admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Wire up packet_id / send time callbacks to webrtc via libjingle.

Browse Source 
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1363573002

Cr-Commit-Position: refs/heads/master@{#10289}
This commit is contained in:
stefan 2015-10-15 07:26:07 -07:00 committed by Commit bot
parent 543b6ca30a
commit c1aeaf0dc3
63 changed files with 560 additions and 206 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

55
talk/app/webrtc/fakemediacontroller.h Normal file
View File

@ -0,0 +1,55 @@
/*
* libjingle
* Copyright 2015 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_APP_WEBRTC_FAKEMEDIACONTROLLER_H_
#define TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_
#include "talk/app/webrtc/mediacontroller.h"
#include "webrtc/base/checks.h"
namespace cricket {
class FakeMediaController : public webrtc::MediaControllerInterface {
public:
explicit FakeMediaController(cricket::ChannelManager* channel_manager,
webrtc::Call* call)
: channel_manager_(channel_manager), call_(call) {
RTC_DCHECK(nullptr != channel_manager_);
RTC_DCHECK(nullptr != call_);
}
~FakeMediaController() override {}
webrtc::Call* call_w() override { return call_; }
cricket::ChannelManager* channel_manager() const override {
return channel_manager_;
}
private:
cricket::ChannelManager* channel_manager_;
webrtc::Call* call_;
};
} // namespace cricket
#endif // TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_

25
talk/app/webrtc/mediacontroller.cc
View File

@ -27,6 +27,7 @@
#include "talk/app/webrtc/mediacontroller.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/call.h"
@ -37,14 +38,16 @@ const int kMinBandwidthBps = 30000;
const int kStartBandwidthBps = 300000;
const int kMaxBandwidthBps = 2000000;
class MediaController : public webrtc::MediaControllerInterface {
class MediaController : public webrtc::MediaControllerInterface,
public sigslot::has_slots<> {
public:
MediaController(rtc::Thread* worker_thread,
webrtc::VoiceEngine* voice_engine)
: worker_thread_(worker_thread) {
cricket::ChannelManager* channel_manager)
: worker_thread_(worker_thread), channel_manager_(channel_manager) {
RTC_DCHECK(nullptr != worker_thread);
worker_thread_->Invoke<void>(
rtc::Bind(&MediaController::Construct_w, this, voice_engine));
rtc::Bind(&MediaController::Construct_w, this,
channel_manager_->media_engine()->GetVoE()));
}
~MediaController() override {
worker_thread_->Invoke<void>(
@ -56,6 +59,10 @@ class MediaController : public webrtc::MediaControllerInterface {
return call_.get();
}
cricket::ChannelManager* channel_manager() const override {
return channel_manager_;
}
private:
void Construct_w(webrtc::VoiceEngine* voice_engine) {
RTC_DCHECK(worker_thread_->IsCurrent());
@ -68,10 +75,11 @@ class MediaController : public webrtc::MediaControllerInterface {
}
void Destruct_w() {
RTC_DCHECK(worker_thread_->IsCurrent());
call_.reset(nullptr);
call_.reset();
}
rtc::Thread* worker_thread_;
rtc::Thread* const worker_thread_;
cricket::ChannelManager* const channel_manager_;
rtc::scoped_ptr<webrtc::Call> call_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController);
@ -81,7 +89,8 @@ class MediaController : public webrtc::MediaControllerInterface {
namespace webrtc {
MediaControllerInterface* MediaControllerInterface::Create(
rtc::Thread* worker_thread, webrtc::VoiceEngine* voice_engine) {
return new MediaController(worker_thread, voice_engine);
rtc::Thread* worker_thread,
cricket::ChannelManager* channel_manager) {
return new MediaController(worker_thread, channel_manager);
}
} // namespace webrtc

10
talk/app/webrtc/mediacontroller.h
View File

@ -30,6 +30,10 @@
#include "webrtc/base/thread.h"
namespace cricket {
class ChannelManager;
} // namespace cricket
namespace webrtc {
class Call;
class VoiceEngine;
@ -38,11 +42,13 @@ class VoiceEngine;
// in the future will create and own RtpSenders and RtpReceivers.
class MediaControllerInterface {
public:
static MediaControllerInterface* Create(rtc::Thread* worker_thread,
webrtc::VoiceEngine* voice_engine);
static MediaControllerInterface* Create(
rtc::Thread* worker_thread,
cricket::ChannelManager* channel_manager);
virtual ~MediaControllerInterface() {}
virtual webrtc::Call* call_w() = 0;
virtual cricket::ChannelManager* channel_manager() const = 0;
};
} // namespace webrtc

10
talk/app/webrtc/peerconnection.cc
View File

@ -630,11 +630,13 @@ bool PeerConnection::Initialize(
// No step delay is used while allocating ports.
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
remote_stream_factory_.reset(new RemoteMediaStreamFactory(
factory_->signaling_thread(), factory_->channel_manager()));
media_controller_.reset(factory_->CreateMediaController());
session_.reset(new WebRtcSession(
factory_->channel_manager(), factory_->signaling_thread(),
remote_stream_factory_.reset(new RemoteMediaStreamFactory(
factory_->signaling_thread(), media_controller_->channel_manager()));
session_.reset(
new WebRtcSession(media_controller_.get(), factory_->signaling_thread(),
factory_->worker_thread(), port_allocator_.get()));
stats_.reset(new StatsCollector(this));

1
talk/app/webrtc/peerconnection.h
View File

@ -361,6 +361,7 @@ class PeerConnection : public PeerConnectionInterface,
IceGatheringState ice_gathering_state_;
rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
rtc::scoped_ptr<MediaControllerInterface> media_controller_;
// Streams added via AddStream.
rtc::scoped_refptr<StreamCollection> local_streams_;

6
talk/app/webrtc/peerconnectionfactory.cc
View File

@ -279,9 +279,11 @@ PeerConnectionFactory::CreateAudioTrack(const std::string& id,
return AudioTrackProxy::Create(signaling_thread_, track);
}
cricket::ChannelManager* PeerConnectionFactory::channel_manager() {
webrtc::MediaControllerInterface* PeerConnectionFactory::CreateMediaController()
const {
RTC_DCHECK(signaling_thread_->IsCurrent());
return channel_manager_.get();
return MediaControllerInterface::Create(worker_thread_,
channel_manager_.get());
}
rtc::Thread* PeerConnectionFactory::signaling_thread() {

3
talk/app/webrtc/peerconnectionfactory.h
View File

@ -31,6 +31,7 @@
#include <string>
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "talk/app/webrtc/mediacontroller.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/session/media/channelmanager.h"
@ -80,7 +81,7 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
bool StartAecDump(rtc::PlatformFile file) override;
virtual cricket::ChannelManager* channel_manager();
virtual webrtc::MediaControllerInterface* CreateMediaController() const;
virtual rtc::Thread* signaling_thread();
virtual rtc::Thread* worker_thread();
const Options& options() const { return options_; }

62
talk/app/webrtc/statscollector_unittest.cc
View File

@ -84,8 +84,8 @@ const uint32_t kSsrcOfTrack = 1234;
class MockWebRtcSession : public webrtc::WebRtcSession {
public:
explicit MockWebRtcSession(cricket::ChannelManager* channel_manager)
: WebRtcSession(channel_manager,
explicit MockWebRtcSession(webrtc::MediaControllerInterface* media_controller)
: WebRtcSession(media_controller,
rtc::Thread::Current(),
rtc::Thread::Current(),
nullptr) {}
@ -506,7 +506,10 @@ class StatsCollectorTest : public testing::Test {
: media_engine_(new cricket::FakeMediaEngine()),
channel_manager_(
new cricket::ChannelManager(media_engine_, rtc::Thread::Current())),
session_(channel_manager_.get()) {
media_controller_(
webrtc::MediaControllerInterface::Create(rtc::Thread::Current(),
channel_manager_.get())),
session_(media_controller_.get()) {
// By default, we ignore session GetStats calls.
EXPECT_CALL(session_, GetTransportStats(_)).WillRepeatedly(Return(false));
// Add default returns for mock classes.
@ -760,6 +763,7 @@ class StatsCollectorTest : public testing::Test {
cricket::FakeMediaEngine* media_engine_;
rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
rtc::scoped_ptr<webrtc::MediaControllerInterface> media_controller_;
MockWebRtcSession session_;
MockPeerConnection pc_;
FakeDataChannelProvider data_channel_provider_;
@ -825,8 +829,8 @@ TEST_F(StatsCollectorTest, BytesCounterHandles64Bits) {
Return(true)));
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, kVideoChannelName, false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, kVideoChannelName, false);
StatsReports reports; // returned values.
cricket::VideoSenderInfo video_sender_info;
cricket::VideoMediaInfo stats_read;
@ -871,8 +875,8 @@ TEST_F(StatsCollectorTest, BandwidthEstimationInfoIsReported) {
Return(true)));
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, kVideoChannelName, false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, kVideoChannelName, false);
StatsReports reports; // returned values.
cricket::VideoSenderInfo video_sender_info;
@ -946,8 +950,8 @@ TEST_F(StatsCollectorTest, TrackObjectExistsWithoutUpdateStats) {
StatsCollectorForTest stats(&pc_);
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, "video", false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, "video", false);
AddOutgoingVideoTrackStats();
stats.AddStream(stream_);
@ -982,8 +986,8 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
Return(true)));
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, kVideoChannelName, false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, kVideoChannelName, false);
AddOutgoingVideoTrackStats();
stats.AddStream(stream_);
@ -1046,8 +1050,8 @@ TEST_F(StatsCollectorTest, TransportObjectLinkedFromSsrcObject) {
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
// The transport_name known by the video channel.
const std::string kVcName("vcname");
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, kVcName, false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, kVcName, false);
AddOutgoingVideoTrackStats();
stats.AddStream(stream_);
@ -1104,8 +1108,8 @@ TEST_F(StatsCollectorTest, RemoteSsrcInfoIsAbsent) {
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
// The transport_name known by the video channel.
const std::string kVcName("vcname");
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, kVcName, false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, kVcName, false);
AddOutgoingVideoTrackStats();
stats.AddStream(stream_);
@ -1130,8 +1134,8 @@ TEST_F(StatsCollectorTest, RemoteSsrcInfoIsPresent) {
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
// The transport_name known by the video channel.
const std::string kVcName("vcname");
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, kVcName, false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, kVcName, false);
AddOutgoingVideoTrackStats();
stats.AddStream(stream_);
@ -1185,8 +1189,8 @@ TEST_F(StatsCollectorTest, ReportsFromRemoteTrack) {
Return(true)));
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
cricket::VideoChannel video_channel(rtc::Thread::Current(),
media_channel, NULL, kVideoChannelName, false);
cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
nullptr, kVideoChannelName, false);
AddIncomingVideoTrackStats();
stats.AddStream(stream_);
@ -1494,8 +1498,8 @@ TEST_F(StatsCollectorTest, GetStatsFromLocalAudioTrack) {
MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
// The transport_name known by the voice channel.
const std::string kVcName("vcname");
cricket::VoiceChannel voice_channel(rtc::Thread::Current(),
media_engine_, media_channel, NULL, kVcName, false);
cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
media_channel, nullptr, kVcName, false);
AddOutgoingAudioTrackStats();
stats.AddStream(stream_);
stats.AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
@ -1529,8 +1533,8 @@ TEST_F(StatsCollectorTest, GetStatsFromRemoteStream) {
MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
// The transport_name known by the voice channel.
const std::string kVcName("vcname");
cricket::VoiceChannel voice_channel(rtc::Thread::Current(),
media_engine_, media_channel, NULL, kVcName, false);
cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
media_channel, nullptr, kVcName, false);
AddIncomingAudioTrackStats();
stats.AddStream(stream_);
@ -1558,8 +1562,8 @@ TEST_F(StatsCollectorTest, GetStatsAfterRemoveAudioStream) {
MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
// The transport_name known by the voice channel.
const std::string kVcName("vcname");
cricket::VoiceChannel voice_channel(rtc::Thread::Current(),
media_engine_, media_channel, NULL, kVcName, false);
cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
media_channel, nullptr, kVcName, false);
AddOutgoingAudioTrackStats();
stats.AddStream(stream_);
stats.AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
@ -1619,8 +1623,8 @@ TEST_F(StatsCollectorTest, LocalAndRemoteTracksWithSameSsrc) {
MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
// The transport_name known by the voice channel.
const std::string kVcName("vcname");
cricket::VoiceChannel voice_channel(rtc::Thread::Current(),
media_engine_, media_channel, NULL, kVcName, false);
cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
media_channel, nullptr, kVcName, false);
// Create a local stream with a local audio track and adds it to the stats.
AddOutgoingAudioTrackStats();
@ -1706,8 +1710,8 @@ TEST_F(StatsCollectorTest, TwoLocalTracksWithSameSsrc) {
MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
// The transport_name known by the voice channel.
const std::string kVcName("vcname");
cricket::VoiceChannel voice_channel(rtc::Thread::Current(),
media_engine_, media_channel, NULL, kVcName, false);
cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
media_channel, nullptr, kVcName, false);
// Create a local stream with a local audio track and adds it to the stats.
AddOutgoingAudioTrackStats();

4
talk/app/webrtc/test/fakedatachannelprovider.h
View File

@ -25,6 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_TEST_FAKEDATACHANNELPROVIDER_H_
#define TALK_APP_WEBRTC_TEST_FAKEDATACHANNELPROVIDER_H_
#include "talk/app/webrtc/datachannel.h"
class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
@ -155,3 +158,4 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
std::set<uint32_t> send_ssrcs_;
std::set<uint32_t> recv_ssrcs_;
};
#endif // TALK_APP_WEBRTC_TEST_FAKEDATACHANNELPROVIDER_H_

26
talk/app/webrtc/webrtcsession.cc
View File

@ -51,7 +51,9 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call.h"
#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/p2p/base/transportchannel.h"
using cricket::ContentInfo;
using cricket::ContentInfos;
@ -529,7 +531,7 @@ class IceRestartAnswerLatch {
bool ice_restart_;
};
WebRtcSession::WebRtcSession(cricket::ChannelManager* channel_manager,
WebRtcSession::WebRtcSession(webrtc::MediaControllerInterface* media_controller,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,
cricket::PortAllocator* port_allocator)
@ -543,7 +545,8 @@ WebRtcSession::WebRtcSession(cricket::ChannelManager* channel_manager,
transport_controller_(new cricket::TransportController(signaling_thread,
worker_thread,
port_allocator)),
channel_manager_(channel_manager),
media_controller_(media_controller),
channel_manager_(media_controller_->channel_manager()),
ice_observer_(NULL),
ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
ice_connection_receiving_(true),
@ -763,9 +766,6 @@ bool WebRtcSession::Initialize(
cricket::PORTALLOCATOR_ENABLE_LOCALHOST_CANDIDATE);
}
media_controller_.reset(MediaControllerInterface::Create(
worker_thread(), channel_manager_->media_engine()->GetVoE()));
return true;
}
@ -1844,7 +1844,7 @@ bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content) {
voice_channel_.reset(channel_manager_->CreateVoiceChannel(
media_controller_.get(), transport_controller_.get(), content->name, true,
media_controller_, transport_controller_.get(), content->name, true,
audio_options_));
if (!voice_channel_) {
return false;
@ -1854,12 +1854,14 @@ bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content) {
this, &WebRtcSession::OnDtlsSetupFailure);
SignalVoiceChannelCreated();
voice_channel_->transport_channel()->SignalSentPacket.connect(
this, &WebRtcSession::OnSentPacket_w);
return true;
}
bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content) {
video_channel_.reset(channel_manager_->CreateVideoChannel(
media_controller_.get(), transport_controller_.get(), content->name, true,
media_controller_, transport_controller_.get(), content->name, true,
video_options_));
if (!video_channel_) {
return false;
@ -1869,6 +1871,8 @@ bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content) {
this, &WebRtcSession::OnDtlsSetupFailure);
SignalVideoChannelCreated();
video_channel_->transport_channel()->SignalSentPacket.connect(
this, &WebRtcSession::OnSentPacket_w);
return true;
}
@ -1889,6 +1893,8 @@ bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content) {
this, &WebRtcSession::OnDtlsSetupFailure);
SignalDataChannelCreated();
data_channel_->transport_channel()->SignalSentPacket.connect(
this, &WebRtcSession::OnSentPacket_w);
return true;
}
@ -2205,4 +2211,10 @@ void WebRtcSession::ReportNegotiatedCiphers(
}
}
void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
const rtc::SentPacket& sent_packet) {
RTC_DCHECK(worker_thread()->IsCurrent());
media_controller_->call_w()->OnSentPacket(sent_packet);
}
} // namespace webrtc

7
talk/app/webrtc/webrtcsession.h
View File

@ -151,7 +151,7 @@ class WebRtcSession : public AudioProviderInterface,
ERROR_TRANSPORT = 2, // transport error of some kind
};
WebRtcSession(cricket::ChannelManager* channel_manager,
WebRtcSession(webrtc::MediaControllerInterface* media_controller,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,
cricket::PortAllocator* port_allocator);
@ -458,6 +458,9 @@ class WebRtcSession : public AudioProviderInterface,
void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
void OnSentPacket_w(cricket::TransportChannel* channel,
const rtc::SentPacket& sent_packet);
rtc::Thread* const signaling_thread_;
rtc::Thread* const worker_thread_;
cricket::PortAllocator* const port_allocator_;
@ -470,7 +473,7 @@ class WebRtcSession : public AudioProviderInterface,
bool initial_offerer_ = false;
rtc::scoped_ptr<cricket::TransportController> transport_controller_;
rtc::scoped_ptr<MediaControllerInterface> media_controller_;
MediaControllerInterface* media_controller_;
rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
rtc::scoped_ptr<cricket::DataChannel> data_channel_;

77
talk/app/webrtc/webrtcsession_unittest.cc
View File

@ -28,6 +28,7 @@
#include <vector>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/fakemediacontroller.h"
#include "talk/app/webrtc/fakemetricsobserver.h"
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
@ -44,6 +45,7 @@
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/base/fakevideorenderer.h"
#include "talk/media/base/mediachannel.h"
#include "talk/media/webrtc/fakewebrtccall.h"
#include "webrtc/p2p/base/stunserver.h"
#include "webrtc/p2p/base/teststunserver.h"
#include "webrtc/p2p/base/testturnserver.h"
@ -245,12 +247,15 @@ class MockIceObserver : public webrtc::IceObserver {
class WebRtcSessionForTest : public webrtc::WebRtcSession {
public:
WebRtcSessionForTest(cricket::ChannelManager* cmgr,
WebRtcSessionForTest(webrtc::MediaControllerInterface* media_controller,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,
cricket::PortAllocator* port_allocator,
webrtc::IceObserver* ice_observer)
: WebRtcSession(cmgr, signaling_thread, worker_thread, port_allocator) {
: WebRtcSession(media_controller,
signaling_thread,
worker_thread,
port_allocator) {
RegisterIceObserver(ice_observer);
}
virtual ~WebRtcSessionForTest() {}
@ -362,18 +367,25 @@ class WebRtcSessionTest
WebRtcSessionTest()
: media_engine_(new cricket::FakeMediaEngine()),
data_engine_(new cricket::FakeDataEngine()),
channel_manager_(new cricket::ChannelManager(
media_engine_, data_engine_, new cricket::CaptureManager(),
channel_manager_(
new cricket::ChannelManager(media_engine_,
data_engine_,
new cricket::CaptureManager(),
rtc::Thread::Current())),
fake_call_(webrtc::Call::Config()),
media_controller_(
webrtc::MediaControllerInterface::Create(rtc::Thread::Current(),
channel_manager_.get())),
tdesc_factory_(new cricket::TransportDescriptionFactory()),
desc_factory_(new cricket::MediaSessionDescriptionFactory(
channel_manager_.get(), tdesc_factory_.get())),
desc_factory_(
new cricket::MediaSessionDescriptionFactory(channel_manager_.get(),
tdesc_factory_.get())),
pss_(new rtc::PhysicalSocketServer),
vss_(new rtc::VirtualSocketServer(pss_.get())),
fss_(new rtc::FirewallSocketServer(vss_.get())),
ss_scope_(fss_.get()),
stun_socket_addr_(rtc::SocketAddress(kStunAddrHost,
cricket::STUN_SERVER_PORT)),
stun_socket_addr_(
rtc::SocketAddress(kStunAddrHost, cricket::STUN_SERVER_PORT)),
stun_server_(cricket::TestStunServer::Create(Thread::Current(),
stun_socket_addr_)),
turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
@ -405,7 +417,7 @@ class WebRtcSessionTest
const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
ASSERT_TRUE(session_.get() == NULL);
session_.reset(new WebRtcSessionForTest(
channel_manager_.get(), rtc::Thread::Current(), rtc::Thread::Current(),
media_controller_.get(), rtc::Thread::Current(), rtc::Thread::Current(),
allocator_.get(), &observer_));
session_->SignalDataChannelOpenMessage.connect(
this, &WebRtcSessionTest::OnDataChannelOpenMessage);
@ -1226,8 +1238,7 @@ class WebRtcSessionTest
// -> Failed.
// The Gathering state should go: New -> Gathering -> Completed.
void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
LoopbackNetworkManager loopback_network_manager(this, config);
void SetupLoopbackCall() {
Init();
SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@ -1238,30 +1249,29 @@ class WebRtcSessionTest
EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
observer_.ice_connection_state_);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringGathering,
observer_.ice_gathering_state_,
kIceCandidatesTimeout);
observer_.ice_gathering_state_, kIceCandidatesTimeout);
EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
observer_.ice_gathering_state_,
kIceCandidatesTimeout);
observer_.ice_gathering_state_, kIceCandidatesTimeout);
std::string sdp;
offer->ToString(&sdp);
SessionDescriptionInterface* desc =
webrtc::CreateSessionDescription(
SessionDescriptionInterface* desc = webrtc::CreateSessionDescription(
JsepSessionDescription::kAnswer, sdp, nullptr);
ASSERT_TRUE(desc != NULL);
SetRemoteDescriptionWithoutError(desc);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
observer_.ice_connection_state_, kIceCandidatesTimeout);
// The ice connection state is "Connected" too briefly to catch in a test.
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
observer_.ice_connection_state_,
kIceCandidatesTimeout);
observer_.ice_connection_state_, kIceCandidatesTimeout);
}
void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
LoopbackNetworkManager loopback_network_manager(this, config);
SetupLoopbackCall();
config.VerifyBestConnectionAfterIceConverge(metrics_observer_);
// Adding firewall rule to block ping requests, which should cause
// transport channel failure.
@ -1300,6 +1310,25 @@ class WebRtcSessionTest
TestLoopbackCall(config);
}
void TestPacketOptions() {
media_controller_.reset(
new cricket::FakeMediaController(channel_manager_.get(), &fake_call_));
LoopbackNetworkConfiguration config;
LoopbackNetworkManager loopback_network_manager(this, config);
SetupLoopbackCall();
uint8_t test_packet[15] = {0};
rtc::PacketOptions options;
options.packet_id = 10;
media_engine_->GetVideoChannel(0)
->SendRtp(test_packet, sizeof(test_packet), options);
const int kPacketTimeout = 2000;
EXPECT_EQ_WAIT(fake_call_.last_sent_packet().packet_id, 10, kPacketTimeout);
EXPECT_GT(fake_call_.last_sent_packet().send_time_ms, -1);
}
// Adds CN codecs to FakeMediaEngine and MediaDescriptionFactory.
void AddCNCodecs() {
const cricket::AudioCodec kCNCodec1(102, "CN", 8000, 0, 1, 0);
@ -1406,6 +1435,8 @@ class WebRtcSessionTest
cricket::FakeMediaEngine* media_engine_;
cricket::FakeDataEngine* data_engine_;
rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
cricket::FakeCall fake_call_;
rtc::scoped_ptr<webrtc::MediaControllerInterface> media_controller_;
rtc::scoped_ptr<cricket::TransportDescriptionFactory> tdesc_factory_;
rtc::scoped_ptr<cricket::MediaSessionDescriptionFactory> desc_factory_;
rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
@ -4154,6 +4185,10 @@ TEST_F(WebRtcSessionTest, CreateOffersAndShutdown) {
}
}
TEST_F(WebRtcSessionTest, TestPacketOptionsAndOnPacketSent) {
TestPacketOptions();
}
// TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
// currently fails because upon disconnection and reconnection OnIceComplete is
// called more than once without returning to IceGatheringGathering.

3
talk/libjingle.gyp
View File

@ -46,6 +46,7 @@
'target_name': 'libjingle_peerconnection_so',
'type': 'shared_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'libjingle_peerconnection',
],
'sources': [
@ -432,8 +433,8 @@
'<(webrtc_root)/webrtc.gyp:webrtc',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(webrtc_root)/sound/sound.gyp:rtc_sound',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
'<(webrtc_root)/libjingle/xmpp/xmpp.gyp:rtc_xmpp',
'<(webrtc_root)/p2p/p2p.gyp:rtc_p2p',

3
talk/libjingle_tests.gyp
View File

@ -142,6 +142,7 @@
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle',
'libjingle.gyp:libjingle_peerconnection',
'libjingle.gyp:libjingle_p2p',
'libjingle_unittest_main',
],
@ -344,6 +345,7 @@
'includes': [ 'build/objc_app.gypi' ],
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'libjingle.gyp:libjingle_peerconnection_objc',
],
'sources': [
@ -375,6 +377,7 @@
'includes': [ 'build/objc_app.gypi' ],
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock',
'<(webrtc_root)/libjingle_examples.gyp:apprtc_signaling',
],

4
talk/media/base/constants.cc
View File

@ -124,6 +124,10 @@ const char kRtpVideoRotationHeaderExtension[] = "urn:3gpp:video-orientation";
const char kRtpVideoRotation6BitsHeaderExtensionForTesting[] =
"urn:3gpp:video-orientation:6";
const int kRtpTransportSequenceNumberHeaderExtensionDefaultId = 5;
const char kRtpTransportSequenceNumberHeaderExtension[] =
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
const int kNumDefaultUnsignalledVideoRecvStreams = 0;
const char kVp8CodecName[] = "VP8";

5
talk/media/base/constants.h
View File

@ -154,6 +154,11 @@ extern const char kRtpVideoRotationHeaderExtension[];
// We don't support 6 bit CVO. Added here for testing purpose.
extern const char kRtpVideoRotation6BitsHeaderExtensionForTesting[];
// Header extension for transport sequence number, see url for details:
// http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
extern const int kRtpTransportSequenceNumberHeaderExtensionDefaultId;
extern const char kRtpTransportSequenceNumberHeaderExtension[];
extern const int kNumDefaultUnsignalledVideoRecvStreams;
extern const char kVp8CodecName[];

6
talk/media/base/fakemediaengine.h
View File

@ -69,18 +69,18 @@ template <class Base> class RtpHelper : public Base {
const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
bool SendRtp(const void* data, int len) {
bool SendRtp(const void* data, int len, const rtc::PacketOptions& options) {
if (!sending_) {
return false;
}
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return Base::SendPacket(&packet);
return Base::SendPacket(&packet, options);
}
bool SendRtcp(const void* data, int len) {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return Base::SendRtcp(&packet);
return Base::SendRtcp(&packet, rtc::PacketOptions());
}
bool CheckRtp(const void* data, int len) {

4
talk/media/base/fakenetworkinterface.h
View File

@ -129,7 +129,7 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
protected:
virtual bool SendPacket(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
const rtc::PacketOptions& options) {
rtc::CritScope cs(&crit_);
uint32_t cur_ssrc = 0;
@ -155,7 +155,7 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
}
virtual bool SendRtcp(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
const rtc::PacketOptions& options) {
rtc::CritScope cs(&crit_);
rtcp_packets_.push_back(*packet);
if (!conf_) {

26
talk/media/base/mediachannel.h
View File

@ -504,12 +504,10 @@ class MediaChannel : public sigslot::has_slots<> {
class NetworkInterface {
public:
enum SocketType { ST_RTP, ST_RTCP };
virtual bool SendPacket(
rtc::Buffer* packet,
rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 0;
virtual bool SendRtcp(
rtc::Buffer* packet,
rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 0;
virtual bool SendPacket(rtc::Buffer* packet,
const rtc::PacketOptions& options) = 0;
virtual bool SendRtcp(rtc::Buffer* packet,
const rtc::PacketOptions& options) = 0;
virtual int SetOption(SocketType type, rtc::Socket::Option opt,
int option) = 0;
virtual ~NetworkInterface() {}
@ -553,12 +551,12 @@ class MediaChannel : public sigslot::has_slots<> {
}
// Base method to send packet using NetworkInterface.
bool SendPacket(rtc::Buffer* packet) {
return DoSendPacket(packet, false);
bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) {
return DoSendPacket(packet, false, options);
}
bool SendRtcp(rtc::Buffer* packet) {
return DoSendPacket(packet, true);
bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) {
return DoSendPacket(packet, true, options);
}
int SetOption(NetworkInterface::SocketType type,
@ -587,13 +585,15 @@ class MediaChannel : public sigslot::has_slots<> {
}
private:
bool DoSendPacket(rtc::Buffer* packet, bool rtcp) {
bool DoSendPacket(rtc::Buffer* packet,
bool rtcp,
const rtc::PacketOptions& options) {
rtc::CritScope cs(&network_interface_crit_);
if (!network_interface_)
return false;
return (!rtcp) ? network_interface_->SendPacket(packet) :
network_interface_->SendRtcp(packet);
return (!rtcp) ? network_interface_->SendPacket(packet, options)
: network_interface_->SendRtcp(packet, options);
}
// |network_interface_| can be accessed from the worker_thread and

2
talk/media/base/rtpdataengine.cc
View File

@ -359,7 +359,7 @@ bool RtpDataMediaChannel::SendData(
<< ", timestamp=" << header.timestamp
<< ", len=" << payload.size();
MediaChannel::SendPacket(&packet);
MediaChannel::SendPacket(&packet, rtc::PacketOptions());
send_limiter_->Use(packet_len, now);
if (result) {
*result = SDR_SUCCESS;

2
talk/media/sctp/sctpdataengine.cc
View File

@ -984,7 +984,7 @@ void SctpDataMediaChannel::OnPacketFromSctpToNetwork(
<< " even after adding " << kSctpOverhead
<< " extra SCTP overhead";
}
MediaChannel::SendPacket(buffer);
MediaChannel::SendPacket(buffer, rtc::PacketOptions());
}
bool SctpDataMediaChannel::SendQueuedStreamResets() {

4
talk/media/sctp/sctpdataengine_unittest.cc
View File

@ -64,7 +64,7 @@ class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
protected:
// Called to send raw packet down the wire (e.g. SCTP an packet).
virtual bool SendPacket(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
const rtc::PacketOptions& options) {
LOG(LS_VERBOSE) << "SctpFakeNetworkInterface::SendPacket";
// TODO(ldixon): Can/should we use Buffer.TransferTo here?
@ -93,7 +93,7 @@ class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
// TODO(ldixon): Refactor parent NetworkInterface class so these are not
// required. They are RTC specific and should be in an appropriate subclass.
virtual bool SendRtcp(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
const rtc::PacketOptions& options) {
LOG(LS_WARNING) << "Unsupported: SctpFakeNetworkInterface::SendRtcp.";
return false;
}

7
talk/media/webrtc/fakewebrtccall.cc
View File

@ -202,8 +202,7 @@ FakeCall::FakeCall(const webrtc::Call::Config& config)
: config_(config),
network_state_(webrtc::kNetworkUp),
num_created_send_streams_(0),
num_created_receive_streams_(0) {
}
num_created_receive_streams_(0) {}
FakeCall::~FakeCall() {
EXPECT_EQ(0u, video_send_streams_.size());
@ -367,4 +366,8 @@ void FakeCall::SetBitrateConfig(
void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
network_state_ = state;
}
void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
last_sent_packet_ = sent_packet;
}
} // namespace cricket

3
talk/media/webrtc/fakewebrtccall.h
View File

@ -164,6 +164,7 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
webrtc::NetworkState GetNetworkState() const;
int GetNumCreatedSendStreams() const;
int GetNumCreatedReceiveStreams() const;
@ -200,9 +201,11 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
void SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
void SignalNetworkState(webrtc::NetworkState state) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
webrtc::Call::Config config_;
webrtc::NetworkState network_state_;
rtc::SentPacket last_sent_packet_;
webrtc::Call::Stats stats_;
std::vector<FakeVideoSendStream*> video_send_streams_;
std::vector<FakeVideoReceiveStream*> video_receive_streams_;

11
talk/media/webrtc/webrtcvideoengine2.cc
View File

@ -557,6 +557,11 @@ WebRtcVideoEngine2::WebRtcVideoEngine2()
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
kRtpVideoRotationHeaderExtensionDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
rtp_header_extensions_.push_back(RtpHeaderExtension(
kRtpTransportSequenceNumberHeaderExtension,
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
}
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
@ -1651,12 +1656,14 @@ bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendPacket(&packet);
rtc::PacketOptions rtc_options;
rtc_options.packet_id = options.packet_id;
return MediaChannel::SendPacket(&packet, rtc_options);
}
bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendRtcp(&packet);
return MediaChannel::SendRtcp(&packet, rtc::PacketOptions());
}
void WebRtcVideoChannel2::StartAllSendStreams() {

56
talk/media/webrtc/webrtcvideoengine2_unittest.cc
View File

@ -40,6 +40,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/video_encoder.h"
namespace {
@ -108,9 +109,13 @@ void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config,
namespace cricket {
class WebRtcVideoEngine2Test : public ::testing::Test {
public:
WebRtcVideoEngine2Test() : WebRtcVideoEngine2Test(nullptr) {}
WebRtcVideoEngine2Test(WebRtcVoiceEngine* voice_engine)
: call_(webrtc::Call::Create(webrtc::Call::Config())),
WebRtcVideoEngine2Test() : WebRtcVideoEngine2Test("") {}
explicit WebRtcVideoEngine2Test(const char* field_trials)
: WebRtcVideoEngine2Test(nullptr, field_trials) {}
WebRtcVideoEngine2Test(WebRtcVoiceEngine* voice_engine,
const char* field_trials)
: override_field_trials_(field_trials),
call_(webrtc::Call::Create(webrtc::Call::Config())),
engine_() {
std::vector<VideoCodec> engine_codecs = engine_.codecs();
RTC_DCHECK(!engine_codecs.empty());
@ -144,6 +149,7 @@ class WebRtcVideoEngine2Test : public ::testing::Test {
cricket::WebRtcVideoDecoderFactory* decoder_factory,
const std::vector<VideoCodec>& codecs);
webrtc::test::ScopedFieldTrials override_field_trials_;
// Used in WebRtcVideoEngine2VoiceTest, but defined here so it's properly
// initialized when the constructor is called.
rtc::scoped_ptr<webrtc::Call> call_;
@ -258,6 +264,26 @@ TEST_F(WebRtcVideoEngine2Test, SupportsAbsoluteSenderTimeHeaderExtension) {
FAIL() << "Absolute Sender Time extension not in header-extension list.";
}
class WebRtcVideoEngine2WithSendSideBweTest : public WebRtcVideoEngine2Test {
public:
WebRtcVideoEngine2WithSendSideBweTest()
: WebRtcVideoEngine2Test("WebRTC-SendSideBwe/Enabled/") {}
};
TEST_F(WebRtcVideoEngine2WithSendSideBweTest,
SupportsTransportSequenceNumberHeaderExtension) {
std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
ASSERT_FALSE(extensions.empty());
for (size_t i = 0; i < extensions.size(); ++i) {
if (extensions[i].uri == kRtpTransportSequenceNumberHeaderExtension) {
EXPECT_EQ(kRtpTransportSequenceNumberHeaderExtensionDefaultId,
extensions[i].id);
return;
}
}
FAIL() << "Transport sequence number extension not in header-extension list.";
}
TEST_F(WebRtcVideoEngine2Test, SupportsVideoRotationHeaderExtension) {
std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
ASSERT_FALSE(extensions.empty());
@ -895,7 +921,9 @@ TEST_F(WebRtcVideoChannel2BaseTest, DISABLED_SendVp8HdAndReceiveAdaptedVp8Vga) {
class WebRtcVideoChannel2Test : public WebRtcVideoEngine2Test {
public:
WebRtcVideoChannel2Test() : last_ssrc_(0) {}
WebRtcVideoChannel2Test() : WebRtcVideoChannel2Test("") {}
explicit WebRtcVideoChannel2Test(const char* field_trials)
: WebRtcVideoEngine2Test(field_trials), last_ssrc_(0) {}
void SetUp() override {
fake_call_.reset(new FakeCall(webrtc::Call::Config()));
engine_.Init();
@ -1171,6 +1199,26 @@ TEST_F(WebRtcVideoChannel2Test, RecvAbsoluteSendTimeHeaderExtensions) {
webrtc::RtpExtension::kAbsSendTime);
}
class WebRtcVideoChannel2WithSendSideBweTest : public WebRtcVideoChannel2Test {
public:
WebRtcVideoChannel2WithSendSideBweTest()
: WebRtcVideoChannel2Test("WebRTC-SendSideBwe/Enabled/") {}
};
// Test support for transport sequence number header extension.
TEST_F(WebRtcVideoChannel2WithSendSideBweTest,
SendTransportSequenceNumberHeaderExtensions) {
TestSetSendRtpHeaderExtensions(
kRtpTransportSequenceNumberHeaderExtension,
webrtc::RtpExtension::kTransportSequenceNumber);
}
TEST_F(WebRtcVideoChannel2WithSendSideBweTest,
RecvTransportSequenceNumberHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(
kRtpTransportSequenceNumberHeaderExtension,
webrtc::RtpExtension::kTransportSequenceNumber);
}
// Test support for video rotation header extension.
TEST_F(WebRtcVideoChannel2Test, SendVideoRotationHeaderExtensions) {
TestSetSendRtpHeaderExtensions(kRtpVideoRotationHeaderExtension,

6
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -52,6 +52,7 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/interface/field_trial.h"
namespace cricket {
namespace {
@ -431,6 +432,11 @@ void WebRtcVoiceEngine::Construct() {
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
rtp_header_extensions_.push_back(RtpHeaderExtension(
kRtpTransportSequenceNumberHeaderExtension,
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
options_ = GetDefaultEngineOptions();
}

6
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -226,13 +226,15 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
const webrtc::PacketOptions& options) override {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return VoiceMediaChannel::SendPacket(&packet);
rtc::PacketOptions rtc_options;
rtc_options.packet_id = options.packet_id;
return VoiceMediaChannel::SendPacket(&packet, rtc_options);
}
bool SendRtcp(const uint8_t* data, size_t len) override {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return VoiceMediaChannel::SendRtcp(&packet);
return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
}
void OnError(int error);

40
talk/session/media/channel.cc
View File

@ -67,7 +67,7 @@ static void SafeSetError(const std::string& message, std::string* error_desc) {
struct PacketMessageData : public rtc::MessageData {
rtc::Buffer packet;
rtc::DiffServCodePoint dscp;
rtc::PacketOptions options;
};
struct ScreencastEventMessageData : public rtc::MessageData {
@ -423,13 +423,13 @@ bool BaseChannel::IsReadyToSend() const {
}
bool BaseChannel::SendPacket(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
return SendPacket(false, packet, dscp);
const rtc::PacketOptions& options) {
return SendPacket(false, packet, options);
}
bool BaseChannel::SendRtcp(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
return SendPacket(true, packet, dscp);
const rtc::PacketOptions& options) {
return SendPacket(true, packet, options);
}
int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
@ -498,8 +498,9 @@ bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
}
bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
bool BaseChannel::SendPacket(bool rtcp,
rtc::Buffer* packet,
const rtc::PacketOptions& options) {
// SendPacket gets called from MediaEngine, typically on an encoder thread.
// If the thread is not our worker thread, we will post to our worker
// so that the real work happens on our worker. This avoids us having to
@ -512,7 +513,7 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
PacketMessageData* data = new PacketMessageData;
data->packet = packet->Pass();
data->dscp = dscp;
data->options = options;
worker_thread_->Post(this, message_id, data);
return true;
}
@ -535,7 +536,8 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
return false;
}
rtc::PacketOptions options(dscp);
rtc::PacketOptions updated_options;
updated_options = options;
// Protect if needed.
if (srtp_filter_.IsActive()) {
bool res;
@ -551,21 +553,22 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
res = srtp_filter_.ProtectRtp(
data, len, static_cast<int>(packet->capacity()), &len);
#else
options.packet_time_params.rtp_sendtime_extension_id =
updated_options.packet_time_params.rtp_sendtime_extension_id =
rtp_abs_sendtime_extn_id_;
res = srtp_filter_.ProtectRtp(
data, len, static_cast<int>(packet->capacity()), &len,
&options.packet_time_params.srtp_packet_index);
&updated_options.packet_time_params.srtp_packet_index);
// If protection succeeds, let's get auth params from srtp.
if (res) {
uint8_t* auth_key = NULL;
int key_len;
res = srtp_filter_.GetRtpAuthParams(
&auth_key, &key_len, &options.packet_time_params.srtp_auth_tag_len);
&auth_key, &key_len,
&updated_options.packet_time_params.srtp_auth_tag_len);
if (res) {
options.packet_time_params.srtp_auth_key.resize(key_len);
options.packet_time_params.srtp_auth_key.assign(auth_key,
auth_key + key_len);
updated_options.packet_time_params.srtp_auth_key.resize(key_len);
updated_options.packet_time_params.srtp_auth_key.assign(
auth_key, auth_key + key_len);
}
}
#endif
@ -605,7 +608,7 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
// Bon voyage.
int ret =
channel->SendPacket(packet->data<char>(), packet->size(), options,
channel->SendPacket(packet->data<char>(), packet->size(), updated_options,
(secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
if (ret != static_cast<int>(packet->size())) {
if (channel->GetError() == EWOULDBLOCK) {
@ -1143,7 +1146,7 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
it != streams.end(); ++it) {
if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) {
if (media_channel()->AddSendStream(*it)) {
LOG(LS_INFO) << "Add send ssrc: " << it->ssrcs[0];
LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0];
} else {
std::ostringstream desc;
desc << "Failed to add send stream ssrc: " << it->first_ssrc();
@ -1244,7 +1247,8 @@ void BaseChannel::OnMessage(rtc::Message *pmsg) {
case MSG_RTPPACKET:
case MSG_RTCPPACKET: {
PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet, data->dscp);
SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet,
data->options);
delete data; // because it is Posted
break;
}

12
talk/session/media/channel.h
View File

@ -199,9 +199,8 @@ class BaseChannel
// NetworkInterface implementation, called by MediaEngine
virtual bool SendPacket(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp);
virtual bool SendRtcp(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp);
const rtc::PacketOptions& options);
virtual bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options);
// From TransportChannel
void OnWritableState(TransportChannel* channel);
@ -214,8 +213,9 @@ class BaseChannel
bool PacketIsRtcp(const TransportChannel* channel, const char* data,
size_t len);
bool SendPacket(bool rtcp, rtc::Buffer* packet,
rtc::DiffServCodePoint dscp);
bool SendPacket(bool rtcp,
rtc::Buffer* packet,
const rtc::PacketOptions& options);
virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
void HandlePacket(bool rtcp, rtc::Buffer* packet,
const rtc::PacketTime& packet_time);
@ -470,8 +470,6 @@ class VideoChannel : public BaseChannel {
bool SendIntraFrame();
bool RequestIntraFrame();
// Configure sending media on the stream with SSRC |ssrc|
// If there is only one sending stream SSRC 0 can be used.
bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
private:

26
talk/session/media/channel_unittest.cc
View File

@ -294,11 +294,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
bool SendRtp1() {
return media_channel1_->SendRtp(rtp_packet_.c_str(),
static_cast<int>(rtp_packet_.size()));
static_cast<int>(rtp_packet_.size()),
rtc::PacketOptions());
}
bool SendRtp2() {
return media_channel2_->SendRtp(rtp_packet_.c_str(),
static_cast<int>(rtp_packet_.size()));
static_cast<int>(rtp_packet_.size()),
rtc::PacketOptions());
}
bool SendRtcp1() {
return media_channel1_->SendRtcp(rtcp_packet_.c_str(),
@ -311,13 +313,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
// Methods to send custom data.
bool SendCustomRtp1(uint32_t ssrc, int sequence_number, int pl_type = -1) {
std::string data(CreateRtpData(ssrc, sequence_number, pl_type));
return media_channel1_->SendRtp(data.c_str(),
static_cast<int>(data.size()));
return media_channel1_->SendRtp(data.c_str(), static_cast<int>(data.size()),
rtc::PacketOptions());
}
bool SendCustomRtp2(uint32_t ssrc, int sequence_number, int pl_type = -1) {
std::string data(CreateRtpData(ssrc, sequence_number, pl_type));
return media_channel2_->SendRtp(data.c_str(),
static_cast<int>(data.size()));
return media_channel2_->SendRtp(data.c_str(), static_cast<int>(data.size()),
rtc::PacketOptions());
}
bool SendCustomRtcp1(uint32_t ssrc) {
std::string data(CreateRtcpData(ssrc));
@ -957,7 +959,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
public:
LastWordMediaChannel() : T::MediaChannel(NULL, typename T::Options()) {}
~LastWordMediaChannel() {
T::MediaChannel::SendRtp(kPcmuFrame, sizeof(kPcmuFrame));
T::MediaChannel::SendRtp(kPcmuFrame, sizeof(kPcmuFrame),
rtc::PacketOptions());
T::MediaChannel::SendRtcp(kRtcpReport, sizeof(kRtcpReport));
}
};
@ -1709,21 +1712,24 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
&error_handler, &SrtpErrorHandler::OnSrtpError);
// Testing failures in sending packets.
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket)));
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
rtc::PacketOptions()));
// The first failure will trigger an error.
EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
EXPECT_EQ(cricket::SrtpFilter::PROTECT, error_handler.mode_);
error_handler.error_ = cricket::SrtpFilter::ERROR_NONE;
error_handler.mode_ = cricket::SrtpFilter::UNPROTECT;
// The next 250 ms failures will not trigger an error.
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket)));
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
rtc::PacketOptions()));
// Wait for a while to ensure no message comes in.
rtc::Thread::Current()->ProcessMessages(200);
EXPECT_EQ(cricket::SrtpFilter::ERROR_NONE, error_handler.error_);
EXPECT_EQ(cricket::SrtpFilter::UNPROTECT, error_handler.mode_);
// Wait for a little more - the error will be triggered again.
rtc::Thread::Current()->ProcessMessages(200);
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket)));
EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
rtc::PacketOptions()));
EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
EXPECT_EQ(cricket::SrtpFilter::PROTECT, error_handler.mode_);

41
talk/session/media/channelmanager_unittest.cc
View File

@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/mediacontroller.h"
#include "talk/app/webrtc/fakemediacontroller.h"
#include "talk/media/base/fakecapturemanager.h"
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/base/fakevideocapturer.h"
@ -50,37 +50,24 @@ static const VideoCodec kVideoCodecs[] = {
VideoCodec(96, "rtx", 100, 200, 300, 0),
};
class FakeMediaController : public webrtc::MediaControllerInterface {
public:
explicit FakeMediaController(webrtc::Call* call) : call_(call) {
RTC_DCHECK(nullptr != call);
}
~FakeMediaController() override {}
webrtc::Call* call_w() override { return call_; }
private:
webrtc::Call* call_;
};
class ChannelManagerTest : public testing::Test {
protected:
ChannelManagerTest()
: fake_call_(webrtc::Call::Config()),
fake_mc_(&fake_call_),
fme_(NULL),
fcm_(NULL),
cm_(NULL) {}
: fme_(new cricket::FakeMediaEngine()),
fdme_(new cricket::FakeDataEngine()),
fcm_(new cricket::FakeCaptureManager()),
cm_(new cricket::ChannelManager(fme_,
fdme_,
fcm_,
rtc::Thread::Current())),
fake_call_(webrtc::Call::Config()),
fake_mc_(cm_, &fake_call_),
transport_controller_(
new cricket::FakeTransportController(ICEROLE_CONTROLLING)) {}
virtual void SetUp() {
fme_ = new cricket::FakeMediaEngine();
fme_->SetAudioCodecs(MAKE_VECTOR(kAudioCodecs));
fme_->SetVideoCodecs(MAKE_VECTOR(kVideoCodecs));
fdme_ = new cricket::FakeDataEngine();
fcm_ = new cricket::FakeCaptureManager();
cm_ = new cricket::ChannelManager(
fme_, fdme_, fcm_, rtc::Thread::Current());
transport_controller_ =
new cricket::FakeTransportController(ICEROLE_CONTROLLING);
}
virtual void TearDown() {
@ -93,12 +80,12 @@ class ChannelManagerTest : public testing::Test {
}
rtc::Thread worker_;
cricket::FakeCall fake_call_;
cricket::FakeMediaController fake_mc_;
cricket::FakeMediaEngine* fme_;
cricket::FakeDataEngine* fdme_;
cricket::FakeCaptureManager* fcm_;
cricket::ChannelManager* cm_;
cricket::FakeCall fake_call_;
cricket::FakeMediaController fake_mc_;
cricket::FakeTransportController* transport_controller_;
};

8
webrtc/base/asyncpacketsocket.h
View File

@ -34,10 +34,11 @@ struct PacketTimeUpdateParams {
// This structure holds meta information for the packet which is about to send
// over network.
struct PacketOptions {
PacketOptions() : dscp(DSCP_NO_CHANGE) {}
explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
PacketOptions() : dscp(DSCP_NO_CHANGE), packet_id(-1) {}
explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp), packet_id(-1) {}
DiffServCodePoint dscp;
int packet_id; // 16 bits, -1 represents "not set".
PacketTimeUpdateParams packet_time_params;
};
@ -109,6 +110,9 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
const SocketAddress&,
const PacketTime&> SignalReadPacket;
// Emitted each time a packet is sent.
sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;
// Emitted when the socket is currently able to send.
sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend;

3
webrtc/base/asynctcpsocket.cc
View File

@ -268,6 +268,9 @@ int AsyncTCPSocket::Send(const void *pv, size_t cb,
return res;
}
rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
SignalSentPacket(this, sent_packet);
// We claim to have sent the whole thing, even if we only sent partial
return static_cast<int>(cb);
}

10
webrtc/base/asyncudpsocket.cc
View File

@ -60,13 +60,19 @@ SocketAddress AsyncUDPSocket::GetRemoteAddress() const {
int AsyncUDPSocket::Send(const void *pv, size_t cb,
const rtc::PacketOptions& options) {
return socket_->Send(pv, cb);
rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
int ret = socket_->Send(pv, cb);
SignalSentPacket(this, sent_packet);
return ret;
}
int AsyncUDPSocket::SendTo(const void *pv, size_t cb,
const SocketAddress& addr,
const rtc::PacketOptions& options) {
return socket_->SendTo(pv, cb, addr);
rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
int ret = socket_->SendTo(pv, cb, addr);
SignalSentPacket(this, sent_packet);
return ret;
}
int AsyncUDPSocket::Close() {

1
webrtc/base/base_tests.gyp
View File

@ -29,6 +29,7 @@
'dependencies': [
'base.gyp:rtc_base',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/test.gyp:field_trial',
],
'direct_dependent_settings': {
'defines': [

9
webrtc/base/socket.h
View File

@ -124,6 +124,15 @@ inline bool IsBlockingError(int e) {
return (e == EWOULDBLOCK) || (e == EAGAIN) || (e == EINPROGRESS);
}
struct SentPacket {
SentPacket() : packet_id(-1), send_time_ms(-1) {}
SentPacket(int packet_id, int64_t send_time_ms)
: packet_id(packet_id), send_time_ms(send_time_ms) {}
int packet_id;
int64_t send_time_ms;
};
// General interface for the socket implementations of various networks. The
// methods match those of normal UNIX sockets very closely.
class Socket {

9
webrtc/base/unittest_main.cc
View File

@ -19,9 +19,16 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/test/field_trial.h"
DEFINE_bool(help, false, "prints this message");
DEFINE_string(log, "", "logging options to use");
DEFINE_string(
force_fieldtrials,
"",
"Field trials control experimental feature code which can be forced. "
"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
" will assign the group Enable to field trial WebRTC-FooFeature.");
#if defined(WEBRTC_WIN)
DEFINE_int(crt_break_alloc, -1, "memory allocation to break on");
DEFINE_bool(default_error_handlers, false,
@ -61,6 +68,8 @@ int main(int argc, char** argv) {
return 0;
}
webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
#if defined(WEBRTC_WIN)
if (!FLAG_default_error_handlers) {
// Make sure any errors don't throw dialogs hanging the test run.

3
webrtc/call.h
View File

@ -16,6 +16,7 @@
#include "webrtc/common_types.h"
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio_send_stream.h"
#include "webrtc/base/socket.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@ -137,6 +138,8 @@ class Call {
const Config::BitrateConfig& bitrate_config) = 0;
virtual void SignalNetworkState(NetworkState state) = 0;
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
virtual ~Call() {}
};

6
webrtc/call/call.cc
View File

@ -77,6 +77,8 @@ class Call : public webrtc::Call, public PacketReceiver {
const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
void SignalNetworkState(NetworkState state) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
private:
DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
size_t length);
@ -411,6 +413,10 @@ void Call::SignalNetworkState(NetworkState state) {
}
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
channel_group_->OnSentPacket(sent_packet);
}
void Call::ConfigureSync(const std::string& sync_group) {
// Set sync only if there was no previous one.
if (config_.voice_engine == nullptr || sync_group.empty())

2
webrtc/config.cc
View File

@ -36,7 +36,7 @@ const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
const char* RtpExtension::kAudioLevel =
"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
const char* RtpExtension::kTransportSequenceNumber =
"http://www.webrtc.org/experiments/rtp-hdrext/transport-sequence-number";
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
bool RtpExtension::IsSupportedForAudio(const std::string& name) {
return name == webrtc::RtpExtension::kAbsSendTime ||

2
webrtc/libjingle_examples.gyp
View File

@ -81,6 +81,7 @@
],
'dependencies': [
'../talk/libjingle.gyp:libjingle_peerconnection',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<@(libjingle_tests_additional_deps)',
],
'conditions': [
@ -139,6 +140,7 @@
'target_name': 'apprtc_common',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'../talk/libjingle.gyp:libjingle_peerconnection_objc',
],
'sources': [

8
webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
View File

@ -49,11 +49,17 @@ void TransportFeedbackAdapter::SetBitrateEstimator(
}
}
void TransportFeedbackAdapter::OnPacketSent(const PacketInfo& info) {
void TransportFeedbackAdapter::OnSentPacket(const PacketInfo& info) {
rtc::CritScope cs(&lock_);
send_time_history_.AddAndRemoveOld(info);
}
void TransportFeedbackAdapter::UpdateSendTime(uint16_t sequence_number,
int64_t send_time_ms) {
rtc::CritScope cs(&lock_);
send_time_history_.UpdateSendTime(sequence_number, send_time_ms);
}
void TransportFeedbackAdapter::OnTransportFeedback(
const rtcp::TransportFeedback& feedback) {
int64_t timestamp_us = feedback.GetBaseTimeUs();

4
webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
View File

@ -33,7 +33,9 @@ class TransportFeedbackAdapter : public TransportFeedbackObserver,
ProcessThread* process_thread);
virtual ~TransportFeedbackAdapter();
void OnPacketSent(const PacketInfo& info) override;
void OnSentPacket(const PacketInfo& info) override;
void UpdateSendTime(uint16_t sequence_number, int64_t send_time_ms);
void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;

14
webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
View File

@ -103,9 +103,9 @@ class TransportFeedbackAdapterTest : public ::testing::Test {
}
// Utility method, to reset arrival_time_ms before adding send time.
void OnPacketSent(PacketInfo info) {
void OnSentPacket(PacketInfo info) {
info.arrival_time_ms = 0;
adapter_->OnPacketSent(info);
adapter_->OnSentPacket(info);
}
SimulatedClock clock_;
@ -125,7 +125,7 @@ TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
packets.push_back(PacketInfo(140, 240, 4, 1500, true));
for (const PacketInfo& packet : packets)
OnPacketSent(packet);
OnSentPacket(packet);
rtcp::TransportFeedback feedback;
feedback.WithBase(packets[0].sequence_number,
@ -160,7 +160,7 @@ TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
for (const PacketInfo& packet : packets) {
if (packet.sequence_number >= kSendSideDropBefore)
OnPacketSent(packet);
OnSentPacket(packet);
}
rtcp::TransportFeedback feedback;
@ -199,7 +199,7 @@ TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
packets.push_back(PacketInfo(kHighArrivalTimeMs, 220, 2, 1500, true));
for (const PacketInfo& packet : packets)
OnPacketSent(packet);
OnSentPacket(packet);
for (size_t i = 0; i < packets.size(); ++i) {
rtc::scoped_ptr<rtcp::TransportFeedback> feedback(
@ -263,8 +263,8 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
// Packets will be added to send history.
for (const PacketInfo& packet : sent_packets)
OnPacketSent(packet);
OnPacketSent(info);
OnSentPacket(packet);
OnSentPacket(info);
// Create expected feedback and send into adapter.
rtc::scoped_ptr<rtcp::TransportFeedback> feedback(

2
webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
View File

@ -313,7 +313,7 @@ class TransportFeedbackObserver {
// Note: Transport-wide sequence number as sequence number. Arrival time
// must be set to 0.
virtual void OnPacketSent(const PacketInfo& info) = 0;
virtual void OnSentPacket(const PacketInfo& info) = 0;
virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
};

6
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -672,8 +672,8 @@ size_t RTPSender::SendPadData(size_t bytes,
break;
if (using_transport_seq && transport_feedback_observer_) {
transport_feedback_observer_->OnPacketSent(PacketInfo(
0, now_ms, options.packet_id, length, true));
transport_feedback_observer_->OnSentPacket(
PacketInfo(0, now_ms, options.packet_id, length, true));
}
bytes_sent += padding_bytes_in_packet;
@ -934,7 +934,7 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
media_has_been_sent_ = true;
}
if (using_transport_seq && transport_feedback_observer_) {
transport_feedback_observer_->OnPacketSent(
transport_feedback_observer_->OnSentPacket(
PacketInfo(0, now_ms, options.packet_id, length, true));
}
UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx,

10
webrtc/p2p/base/dtlstransportchannel.cc
View File

@ -101,6 +101,8 @@ DtlsTransportChannelWrapper::DtlsTransportChannelWrapper(
&DtlsTransportChannelWrapper::OnWritableState);
channel_->SignalReadPacket.connect(this,
&DtlsTransportChannelWrapper::OnReadPacket);
channel_->SignalSentPacket.connect(
this, &DtlsTransportChannelWrapper::OnSentPacket);
channel_->SignalReadyToSend.connect(this,
&DtlsTransportChannelWrapper::OnReadyToSend);
channel_->SignalGatheringState.connect(
@ -510,6 +512,14 @@ void DtlsTransportChannelWrapper::OnReadPacket(
}
}
void DtlsTransportChannelWrapper::OnSentPacket(
TransportChannel* channel,
const rtc::SentPacket& sent_packet) {
ASSERT(rtc::Thread::Current() == worker_thread_);
SignalSentPacket(this, sent_packet);
}
void DtlsTransportChannelWrapper::OnReadyToSend(TransportChannel* channel) {
if (writable()) {
SignalReadyToSend(this);

5
webrtc/p2p/base/dtlstransportchannel.h
View File

@ -209,6 +209,8 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
void OnWritableState(TransportChannel* channel);
void OnReadPacket(TransportChannel* channel, const char* data, size_t size,
const rtc::PacketTime& packet_time, int flags);
void OnSentPacket(TransportChannel* channel,
const rtc::SentPacket& sent_packet);
void OnReadyToSend(TransportChannel* channel);
void OnReceivingState(TransportChannel* channel);
void OnDtlsEvent(rtc::StreamInterface* stream_, int sig, int err);
@ -223,7 +225,8 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
Transport* transport_; // The transport_ that created us.
rtc::Thread* worker_thread_; // Everything should occur on this thread.
TransportChannelImpl* channel_; // Underlying channel, owned by transport_.
// Underlying channel, owned by transport_.
TransportChannelImpl* const channel_;
rtc::scoped_ptr<rtc::SSLStreamAdapter> dtls_; // The DTLS stream
StreamInterfaceChannel* downward_; // Wrapper for channel_, owned by dtls_.
std::vector<std::string> srtp_ciphers_; // SRTP ciphers to use with DTLS.

21
webrtc/p2p/base/dtlstransportchannel_unittest.cc
View File

@ -33,6 +33,7 @@ static const char kIceUfrag1[] = "TESTICEUFRAG0001";
static const char kIcePwd1[] = "TESTICEPWD00000000000001";
static const size_t kPacketNumOffset = 8;
static const size_t kPacketHeaderLen = 12;
static const int kFakePacketId = 0x1234;
static bool IsRtpLeadByte(uint8_t b) {
return ((b & 0xC0) == 0x80);
@ -86,6 +87,8 @@ class DtlsTestClient : public sigslot::has_slots<> {
&DtlsTestClient::OnTransportChannelWritableState);
channel->SignalReadPacket.connect(this,
&DtlsTestClient::OnTransportChannelReadPacket);
channel->SignalSentPacket.connect(
this, &DtlsTestClient::OnTransportChannelSentPacket);
channels_.push_back(channel);
// Hook the raw packets so that we can verify they are encrypted.
@ -259,6 +262,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
// Only set the bypass flag if we've activated DTLS.
int flags = (certificate_ && srtp) ? cricket::PF_SRTP_BYPASS : 0;
rtc::PacketOptions packet_options;
packet_options.packet_id = kFakePacketId;
int rv = channels_[channel]->SendPacket(
packet.get(), size, packet_options, flags);
ASSERT_GT(rv, 0);
@ -338,6 +342,13 @@ class DtlsTestClient : public sigslot::has_slots<> {
ASSERT_EQ(expected_flags, flags);
}
void OnTransportChannelSentPacket(cricket::TransportChannel* channel,
const rtc::SentPacket& sent_packet) {
sent_packet_ = sent_packet;
}
rtc::SentPacket sent_packet() const { return sent_packet_; }
// Hook into the raw packet stream to make sure DTLS packets are encrypted.
void OnFakeTransportChannelReadPacket(cricket::TransportChannel* channel,
const char* data, size_t size,
@ -378,6 +389,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
bool negotiated_dtls_;
bool received_dtls_client_hello_;
bool received_dtls_server_hello_;
rtc::SentPacket sent_packet_;
};
@ -558,6 +570,15 @@ TEST_F(DtlsTransportChannelTest, TestTransfer) {
TestTransfer(0, 1000, 100, false);
}
// Connect without DTLS, and transfer some data.
TEST_F(DtlsTransportChannelTest, TestOnSentPacket) {
ASSERT_TRUE(Connect());
EXPECT_EQ(client1_.sent_packet().send_time_ms, -1);
TestTransfer(0, 1000, 100, false);
EXPECT_EQ(kFakePacketId, client1_.sent_packet().packet_id);
EXPECT_GE(client1_.sent_packet().send_time_ms, 0);
}
// Create two channels without DTLS, and transfer some data.
TEST_F(DtlsTransportChannelTest, TestTransferTwoChannels) {
SetChannelCount(2);

4
webrtc/p2p/base/faketransportcontroller.h
View File

@ -31,10 +31,12 @@ namespace cricket {
class FakeTransport;
namespace {
struct PacketMessageData : public rtc::MessageData {
PacketMessageData(const char* data, size_t len) : packet(data, len) {}
rtc::Buffer packet;
};
} // namespace
// Fake transport channel class, which can be passed to anything that needs a
// transport channel. Can be informed of another FakeTransportChannel via
@ -208,6 +210,8 @@ class FakeTransportChannel : public TransportChannelImpl,
} else {
rtc::Thread::Current()->Send(this, 0, packet);
}
rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
SignalSentPacket(this, sent_packet);
return static_cast<int>(len);
}
int SetOption(rtc::Socket::Option opt, int value) override { return true; }

8
webrtc/p2p/base/p2ptransportchannel.cc
View File

@ -435,6 +435,7 @@ void P2PTransportChannel::OnPortReady(PortAllocatorSession *session,
port->SignalDestroyed.connect(this, &P2PTransportChannel::OnPortDestroyed);
port->SignalRoleConflict.connect(
this, &P2PTransportChannel::OnRoleConflict);
port->SignalSentPacket.connect(this, &P2PTransportChannel::OnSentPacket);
// Attempt to create a connection from this new port to all of the remote
// candidates that we were given so far.
@ -1356,6 +1357,13 @@ void P2PTransportChannel::OnReadPacket(Connection* connection,
}
}
void P2PTransportChannel::OnSentPacket(PortInterface* port,
const rtc::SentPacket& sent_packet) {
ASSERT(worker_thread_ == rtc::Thread::Current());
SignalSentPacket(this, sent_packet);
}
void P2PTransportChannel::OnReadyToSend(Connection* connection) {
if (connection == best_connection_ && writable()) {
SignalReadyToSend(this);

1
webrtc/p2p/base/p2ptransportchannel.h
View File

@ -207,6 +207,7 @@ class P2PTransportChannel : public TransportChannelImpl,
void OnConnectionStateChange(Connection* connection);
void OnReadPacket(Connection *connection, const char *data, size_t len,
const rtc::PacketTime& packet_time);
void OnSentPacket(PortInterface* port, const rtc::SentPacket& sent_packet);
void OnReadyToSend(Connection* connection);
void OnConnectionDestroyed(Connection *connection);

4
webrtc/p2p/base/port.cc
View File

@ -310,6 +310,10 @@ void Port::OnReadPacket(
}
}
void Port::OnSentPacket(const rtc::SentPacket& sent_packet) {
PortInterface::SignalSentPacket(this, sent_packet);
}
void Port::OnReadyToSend() {
AddressMap::iterator iter = connections_.begin();
for (; iter != connections_.end(); ++iter) {

3
webrtc/p2p/base/port.h
View File

@ -275,6 +275,9 @@ class Port : public PortInterface, public rtc::MessageHandler,
IceMessage* stun_msg,
const std::string& remote_ufrag);
// Called when a packet has been sent to the socket.
void OnSentPacket(const rtc::SentPacket& sent_packet);
// Called when the socket is currently able to send.
void OnReadyToSend();

4
webrtc/p2p/base/portinterface.h
View File

@ -14,6 +14,7 @@
#include <string>
#include "webrtc/p2p/base/transport.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/socketaddress.h"
namespace rtc {
@ -112,6 +113,9 @@ class PortInterface {
sigslot::signal4<PortInterface*, const char*, size_t,
const rtc::SocketAddress&> SignalReadPacket;
// Emitted each time a packet is sent on this port.
sigslot::signal2<PortInterface*, const rtc::SentPacket&> SignalSentPacket;
virtual std::string ToString() const = 0;
protected:

10
webrtc/p2p/base/relayport.cc
View File

@ -144,6 +144,10 @@ class RelayEntry : public rtc::MessageHandler,
const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
void OnSentPacket(rtc::AsyncPacketSocket* socket,
const rtc::SentPacket& sent_packet);
// Called when the socket is currently able to send.
void OnReadyToSend(rtc::AsyncPacketSocket* socket);
@ -508,6 +512,7 @@ void RelayEntry::Connect() {
// Otherwise, create the new connection and configure any socket options.
socket->SignalReadPacket.connect(this, &RelayEntry::OnReadPacket);
socket->SignalSentPacket.connect(this, &RelayEntry::OnSentPacket);
socket->SignalReadyToSend.connect(this, &RelayEntry::OnReadyToSend);
current_connection_ = new RelayConnection(ra, socket, port()->thread());
for (size_t i = 0; i < port_->options().size(); ++i) {
@ -747,6 +752,11 @@ void RelayEntry::OnReadPacket(
PROTO_UDP, packet_time);
}
void RelayEntry::OnSentPacket(rtc::AsyncPacketSocket* socket,
const rtc::SentPacket& sent_packet) {
port_->OnSentPacket(sent_packet);
}
void RelayEntry::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
if (connected()) {
port_->OnReadyToSend();

6
webrtc/p2p/base/stunport.cc
View File

@ -217,6 +217,7 @@ bool UDPPort::Init() {
}
socket_->SignalReadPacket.connect(this, &UDPPort::OnReadPacket);
}
socket_->SignalSentPacket.connect(this, &UDPPort::OnSentPacket);
socket_->SignalReadyToSend.connect(this, &UDPPort::OnReadyToSend);
socket_->SignalAddressReady.connect(this, &UDPPort::OnLocalAddressReady);
requests_.SignalSendPacket.connect(this, &UDPPort::OnSendPacket);
@ -329,6 +330,11 @@ void UDPPort::OnReadPacket(
}
}
void UDPPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
const rtc::SentPacket& sent_packet) {
Port::OnSentPacket(sent_packet);
}
void UDPPort::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
Port::OnReadyToSend();
}

3
webrtc/p2p/base/stunport.h
View File

@ -140,6 +140,9 @@ class UDPPort : public Port {
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
void OnSentPacket(rtc::AsyncPacketSocket* socket,
const rtc::SentPacket& sent_packet);
void OnReadyToSend(rtc::AsyncPacketSocket* socket);
// This method will send STUN binding request if STUN server address is set.

5
webrtc/p2p/base/transportchannel.h
View File

@ -48,7 +48,7 @@ enum TransportChannelState {
// between the two sides of a session.
class TransportChannel : public sigslot::has_slots<> {
public:
explicit TransportChannel(const std::string& transport_name, int component)
TransportChannel(const std::string& transport_name, int component)
: transport_name_(transport_name),
component_(component),
writable_(false),
@ -134,6 +134,9 @@ class TransportChannel : public sigslot::has_slots<> {
sigslot::signal5<TransportChannel*, const char*,
size_t, const rtc::PacketTime&, int> SignalReadPacket;
// Signalled each time a packet is sent on this channel.
sigslot::signal2<TransportChannel*, const rtc::SentPacket&> SignalSentPacket;
// This signal occurs when there is a change in the way that packets are
// being routed, i.e. to a different remote location. The candidate
// indicates where and how we are currently sending media.

2
webrtc/transport.h
View File

@ -17,6 +17,8 @@
namespace webrtc {
// TODO(holmer): Look into unifying this with the PacketOptions in
// asyncpacketsocket.h.
struct PacketOptions {
// A 16 bits positive id. Negative ids are invalid and should be interpreted
// as packet_id not being set.

7
webrtc/video_engine/vie_channel_group.cc
View File

@ -445,4 +445,11 @@ void ChannelGroup::OnNetworkChanged(uint32_t target_bitrate_bps,
PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000,
pad_up_to_bitrate_bps / 1000);
}
void ChannelGroup::OnSentPacket(const rtc::SentPacket& sent_packet) {
if (transport_feedback_adapter_) {
transport_feedback_adapter_->UpdateSendTime(sent_packet.packet_id,
sent_packet.send_time_ms);
}
}
} // namespace webrtc

3
webrtc/video_engine/vie_channel_group.h
View File

@ -18,6 +18,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/socket.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@ -82,6 +83,8 @@ class ChannelGroup : public BitrateObserver {
uint8_t fraction_loss,
int64_t rtt) override;
void OnSentPacket(const rtc::SentPacket& sent_packet);
private:
typedef std::map<int, ViEChannel*> ChannelMap;
typedef std::map<int, ViEEncoder*> EncoderMap;