSE(voe_.GetSend(channel_num3));
-
- // Stop sending.
- EXPECT_TRUE(channel_->SetSend(cricket::SEND_NOTHING));
- EXPECT_FALSE(voe_.GetSend(channel_num1));
- EXPECT_FALSE(voe_.GetSend(channel_num2));
- EXPECT_FALSE(voe_.GetSend(channel_num3));
-
- // Stop playout.
- EXPECT_TRUE(channel_->SetPlayout(false));
- EXPECT_FALSE(voe_.GetPlayout(channel_num1));
- EXPECT_FALSE(voe_.GetPlayout(channel_num2));
- EXPECT_FALSE(voe_.GetPlayout(channel_num3));
-
- // Restart playout and make sure the default channel still is not played out.
- EXPECT_TRUE(channel_->SetPlayout(true));
- EXPECT_FALSE(voe_.GetPlayout(channel_num1));
- EXPECT_TRUE(voe_.GetPlayout(channel_num2));
- EXPECT_TRUE(voe_.GetPlayout(channel_num3));
-
- // Now remove the new streams and verify that the default channel is
- // played out again.
- EXPECT_TRUE(channel_->RemoveStream(3));
- EXPECT_TRUE(channel_->RemoveStream(2));
-
- EXPECT_TRUE(voe_.GetPlayout(channel_num1));
-}
-
// Test that we can set the outgoing SSRC properly with multiple streams.
TEST_F(WebRtcVoiceEngineTest, SetSendSsrcWithMultipleStreams) {
EXPECT_TRUE(SetupEngine());
@@ -839,11 +856,13 @@
EXPECT_EQ(0x99U, send_ssrc);
}
-// Test that we properly handle failures to add a stream.
-TEST_F(WebRtcVoiceEngineTest, AddStreamFail) {
+// Test that we can properly receive packets.
+TEST_F(WebRtcVoiceEngineTest, Recv) {
EXPECT_TRUE(SetupEngine());
- voe_.set_fail_create_channel(true);
- EXPECT_FALSE(channel_->AddStream(2));
+ int channel_num = voe_.GetLastChannel();
+ DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ EXPECT_TRUE(voe_.CheckPacket(channel_num, kPcmuFrame,
+ sizeof(kPcmuFrame)));
}
// Test that we can properly receive packets on multiple streams.
@@ -888,6 +907,13 @@
EXPECT_TRUE(channel_->RemoveStream(1));
}
+// Test that we properly handle failures to add a stream.
+TEST_F(WebRtcVoiceEngineTest, AddStreamFail) {
+ EXPECT_TRUE(SetupEngine());
+ voe_.set_fail_create_channel(true);
+ EXPECT_FALSE(channel_->AddStream(2));
+}
+
// Test that we properly clean up any streams that were added, even if
// not explicitly removed.
TEST_F(WebRtcVoiceEngineTest, StreamCleanup) {
