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From baef69058c64651cd6d5a3d81cce6a9fa5b5c937 Mon Sep 17 00:00:00 2001
From: Arlo Breault <arlolra@gmail.com>
Date: Fri, 1 Jul 2016 19:07:47 -0700
Subject: [PATCH] With a rat fur
---
webrtc/api/api.gyp | 365 +++++++++++++++++++++-----------
webrtc/api/mediacontroller.cc | 6 +
webrtc/api/peerconnection.cc | 58 ++++-
webrtc/api/peerconnection.h | 14 +-
webrtc/api/peerconnectionfactory.cc | 20 +-
webrtc/api/peerconnectionfactory.h | 11 +-
webrtc/api/peerconnectionfactoryproxy.h | 6 +
webrtc/api/peerconnectioninterface.h | 8 +
webrtc/api/peerconnectionproxy.h | 6 +
webrtc/api/statscollector.cc | 4 +
webrtc/build/common.gypi | 1 +
webrtc/call.h | 4 +
webrtc/call/call.cc | 39 ++++
webrtc/call/webrtc_call.gypi | 8 +-
webrtc/media/media.gyp | 73 ++++++-
webrtc/pc/channelmanager.cc | 6 +
webrtc/pc/mediasession.cc | 2 +
webrtc/pc/pc.gyp | 2 +-
webrtc/webrtc.gyp | 2 +-
19 files changed, 488 insertions(+), 147 deletions(-)
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index 377ef8f..2483600 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -121,138 +121,253 @@
}, # libjingle_peerconnection_java
]
}],
- ], # conditions
- 'targets': [
- {
- 'target_name': 'libjingle_peerconnection',
- 'type': 'static_library',
- 'dependencies': [
- '<(webrtc_root)/media/media.gyp:rtc_media',
- '<(webrtc_root)/pc/pc.gyp:rtc_pc',
- ],
- 'sources': [
- 'audiotrack.cc',
- 'audiotrack.h',
- 'datachannel.cc',
- 'datachannel.h',
- 'datachannelinterface.h',
- 'dtlsidentitystore.cc',
- 'dtlsidentitystore.h',
- 'dtmfsender.cc',
- 'dtmfsender.h',
- 'dtmfsenderinterface.h',
- 'jsep.h',
- 'jsepicecandidate.cc',
- 'jsepicecandidate.h',
- 'jsepsessiondescription.cc',
- 'jsepsessiondescription.h',
- 'localaudiosource.cc',
- 'localaudiosource.h',
- 'mediaconstraintsinterface.cc',
- 'mediaconstraintsinterface.h',
- 'mediacontroller.cc',
- 'mediacontroller.h',
- 'mediastream.cc',
- 'mediastream.h',
- 'mediastreaminterface.h',
- 'mediastreamobserver.cc',
- 'mediastreamobserver.h',
- 'mediastreamprovider.h',
- 'mediastreamproxy.h',
- 'mediastreamtrack.h',
- 'mediastreamtrackproxy.h',
- 'notifier.h',
- 'peerconnection.cc',
- 'peerconnection.h',
- 'peerconnectionfactory.cc',
- 'peerconnectionfactory.h',
- 'peerconnectionfactoryproxy.h',
- 'peerconnectioninterface.h',
- 'peerconnectionproxy.h',
- 'proxy.h',
- 'remoteaudiosource.cc',
- 'remoteaudiosource.h',
- 'rtpparameters.h',
- 'rtpreceiver.cc',
- 'rtpreceiver.h',
- 'rtpreceiverinterface.h',
- 'rtpsender.cc',
- 'rtpsender.h',
- 'rtpsenderinterface.h',
- 'sctputils.cc',
- 'sctputils.h',
- 'statscollector.cc',
- 'statscollector.h',
- 'statstypes.cc',
- 'statstypes.h',
- 'streamcollection.h',
- 'videocapturertracksource.cc',
- 'videocapturertracksource.h',
- 'videosourceproxy.h',
- 'videotrack.cc',
- 'videotrack.h',
- 'videotracksource.cc',
- 'videotracksource.h',
- 'webrtcsdp.cc',
- 'webrtcsdp.h',
- 'webrtcsession.cc',
- 'webrtcsession.h',
- 'webrtcsessiondescriptionfactory.cc',
- 'webrtcsessiondescriptionfactory.h',
- ],
- # TODO(kjellander): Make the code compile without disabling these flags.
- # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
- 'cflags': [
- '-Wno-sign-compare',
- ],
- 'cflags_cc!': [
- '-Woverloaded-virtual',
- ],
- 'conditions': [
- ['clang==1', {
- 'cflags!': [
- '-Wextra',
+ ['build_have_not == 1', {
+ 'targets': [
+ {
+ 'target_name': 'libjingle_peerconnection',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/media/media.gyp:rtc_media',
+ '<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
- 'xcode_settings': {
- 'WARNING_CFLAGS!': ['-Wextra'],
- },
- }, {
+ 'sources': [
+ 'audiotrack.cc',
+ 'audiotrack.h',
+ 'datachannel.cc',
+ 'datachannel.h',
+ 'datachannelinterface.h',
+ 'dtlsidentitystore.cc',
+ 'dtlsidentitystore.h',
+ 'dtmfsender.cc',
+ 'dtmfsender.h',
+ 'dtmfsenderinterface.h',
+ 'jsep.h',
+ 'jsepicecandidate.cc',
+ 'jsepicecandidate.h',
+ 'jsepsessiondescription.cc',
+ 'jsepsessiondescription.h',
+ 'localaudiosource.cc',
+ 'localaudiosource.h',
+ 'mediaconstraintsinterface.cc',
+ 'mediaconstraintsinterface.h',
+ 'mediacontroller.cc',
+ 'mediacontroller.h',
+ 'mediastream.cc',
+ 'mediastream.h',
+ 'mediastreaminterface.h',
+ 'mediastreamobserver.cc',
+ 'mediastreamobserver.h',
+ 'mediastreamprovider.h',
+ 'mediastreamproxy.h',
+ 'mediastreamtrack.h',
+ 'mediastreamtrackproxy.h',
+ 'notifier.h',
+ 'peerconnection.cc',
+ 'peerconnection.h',
+ 'peerconnectionfactory.cc',
+ 'peerconnectionfactory.h',
+ 'peerconnectionfactoryproxy.h',
+ 'peerconnectioninterface.h',
+ 'peerconnectionproxy.h',
+ 'proxy.h',
+ 'remoteaudiosource.cc',
+ 'remoteaudiosource.h',
+ 'rtpparameters.h',
+ 'rtpreceiver.cc',
+ 'rtpreceiver.h',
+ 'rtpreceiverinterface.h',
+ 'rtpsender.cc',
+ 'rtpsender.h',
+ 'rtpsenderinterface.h',
+ 'sctputils.cc',
+ 'sctputils.h',
+ 'statscollector.cc',
+ 'statscollector.h',
+ 'statstypes.cc',
+ 'statstypes.h',
+ 'streamcollection.h',
+ 'videocapturertracksource.cc',
+ 'videocapturertracksource.h',
+ 'videosourceproxy.h',
+ 'videotrack.cc',
+ 'videotrack.h',
+ 'videotracksource.cc',
+ 'videotracksource.h',
+ 'webrtcsdp.cc',
+ 'webrtcsdp.h',
+ 'webrtcsession.cc',
+ 'webrtcsession.h',
+ 'webrtcsessiondescriptionfactory.cc',
+ 'webrtcsessiondescriptionfactory.h',
+ ],
+ # TODO(kjellander): Make the code compile without disabling these flags.
+ # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags': [
- '-Wno-maybe-uninitialized', # Only exists for GCC.
+ '-Wno-sign-compare',
],
- }],
- ['OS=="win"', {
- # Disable warning for signed/unsigned mismatch.
- 'msvs_settings': {
- 'VCCLCompilerTool': {
- 'AdditionalOptions!': ['/we4389'],
- },
- },
- }],
- ['OS=="win" and clang==1', {
- 'msvs_settings': {
- 'VCCLCompilerTool': {
- 'AdditionalOptions': [
- '-Wno-sign-compare',
+ 'cflags_cc!': [
+ '-Woverloaded-virtual',
+ ],
+ 'conditions': [
+ ['clang==1', {
+ 'cflags!': [
+ '-Wextra',
],
- },
- },
- }],
- ['use_quic==1', {
+ 'xcode_settings': {
+ 'WARNING_CFLAGS!': ['-Wextra'],
+ },
+ }, {
+ 'cflags': [
+ '-Wno-maybe-uninitialized', # Only exists for GCC.
+ ],
+ }],
+ ['OS=="win"', {
+ # Disable warning for signed/unsigned mismatch.
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions!': ['/we4389'],
+ },
+ },
+ }],
+ ['OS=="win" and clang==1', {
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions': [
+ '-Wno-sign-compare',
+ ],
+ },
+ },
+ }],
+ ['use_quic==1', {
+ 'dependencies': [
+ '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+ ],
+ 'sources': [
+ 'quicdatachannel.cc',
+ 'quicdatachannel.h',
+ 'quicdatatransport.cc',
+ 'quicdatatransport.h',
+ ],
+ 'export_dependent_settings': [
+ '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+ ],
+ }],
+ ],
+ }, # target libjingle_peerconnection
+ ], # targets
+ }],
+ ['build_have_not == 0', {
+ 'targets': [
+ {
+ 'target_name': 'libjingle_peerconnection_not',
+ 'type': 'static_library',
'dependencies': [
- '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+ '<(webrtc_root)/media/media.gyp:rtc_media_not',
+ '<(webrtc_root)/pc/pc.gyp:rtc_pc',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
],
'sources': [
- 'quicdatachannel.cc',
- 'quicdatachannel.h',
- 'quicdatatransport.cc',
- 'quicdatatransport.h',
+ 'datachannel.cc',
+ 'datachannel.h',
+ 'datachannelinterface.h',
+ 'dtlsidentitystore.cc',
+ 'dtlsidentitystore.h',
+ 'dtmfsender.cc',
+ 'dtmfsender.h',
+ 'dtmfsenderinterface.h',
+ 'jsep.h',
+ 'jsepicecandidate.cc',
+ 'jsepicecandidate.h',
+ 'jsepsessiondescription.cc',
+ 'jsepsessiondescription.h',
+ 'mediaconstraintsinterface.cc',
+ 'mediaconstraintsinterface.h',
+ 'mediacontroller.cc',
+ 'mediacontroller.h',
+ 'mediastreamobserver.h',
+ 'notifier.h',
+ 'peerconnection.cc',
+ 'peerconnection.h',
+ 'peerconnectionfactory.cc',
+ 'peerconnectionfactory.h',
+ 'peerconnectionfactoryproxy.h',
+ 'peerconnectioninterface.h',
+ 'peerconnectionproxy.h',
+ 'proxy.h',
+ 'rtpparameters.h',
+ 'rtpreceiver.h',
+ 'rtpreceiverinterface.h',
+ 'rtpsender.h',
+ 'rtpsenderinterface.h',
+ 'sctputils.cc',
+ 'sctputils.h',
+ 'statscollector.cc',
+ 'statscollector.h',
+ 'statstypes.cc',
+ 'statstypes.h',
+ 'streamcollection.h',
+ 'webrtcsdp.cc',
+ 'webrtcsdp.h',
+ 'webrtcsession.cc',
+ 'webrtcsession.h',
+ 'webrtcsessiondescriptionfactory.cc',
+ 'webrtcsessiondescriptionfactory.h',
],
- 'export_dependent_settings': [
- '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+ # TODO(kjellander): Make the code compile without disabling these flags.
+ # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
+ 'cflags': [
+ '-Wno-sign-compare',
],
- }],
- ],
- }, # target libjingle_peerconnection
- ], # targets
+ 'cflags_cc!': [
+ '-Woverloaded-virtual',
+ ],
+ 'conditions': [
+ ['clang==1', {
+ 'cflags!': [
+ '-Wextra',
+ ],
+ 'xcode_settings': {
+ 'WARNING_CFLAGS!': ['-Wextra'],
+ },
+ }, {
+ 'cflags': [
+ '-Wno-maybe-uninitialized', # Only exists for GCC.
+ ],
+ }],
+ ['OS=="win"', {
+ # Disable warning for signed/unsigned mismatch.
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions!': ['/we4389'],
+ },
+ },
+ }],
+ ['OS=="win" and clang==1', {
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions': [
+ '-Wno-sign-compare',
+ ],
+ },
+ },
+ }],
+ ['use_quic==1', {
+ 'dependencies': [
+ '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+ ],
+ 'sources': [
+ 'quicdatachannel.cc',
+ 'quicdatachannel.h',
+ 'quicdatatransport.cc',
+ 'quicdatatransport.h',
+ ],
+ 'export_dependent_settings': [
+ '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+ ],
+ }],
+ ],
+ }, # target libjingle_peerconnection_not
+ ], # targets
+ }],
+ ], # conditions
}
diff --git a/webrtc/api/mediacontroller.cc b/webrtc/api/mediacontroller.cc
index 2e4501b..651e769 100644
--- a/webrtc/api/mediacontroller.cc
+++ b/webrtc/api/mediacontroller.cc
@@ -35,9 +35,15 @@ class MediaController : public webrtc::MediaControllerInterface,
media_config_(media_config),
channel_manager_(channel_manager) {
RTC_DCHECK(worker_thread);
+
+ #ifdef HAVE_NOT
+
worker_thread_->Invoke<void>(
rtc::Bind(&MediaController::Construct_w, this,
channel_manager_->media_engine()));
+
+ #endif // HAVE_NOT
+
}
~MediaController() override {
Close();
diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index dda0eeb..04c6be5 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -49,10 +49,14 @@ using webrtc::PeerConnectionInterface;
using webrtc::RtpSenderInterface;
using webrtc::StreamCollection;
+#ifdef HAVE_NOT
+
static const char kDefaultStreamLabel[] = "default";
static const char kDefaultAudioTrackLabel[] = "defaulta0";
static const char kDefaultVideoTrackLabel[] = "defaultv0";
+#endif // HAVE_NOT
+
// The min number of tokens must present in Turn host uri.
// e.g. user@turn.example.org
static const size_t kTurnHostTokensNum = 2;
@@ -313,6 +317,8 @@ bool ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
return true;
}
+#ifdef HAVE_NOT
+
// Check if we can send |new_stream| on a PeerConnection.
bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
webrtc::MediaStreamInterface* new_stream) {
@@ -327,6 +333,8 @@ bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
return true;
}
+#endif // HAVE_NOT
+
bool MediaContentDirectionHasSend(cricket::MediaContentDirection dir) {
return dir == cricket::MD_SENDONLY || dir == cricket::MD_SENDRECV;
}
@@ -626,6 +634,8 @@ PeerConnection::remote_streams() {
return remote_streams_;
}
+#ifdef HAVE_NOT
+
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
if (IsClosed()) {
@@ -801,6 +811,8 @@ rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
return new_sender;
}
+#endif // HAVE_NOT
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
return senders_;
@@ -1010,6 +1022,8 @@ void PeerConnection::SetLocalDescription(
AllocateSctpSids(role);
}
+ #ifdef HAVE_NOT
+
// Update state and SSRC of local MediaStreams and DataChannels based on the
// local session description.
const cricket::ContentInfo* audio_content =
@@ -1038,6 +1052,8 @@ void PeerConnection::SetLocalDescription(
}
}
+ #endif // HAVE_NOT
+
const cricket::ContentInfo* data_content =
GetFirstDataContent(desc->description());
if (data_content) {
@@ -1089,14 +1105,26 @@ void PeerConnection::SetRemoteDescription(
}
const cricket::SessionDescription* remote_desc = desc->description();
+
+ const cricket::DataContentDescription* data_desc =
+ GetFirstDataContentDescription(remote_desc);
+
+ // Update the DataChannels with the information from the remote peer.
+ if (data_desc) {
+ if (rtc::starts_with(data_desc->protocol().data(),
+ cricket::kMediaProtocolRtpPrefix)) {
+ UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
+ }
+ }
+
+ #ifdef HAVE_NOT
+
const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
const cricket::AudioContentDescription* audio_desc =
GetFirstAudioContentDescription(remote_desc);
const cricket::VideoContentDescription* video_desc =
GetFirstVideoContentDescription(remote_desc);
- const cricket::DataContentDescription* data_desc =
- GetFirstDataContentDescription(remote_desc);
// Check if the descriptions include streams, just in case the peer supports
// MSID, but doesn't indicate so with "a=msid-semantic".
@@ -1140,14 +1168,6 @@ void PeerConnection::SetRemoteDescription(
}
}
- // Update the DataChannels with the information from the remote peer.
- if (data_desc) {
- if (rtc::starts_with(data_desc->protocol().data(),
- cricket::kMediaProtocolRtpPrefix)) {
- UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
- }
- }
-
// Iterate new_streams and notify the observer about new MediaStreams.
for (size_t i = 0; i < new_streams->count(); ++i) {
MediaStreamInterface* new_stream = new_streams->at(i);
@@ -1155,6 +1175,8 @@ void PeerConnection::SetRemoteDescription(
observer_->OnAddStream(new_stream);
}
+ #endif // HAVE_NOT
+
UpdateEndedRemoteMediaStreams();
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
@@ -1297,6 +1319,8 @@ void PeerConnection::OnMessage(rtc::Message* msg) {
}
}
+#ifdef HAVE_NOT
+
void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
const std::string& track_id,
uint32_t ssrc) {
@@ -1314,6 +1338,8 @@ void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
session_.get())));
}
+#endif // HAVE_NOT
+
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
// description.
void PeerConnection::DestroyReceiver(const std::string& track_id) {
@@ -1392,6 +1418,8 @@ void PeerConnection::ChangeSignalingState(
observer_->OnSignalingChange(signaling_state_);
}
+#ifdef HAVE_NOT
+
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream) {
auto sender = FindSenderForTrack(track);
@@ -1468,6 +1496,8 @@ void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
senders_.erase(sender);
}
+#endif // HAVE_NOT
+
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
const std::string& error) {
@@ -1580,6 +1610,8 @@ bool PeerConnection::GetOptionsForAnswer(
return true;
}
+#ifdef HAVE_NOT
+
void PeerConnection::RemoveTracks(cricket::MediaType media_type) {
UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
@@ -1706,6 +1738,8 @@ void PeerConnection::OnRemoteTrackRemoved(const std::string& stream_label,
}
}
+#endif // HAVE_NOT
+
void PeerConnection::UpdateEndedRemoteMediaStreams() {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
for (size_t i = 0; i < remote_streams_->count(); ++i) {
@@ -1721,6 +1755,8 @@ void PeerConnection::UpdateEndedRemoteMediaStreams() {
}
}
+#ifdef HAVE_NOT
+
void PeerConnection::UpdateLocalTracks(
const std::vector<cricket::StreamParams>& streams,
cricket::MediaType media_type) {
@@ -1759,6 +1795,8 @@ void PeerConnection::UpdateLocalTracks(
}
}
+#endif // HAVE_NOT
+
void PeerConnection::OnLocalTrackSeen(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h
index 8ba7e58..99a695a 100644
--- a/webrtc/api/peerconnection.h
+++ b/webrtc/api/peerconnection.h
@@ -75,15 +75,19 @@ class PeerConnection : public PeerConnectionInterface,
rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
+
+ virtual WebRtcSession* session() { return session_.get(); }
+
+ #ifdef HAVE_NOT
+
bool AddStream(MediaStreamInterface* local_stream) override;
void RemoveStream(MediaStreamInterface* local_stream) override;
rtc::scoped_refptr<RtpSenderInterface> AddTrack(
MediaStreamTrackInterface* track,
std::vector<MediaStreamInterface*> streams) override;
- bool RemoveTrack(RtpSenderInterface* sender) override;
- virtual WebRtcSession* session() { return session_.get(); }
+ bool RemoveTrack(RtpSenderInterface* sender) override;
rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
AudioTrackInterface* track) override;
@@ -92,6 +96,8 @@ class PeerConnection : public PeerConnectionInterface,
const std::string& kind,
const std::string& stream_id) override;
+ #endif // HAVE_NOT
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const override;
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
@@ -195,6 +201,8 @@ class PeerConnection : public PeerConnectionInterface,
void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
void ChangeSignalingState(SignalingState signaling_state);
+ #ifdef HAVE_NOT
+
// Signals from MediaStreamObserver.
void OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream);
@@ -205,6 +213,8 @@ class PeerConnection : public PeerConnectionInterface,
void OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream);
+ #endif // HAVE_NOT
+
rtc::Thread* signaling_thread() const {
return factory_->signaling_thread();
}
diff --git a/webrtc/api/peerconnectionfactory.cc b/webrtc/api/peerconnectionfactory.cc
index 9cb5b46..60b4bde 100644
--- a/webrtc/api/peerconnectionfactory.cc
+++ b/webrtc/api/peerconnectionfactory.cc
@@ -170,12 +170,18 @@ bool PeerConnectionFactory::Initialize() {
return false;
}
+ cricket::MediaEngineInterface* media_engine = nullptr;
+
+ #ifdef HAVE_NOT
+
// TODO: Need to make sure only one VoE is created inside
// WebRtcMediaEngine.
- cricket::MediaEngineInterface* media_engine =
+ media_engine =
worker_thread_->Invoke<cricket::MediaEngineInterface*>(rtc::Bind(
&PeerConnectionFactory::CreateMediaEngine_w, this));
+ #endif // HAVE_NOT
+
channel_manager_.reset(new cricket::ChannelManager(
media_engine, worker_thread_, network_thread_));
@@ -190,6 +196,8 @@ bool PeerConnectionFactory::Initialize() {
return true;
}
+#ifdef HAVE_NOT
+
rtc::scoped_refptr<AudioSourceInterface>
PeerConnectionFactory::CreateAudioSource(
const MediaConstraintsInterface* constraints) {
@@ -228,6 +236,8 @@ PeerConnectionFactory::CreateVideoSource(cricket::VideoCapturer* capturer) {
source);
}
+#endif // HAVE_NOT
+
bool PeerConnectionFactory::StartAecDump(rtc::PlatformFile file,
int64_t max_size_bytes) {
RTC_DCHECK(signaling_thread_->IsCurrent());
@@ -301,6 +311,8 @@ PeerConnectionFactory::CreatePeerConnection(
return PeerConnectionProxy::Create(signaling_thread(), pc);
}
+#ifdef HAVE_NOT
+
rtc::scoped_refptr<MediaStreamInterface>
PeerConnectionFactory::CreateLocalMediaStream(const std::string& label) {
RTC_DCHECK(signaling_thread_->IsCurrent());
@@ -325,6 +337,8 @@ PeerConnectionFactory::CreateAudioTrack(const std::string& id,
return AudioTrackProxy::Create(signaling_thread_, track);
}
+#endif // HAVE_NOT
+
webrtc::MediaControllerInterface* PeerConnectionFactory::CreateMediaController(
const cricket::MediaConfig& config) const {
RTC_DCHECK(signaling_thread_->IsCurrent());
@@ -347,6 +361,8 @@ rtc::Thread* PeerConnectionFactory::network_thread() {
return network_thread_;
}
+#ifdef HAVE_NOT
+
cricket::MediaEngineInterface* PeerConnectionFactory::CreateMediaEngine_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
return cricket::WebRtcMediaEngineFactory::Create(
@@ -354,4 +370,6 @@ cricket::MediaEngineInterface* PeerConnectionFactory::CreateMediaEngine_w() {
video_decoder_factory_.get());
}
+#endif // HAVE_NOT
+
} // namespace webrtc
diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h
index 21165cf..26fb972 100644
--- a/webrtc/api/peerconnectionfactory.h
+++ b/webrtc/api/peerconnectionfactory.h
@@ -54,6 +54,8 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
bool Initialize();
+ #ifdef HAVE_NOT
+
rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) override;
@@ -81,6 +83,11 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
CreateAudioTrack(const std::string& id,
AudioSourceInterface* audio_source) override;
+ #endif // HAVE_NOT
+
+ virtual webrtc::MediaControllerInterface* CreateMediaController(
+ const cricket::MediaConfig& config) const;
+
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
void StopAecDump() override;
bool StartRtcEventLog(rtc::PlatformFile file) override {
@@ -90,8 +97,6 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
int64_t max_size_bytes) override;
void StopRtcEventLog() override;
- virtual webrtc::MediaControllerInterface* CreateMediaController(
- const cricket::MediaConfig& config) const;
virtual rtc::Thread* signaling_thread();
virtual rtc::Thread* worker_thread();
virtual rtc::Thread* network_thread();
@@ -109,7 +114,9 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
virtual ~PeerConnectionFactory();
private:
+ #ifdef HAVE_NOT
cricket::MediaEngineInterface* CreateMediaEngine_w();
+ #endif // HAVE_NOT
bool owns_ptrs_;
bool wraps_current_thread_;
diff --git a/webrtc/api/peerconnectionfactoryproxy.h b/webrtc/api/peerconnectionfactoryproxy.h
index c357de9..0a75d1f 100644
--- a/webrtc/api/peerconnectionfactoryproxy.h
+++ b/webrtc/api/peerconnectionfactoryproxy.h
@@ -46,6 +46,9 @@ BEGIN_SIGNALING_PROXY_MAP(PeerConnectionFactory)
rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot,
this, a1, a3.release(), a4.release(), a5));
}
+
+ #ifdef HAVE_NOT
+
PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
CreateLocalMediaStream, const std::string&)
PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
@@ -66,6 +69,9 @@ BEGIN_SIGNALING_PROXY_MAP(PeerConnectionFactory)
VideoTrackSourceInterface*)
PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
CreateAudioTrack, const std::string&, AudioSourceInterface*)
+
+ #endif // HAVE_NOT
+
PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
PROXY_METHOD0(void, StopAecDump)
PROXY_METHOD1(bool, StartRtcEventLog, rtc::PlatformFile)
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index 4fa9bf2..26055d4 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -344,6 +344,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
virtual rtc::scoped_refptr<StreamCollectionInterface>
remote_streams() = 0;
+ #ifdef HAVE_NOT
+
// Add a new MediaStream to be sent on this PeerConnection.
// Note that a SessionDescription negotiation is needed before the
// remote peer can receive the stream.
@@ -386,6 +388,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
return rtc::scoped_refptr<RtpSenderInterface>();
}
+ #endif // HAVE_NOT
+
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const {
return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
@@ -589,6 +593,8 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) = 0;
+ #ifdef HAVE_NOT
+
virtual rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) = 0;
@@ -623,6 +629,8 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
CreateAudioTrack(const std::string& label,
AudioSourceInterface* source) = 0;
+ #endif // HAVE_NOT
+
// Starts AEC dump using existing file. Takes ownership of |file| and passes
// it on to VoiceEngine (via other objects) immediately, which will take
// the ownerhip. If the operation fails, the file will be closed.
diff --git a/webrtc/api/peerconnectionproxy.h b/webrtc/api/peerconnectionproxy.h
index d35d5ba..eb74042 100644
--- a/webrtc/api/peerconnectionproxy.h
+++ b/webrtc/api/peerconnectionproxy.h
@@ -22,6 +22,9 @@ BEGIN_SIGNALING_PROXY_MAP(PeerConnection)
local_streams)
PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
remote_streams)
+
+ #ifdef HAVE_NOT
+
PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
@@ -35,6 +38,9 @@ BEGIN_SIGNALING_PROXY_MAP(PeerConnection)
CreateSender,
const std::string&,
const std::string&)
+
+ #endif // HAVE_NOT
+
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
GetSenders)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
diff --git a/webrtc/api/statscollector.cc b/webrtc/api/statscollector.cc
index df1d0aa..ec69b8f 100644
--- a/webrtc/api/statscollector.cc
+++ b/webrtc/api/statscollector.cc
@@ -847,6 +847,8 @@ void StatsCollector::ExtractVideoInfo(
void StatsCollector::ExtractSenderInfo() {
RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
+ #ifdef HAVE_NOT
+
for (const auto& sender : pc_->GetSenders()) {
// TODO(nisse): SSRC == 0 currently means none. Delete check when
// that is fixed.
@@ -875,6 +877,8 @@ void StatsCollector::ExtractSenderInfo() {
report->AddInt(StatsReport::kStatsValueNameFrameHeightInput,
stats.input_height);
}
+
+ #endif // HAVE_NOT
}
void StatsCollector::ExtractDataInfo() {
diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi
index 26acf4b..57c02eb 100644
--- a/webrtc/build/common.gypi
+++ b/webrtc/build/common.gypi
@@ -69,6 +69,7 @@
},
'build_with_chromium%': '<(build_with_chromium)',
'build_with_mozilla%': '<(build_with_mozilla)',
+ 'build_have_not%': 0,
'build_libevent%': '<(build_libevent)',
'enable_libevent%': '<(enable_libevent)',
'webrtc_root%': '<(webrtc_root)',
diff --git a/webrtc/call.h b/webrtc/call.h
index 80134fa..7cbe6ba 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -99,6 +99,8 @@ class Call {
static Call* Create(const Call::Config& config);
+ #ifdef HAVE_NOT
+
virtual AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) = 0;
virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
@@ -118,6 +120,8 @@ class Call {
virtual void DestroyVideoReceiveStream(
VideoReceiveStream* receive_stream) = 0;
+ #endif // HAVE_NOT
+
// All received RTP and RTCP packets for the call should be inserted to this
// PacketReceiver. The PacketReceiver pointer is valid as long as the
// Call instance exists.
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index f7c66db..a3f18dc 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -61,6 +61,8 @@ class Call : public webrtc::Call,
PacketReceiver* Receiver() override;
+ #ifdef HAVE_NOT
+
webrtc::AudioSendStream* CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) override;
void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
@@ -80,6 +82,8 @@ class Call : public webrtc::Call,
void DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) override;
+ #endif // HAVE_NOT
+
Stats GetStats() const override;
DeliveryStatus DeliverPacket(MediaType media_type,
@@ -112,11 +116,13 @@ class Call : public webrtc::Call,
EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
VoiceEngine* voice_engine() {
+ #ifdef HAVE_NOT
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(config_.audio_state.get());
if (audio_state)
return audio_state->voice_engine();
else
+ #endif // HAVE_NOT
return nullptr;
}
@@ -223,11 +229,16 @@ Call::Call(const Call::Config& config)
RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
config.bitrate_config.start_bitrate_bps);
}
+
+ #ifdef HAVE_NOT
+
if (config.audio_state.get()) {
ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
event_log_ = voe_codec->GetEventLog();
}
+ #endif // HAVE_NOT
+
Trace::CreateTrace();
call_stats_->RegisterStatsObserver(congestion_controller_.get());
@@ -323,6 +334,8 @@ PacketReceiver* Call::Receiver() {
return this;
}
+#ifdef HAVE_NOT
+
webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
@@ -522,6 +535,8 @@ void Call::DestroyVideoReceiveStream(
delete receive_stream_impl;
}
+#endif // HAVE_NOT
+
Call::Stats Call::GetStats() const {
// TODO(solenberg): Some test cases in EndToEndTest use this from a different
// thread. Re-enable once that is fixed.
@@ -581,6 +596,9 @@ void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
}
UpdateAggregateNetworkState();
+
+ #ifdef HAVE_NOT
+
{
ReadLockScoped read_lock(*send_crit_);
for (auto& kv : audio_send_ssrcs_) {
@@ -599,6 +617,9 @@ void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
kv.second->SignalNetworkState(video_network_state_);
}
}
+
+ #endif // HAVE_NOT
+
}
// TODO(honghaiz): Add tests for this method.
@@ -684,8 +705,10 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
if (video_send_streams_.empty())
return;
+ #ifdef HAVE_NOT
for (VideoSendStream* stream : video_send_streams_)
pad_up_to_bitrate_bps += stream->GetPaddingNeededBps();
+ #endif // HAVE_NOT
}
// Allocated bitrate might be higher than bitrate estimate if enforcing min
// bitrate, or lower if estimate is higher than the sum of max bitrates, so
@@ -709,6 +732,8 @@ void Call::ConfigureSync(const std::string& sync_group) {
if (voice_engine() == nullptr || sync_group.empty())
return;
+ #ifdef HAVE_NOT
+
AudioReceiveStream* sync_audio_stream = nullptr;
// Find existing audio stream.
const auto it = sync_stream_mapping_.find(sync_group);
@@ -750,6 +775,9 @@ void Call::ConfigureSync(const std::string& sync_group) {
video_stream->SetSyncChannel(voice_engine(), -1);
}
}
+
+ #endif // HAVE_NOT
+
}
PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
@@ -761,6 +789,9 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
// there's no receiver of the packet.
received_rtcp_bytes_ += length;
bool rtcp_delivered = false;
+
+ #ifdef HAVE_NOT
+
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*receive_crit_);
for (VideoReceiveStream* stream : video_receive_streams_) {
@@ -790,6 +821,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
}
}
+ #endif // HAVE_NOT
+
if (event_log_ && rtcp_delivered)
event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
@@ -809,8 +842,11 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
if (first_rtp_packet_received_ms_ == -1)
first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_;
+ #ifdef HAVE_NOT
+
uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
ReadLockScoped read_lock(*receive_crit_);
+
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
@@ -835,6 +871,9 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
return status;
}
}
+
+ #endif // HAVE_NOT
+
return DELIVERY_UNKNOWN_SSRC;
}
diff --git a/webrtc/call/webrtc_call.gypi b/webrtc/call/webrtc_call.gypi
index 455a11a..351f1b5 100644
--- a/webrtc/call/webrtc_call.gypi
+++ b/webrtc/call/webrtc_call.gypi
@@ -15,10 +15,10 @@
'<(webrtc_root)/webrtc.gyp:rtc_event_log',
],
'webrtc_call_sources': [
- 'call/bitrate_allocator.cc',
- 'call/call.cc',
- 'call/transport_adapter.cc',
- 'call/transport_adapter.h',
+ '<(webrtc_root)/call/bitrate_allocator.cc',
+ '<(webrtc_root)/call/call.cc',
+ '<(webrtc_root)/call/transport_adapter.cc',
+ '<(webrtc_root)/call/transport_adapter.h',
],
},
}
diff --git a/webrtc/media/media.gyp b/webrtc/media/media.gyp
index deb730a..60bb7e4 100644
--- a/webrtc/media/media.gyp
+++ b/webrtc/media/media.gyp
@@ -7,9 +7,80 @@
# be found in the AUTHORS file in the root of the source tree.
{
- 'includes': [ '../build/common.gypi', ],
+ 'includes': [
+ '../build/common.gypi',
+ '../call/webrtc_call.gypi',
+ ],
'targets': [
{
+ 'target_name': 'rtc_media_not',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/p2p/p2p.gyp:rtc_p2p',
+ '<@(webrtc_call_dependencies)',
+ ],
+ 'sources': [
+ '<(webrtc_root)/call.h',
+ '<@(webrtc_call_sources)',
+ '<(webrtc_root)/video/call_stats.cc',
+ '<(webrtc_root)/video/call_stats.h',
+ '<(webrtc_root)/video/send_delay_stats.cc',
+ '<(webrtc_root)/video/send_delay_stats.h',
+ '<(webrtc_root)/video/vie_remb.cc',
+ '<(webrtc_root)/video/vie_remb.h',
+ '<(webrtc_root)/modules/utility/include/process_thread.h',
+ '<(webrtc_root)/modules/utility/source/process_thread_impl.cc',
+ '<(webrtc_root)/modules/utility/source/process_thread_impl.h',
+ 'base/codec.cc',
+ 'base/codec.h',
+ 'base/mediachannel.h',
+ 'base/mediaconstants.cc',
+ 'base/mediaconstants.h',
+ 'base/mediaengine.cc',
+ 'base/mediaengine.h',
+ 'base/rtpdataengine.cc',
+ 'base/rtpdataengine.h',
+ 'base/rtputils.cc',
+ 'base/rtputils.h',
+ 'base/streamparams.cc',
+ 'base/streamparams.h',
+ 'base/turnutils.cc',
+ 'base/turnutils.h',
+ 'sctp/sctpdataengine.cc',
+ 'sctp/sctpdataengine.h',
+ ],
+ # TODO(kjellander): Make the code compile without disabling these flags.
+ # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
+ 'cflags': [
+ '-Wno-deprecated-declarations',
+ ],
+ 'cflags!': [
+ '-Wextra',
+ ],
+ 'cflags_cc!': [
+ '-Woverloaded-virtual',
+ ],
+ 'msvs_disabled_warnings': [
+ 4245, # conversion from 'int' to 'size_t', signed/unsigned mismatch.
+ 4267, # conversion from 'size_t' to 'int', possible loss of data.
+ 4389, # signed/unsigned mismatch.
+ ],
+ 'conditions': [
+ ['build_usrsctp==1', {
+ 'include_dirs': [
+ # TODO(jiayl): move this into the direct_dependent_settings of
+ # usrsctp.gyp.
+ '<(DEPTH)/third_party/usrsctp/usrsctplib',
+ ],
+ 'dependencies': [
+ '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
+ ],
+ }],
+ ],
+ }, # target rtc_media_not
+ {
'target_name': 'rtc_media',
'type': 'static_library',
'dependencies': [
diff --git a/webrtc/pc/channelmanager.cc b/webrtc/pc/channelmanager.cc
index 56dd135..06f3892 100644
--- a/webrtc/pc/channelmanager.cc
+++ b/webrtc/pc/channelmanager.cc
@@ -153,6 +153,8 @@ bool ChannelManager::Init() {
rtc::Bind(&rtc::Thread::SetAllowBlockingCalls, network_thread_, false));
}
+ #ifdef HAVE_NOT
+
initialized_ = worker_thread_->Invoke<bool>(
Bind(&ChannelManager::InitMediaEngine_w, this));
ASSERT(initialized_);
@@ -168,6 +170,10 @@ bool ChannelManager::Init() {
<< audio_output_volume_;
}
+ #else
+ initialized_ = true;
+ #endif // HAVE_NOT
+
return initialized_;
}
diff --git a/webrtc/pc/mediasession.cc b/webrtc/pc/mediasession.cc
index 0fa20d8..f57471a 100644
--- a/webrtc/pc/mediasession.cc
+++ b/webrtc/pc/mediasession.cc
@@ -1234,11 +1234,13 @@ MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
: secure_(SEC_DISABLED),
add_legacy_(true),
transport_desc_factory_(transport_desc_factory) {
+ #ifdef HAVE_NOT
channel_manager->GetSupportedAudioCodecs(&audio_codecs_);
channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_);
channel_manager->GetSupportedVideoCodecs(&video_codecs_);
channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_);
channel_manager->GetSupportedDataCodecs(&data_codecs_);
+ #endif // HAVE_NOT
}
SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
diff --git a/webrtc/pc/pc.gyp b/webrtc/pc/pc.gyp
index 25ebc5d..59330c1 100755
--- a/webrtc/pc/pc.gyp
+++ b/webrtc/pc/pc.gyp
@@ -21,7 +21,7 @@
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
- '<(webrtc_root)/media/media.gyp:rtc_media',
+ # '<(webrtc_root)/media/media.gyp:rtc_media',
],
'conditions': [
['build_libsrtp==1', {
diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp
index 793bf96..51fe339 100644
--- a/webrtc/webrtc.gyp
+++ b/webrtc/webrtc.gyp
@@ -35,7 +35,7 @@
],
},
}],
- ['build_with_chromium==0 and'
+ ['build_with_chromium==0 and build_have_not==1 and'
'(OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7"))', {
# TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/
# has been moved to webrtc/. It can't be processed by Chromium since the
--
2.9.0
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