Create //webrtc/api:libjingle_peerconnection_api + refactorings.
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 5633f28..e067791 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -284,7 +284,6 @@
":video_engine_tests",
":webrtc_nonparallel_tests",
":webrtc_perf_tests",
- "api:peerconnection_unittests",
"base:rtc_base_tests_utils",
"common_audio:common_audio_unittests",
"common_video:common_video_unittests",
@@ -295,6 +294,7 @@
"modules/audio_processing:audio_processing_tests",
"modules/rtp_rtcp:test_packet_masks_metrics",
"modules/video_capture:video_capture_internal_impl",
+ "pc:peerconnection_unittests",
"pc:rtc_pc_unittests",
"stats:rtc_stats_unittests",
"system_wrappers:system_wrappers_unittests",
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index fa3c629..9d893e0 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -14,7 +14,7 @@
group("api") {
public_deps = [
- ":libjingle_peerconnection",
+ ":libjingle_peerconnection_api",
]
}
@@ -34,122 +34,63 @@
]
}
-config("libjingle_peerconnection_warnings_config") {
- # GN orders flags on a target before flags from configs. The default config
- # adds these flags so to cancel them out they need to come from a config and
- # cannot be on the target directly.
- if (!is_win && !is_clang) {
- cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
- }
-}
-
-rtc_static_library("libjingle_peerconnection") {
+rtc_static_library("libjingle_peerconnection_api") {
check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828)
cflags = []
sources = [
- "audiotrack.cc",
- "audiotrack.h",
- "datachannel.cc",
"datachannel.h",
"datachannelinterface.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.cc",
"mediastreaminterface.h",
- "mediastreamobserver.cc",
- "mediastreamobserver.h",
"mediastreamproxy.h",
"mediastreamtrack.h",
"mediastreamtrackproxy.h",
+ "mediatypes.cc",
+ "mediatypes.h",
"notifier.h",
- "ortcfactory.cc",
- "ortcfactory.h",
"ortcfactoryinterface.h",
- "peerconnection.cc",
- "peerconnection.h",
- "peerconnectionfactory.cc",
- "peerconnectionfactory.h",
"peerconnectionfactoryproxy.h",
"peerconnectioninterface.h",
"peerconnectionproxy.h",
"proxy.h",
- "remoteaudiosource.cc",
- "remoteaudiosource.h",
- "rtcstatscollector.cc",
- "rtcstatscollector.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",
"trackmediainfomap.cc",
"trackmediainfomap.h",
"udptransportinterface.h",
- "videocapturertracksource.cc",
- "videocapturertracksource.h",
+ "umametrics.h",
"videosourceproxy.h",
- "videotrack.cc",
- "videotrack.h",
- "videotracksource.cc",
"videotracksource.h",
- "webrtcsdp.cc",
- "webrtcsdp.h",
- "webrtcsession.cc",
- "webrtcsession.h",
- "webrtcsessiondescriptionfactory.cc",
- "webrtcsessiondescriptionfactory.h",
]
- configs += [ ":libjingle_peerconnection_warnings_config" ]
-
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
- ":call_api",
":rtc_stats_api",
- "../call",
- "../media",
- "../pc",
- "../stats",
]
+}
- if (rtc_use_quic) {
- sources += [
- "quicdatachannel.cc",
- "quicdatachannel.h",
- "quicdatatransport.cc",
- "quicdatatransport.h",
- ]
- deps += [ "//third_party/libquic" ]
- public_deps = [
- "//third_party/libquic",
- ]
- }
+# TODO(ossu): Remove once downstream projects have updated.
+rtc_source_set("libjingle_peerconnection") {
+ deps = [
+ "../pc:libjingle_peerconnection",
+ ]
}
rtc_source_set("rtc_stats_api") {
@@ -157,6 +98,7 @@
sources = [
"stats/rtcstats.h",
"stats/rtcstats_objects.h",
+ "stats/rtcstatscollectorcallback.h",
"stats/rtcstatsreport.h",
]
@@ -210,134 +152,6 @@
}
if (rtc_include_tests) {
- config("peerconnection_unittests_config") {
- # The warnings below are enabled by default. Since GN orders compiler flags
- # for a target before flags from configs, the only way to disable such
- # warnings is by having them in a separate config, loaded from the target.
- # TODO(kjellander): Make the code compile without disabling these flags.
- # See https://bugs.webrtc.org/3307.
- if (is_clang && is_win) {
- cflags = [
- # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
- # for -Wno-sign-compare
- "-Wno-sign-compare",
- "-Wno-unused-function",
- ]
- }
-
- if (!is_win) {
- cflags = [ "-Wno-sign-compare" ]
- }
- }
-
- rtc_test("peerconnection_unittests") {
- check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828)
- testonly = true
- sources = [
- "datachannel_unittest.cc",
- "dtmfsender_unittest.cc",
- "jsepsessiondescription_unittest.cc",
- "localaudiosource_unittest.cc",
- "mediaconstraintsinterface_unittest.cc",
- "mediastream_unittest.cc",
- "ortcfactory_unittest.cc",
- "peerconnection_unittest.cc",
- "peerconnectionendtoend_unittest.cc",
- "peerconnectionfactory_unittest.cc",
- "peerconnectioninterface_unittest.cc",
- "proxy_unittest.cc",
- "rtcstats_integrationtest.cc",
- "rtcstatscollector_unittest.cc",
- "rtpsenderreceiver_unittest.cc",
- "sctputils_unittest.cc",
- "statscollector_unittest.cc",
- "test/fakeaudiocapturemodule.cc",
- "test/fakeaudiocapturemodule.h",
- "test/fakeaudiocapturemodule_unittest.cc",
- "test/fakeconstraints.h",
- "test/fakedatachannelprovider.h",
- "test/fakeperiodicvideocapturer.h",
- "test/fakertccertificategenerator.h",
- "test/fakevideotrackrenderer.h",
- "test/mock_datachannel.h",
- "test/mock_peerconnection.h",
- "test/mock_rtpreceiver.h",
- "test/mock_rtpsender.h",
- "test/mock_webrtcsession.h",
- "test/mockpeerconnectionobservers.h",
- "test/peerconnectiontestwrapper.cc",
- "test/peerconnectiontestwrapper.h",
- "test/rtcstatsobtainer.h",
- "test/testsdpstrings.h",
- "trackmediainfomap_unittest.cc",
- "videocapturertracksource_unittest.cc",
- "videotrack_unittest.cc",
- "webrtcsdp_unittest.cc",
- "webrtcsession_unittest.cc",
- ]
-
- if (rtc_enable_sctp) {
- defines = [ "HAVE_SCTP" ]
- }
-
- configs += [ ":peerconnection_unittests_config" ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-
- # TODO(jschuh): Bug 1348: fix this warning.
- configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
-
- if (is_win) {
- cflags = [
- "/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
- "/wd4389", # signed/unsigned mismatch.
- ]
- }
-
- if (rtc_use_quic) {
- public_deps = [
- "//third_party/libquic",
- ]
- sources += [
- "quicdatachannel_unittest.cc",
- "quicdatatransport_unittest.cc",
- ]
- }
-
- deps = []
- if (is_android) {
- sources += [
- "test/androidtestinitializer.cc",
- "test/androidtestinitializer.h",
- ]
- deps += [
- "//testing/android/native_test:native_test_support",
- "//webrtc/sdk/android:libjingle_peerconnection_java",
- "//webrtc/sdk/android:libjingle_peerconnection_jni",
- ]
- }
-
- deps += [
- ":fakemetricsobserver",
- ":libjingle_peerconnection",
- "..:webrtc_common",
- "../base:rtc_base_tests_utils",
- "../media:rtc_unittest_main",
- "../pc:rtc_pc",
- "../system_wrappers:metrics_default",
- "//testing/gmock",
- ]
-
- if (is_android) {
- deps += [ "//testing/android/native_test:native_test_support" ]
-
- shard_timeout = 900
- }
- }
-
rtc_source_set("mock_audio_mixer") {
testonly = true
sources = [
@@ -354,6 +168,22 @@
]
}
+ rtc_source_set("libjingle_peerconnection_test_api") {
+ testonly = true
+ sources = [
+ "test/fakeconstraints.h",
+ ]
+
+ public_deps = [
+ ":libjingle_peerconnection_api",
+ ]
+
+ deps = [
+ "../base:rtc_base_approved",
+ "//webrtc/test:test_support",
+ ]
+ }
+
rtc_source_set("fakemetricsobserver") {
testonly = true
sources = [
@@ -361,7 +191,7 @@
"fakemetricsobserver.h",
]
deps = [
- ":libjingle_peerconnection",
+ ":libjingle_peerconnection_api",
"../base:rtc_base_approved",
]
if (!build_with_chromium && is_clang) {
diff --git a/webrtc/api/DEPS b/webrtc/api/DEPS
index 75f756d..01d719a 100644
--- a/webrtc/api/DEPS
+++ b/webrtc/api/DEPS
@@ -5,35 +5,10 @@
"+webrtc/media",
"+webrtc/p2p",
"+webrtc/pc",
- "+webrtc/logging/rtc_event_log",
- "+webrtc/modules/audio_device",
- "+webrtc/modules/rtp_rtcp",
- "+webrtc/modules/video_coding",
- "+webrtc/modules/video_render",
- "+webrtc/system_wrappers",
]
specific_include_rules = {
- "androidtestinitializer\.cc": [
- "+base/android", # Allowed only for Android tests.
- "+webrtc/voice_engine",
- ],
- # The call/call.h exceptions are here only until the peerconnection
- # implementation has been moved out of api/. See:
- # http://bugs.webrtc.org/5883
- "mediacontroller\.cc": [
- "+webrtc/call/call.h"
- ],
- "peerconnection\.cc": [
- "+webrtc/call/call.h"
- ],
"peerconnection_jni\.cc": [
"+webrtc/voice_engine",
],
- "peerconnectionfactory\.cc": [
- "+webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h",
- ],
- "webrtcsession\.cc": [
- "+webrtc/call/call.h"
- ],
}
diff --git a/webrtc/api/datachannel.h b/webrtc/api/datachannel.h
index cdb2e62..28112ad 100644
--- a/webrtc/api/datachannel.h
+++ b/webrtc/api/datachannel.h
@@ -11,288 +11,8 @@
#ifndef WEBRTC_API_DATACHANNEL_H_
#define WEBRTC_API_DATACHANNEL_H_
-#include <deque>
-#include <set>
-#include <string>
-
-#include "webrtc/api/datachannelinterface.h"
-#include "webrtc/api/proxy.h"
-#include "webrtc/base/messagehandler.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/base/sigslot.h"
-#include "webrtc/media/base/mediachannel.h"
-#include "webrtc/pc/channel.h"
-
-namespace webrtc {
-
-class DataChannel;
-
-class DataChannelProviderInterface {
- public:
- // Sends the data to the transport.
- virtual bool SendData(const cricket::SendDataParams& params,
- const rtc::CopyOnWriteBuffer& payload,
- cricket::SendDataResult* result) = 0;
- // Connects to the transport signals.
- virtual bool ConnectDataChannel(DataChannel* data_channel) = 0;
- // Disconnects from the transport signals.
- virtual void DisconnectDataChannel(DataChannel* data_channel) = 0;
- // Adds the data channel SID to the transport for SCTP.
- virtual void AddSctpDataStream(int sid) = 0;
- // Removes the data channel SID from the transport for SCTP.
- virtual void RemoveSctpDataStream(int sid) = 0;
- // Returns true if the transport channel is ready to send data.
- virtual bool ReadyToSendData() const = 0;
-
- protected:
- virtual ~DataChannelProviderInterface() {}
-};
-
-struct InternalDataChannelInit : public DataChannelInit {
- enum OpenHandshakeRole {
- kOpener,
- kAcker,
- kNone
- };
- // The default role is kOpener because the default |negotiated| is false.
- InternalDataChannelInit() : open_handshake_role(kOpener) {}
- explicit InternalDataChannelInit(const DataChannelInit& base)
- : DataChannelInit(base), open_handshake_role(kOpener) {
- // If the channel is externally negotiated, do not send the OPEN message.
- if (base.negotiated) {
- open_handshake_role = kNone;
- }
- }
-
- OpenHandshakeRole open_handshake_role;
-};
-
-// Helper class to allocate unique IDs for SCTP DataChannels
-class SctpSidAllocator {
- public:
- // Gets the first unused odd/even id based on the DTLS role. If |role| is
- // SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
- // otherwise, the id starts from 1 and takes odd numbers.
- // Returns false if no id can be allocated.
- bool AllocateSid(rtc::SSLRole role, int* sid);
-
- // Attempts to reserve a specific sid. Returns false if it's unavailable.
- bool ReserveSid(int sid);
-
- // Indicates that |sid| isn't in use any more, and is thus available again.
- void ReleaseSid(int sid);
-
- private:
- // Checks if |sid| is available to be assigned to a new SCTP data channel.
- bool IsSidAvailable(int sid) const;
-
- std::set<int> used_sids_;
-};
-
-// DataChannel is a an implementation of the DataChannelInterface based on
-// libjingle's data engine. It provides an implementation of unreliable or
-// reliabledata channels. Currently this class is specifically designed to use
-// both RtpDataEngine and SctpDataEngine.
-
-// DataChannel states:
-// kConnecting: The channel has been created the transport might not yet be
-// ready.
-// kOpen: The channel have a local SSRC set by a call to UpdateSendSsrc
-// and a remote SSRC set by call to UpdateReceiveSsrc and the transport
-// has been writable once.
-// kClosing: DataChannelInterface::Close has been called or UpdateReceiveSsrc
-// has been called with SSRC==0
-// kClosed: Both UpdateReceiveSsrc and UpdateSendSsrc has been called with
-// SSRC==0.
-class DataChannel : public DataChannelInterface,
- public sigslot::has_slots<>,
- public rtc::MessageHandler {
- public:
- static rtc::scoped_refptr<DataChannel> Create(
- DataChannelProviderInterface* provider,
- cricket::DataChannelType dct,
- const std::string& label,
- const InternalDataChannelInit& config);
-
- virtual void RegisterObserver(DataChannelObserver* observer);
- virtual void UnregisterObserver();
-
- virtual std::string label() const { return label_; }
- virtual bool reliable() const;
- virtual bool ordered() const { return config_.ordered; }
- virtual uint16_t maxRetransmitTime() const {
- return config_.maxRetransmitTime;
- }
- virtual uint16_t maxRetransmits() const { return config_.maxRetransmits; }
- virtual std::string protocol() const { return config_.protocol; }
- virtual bool negotiated() const { return config_.negotiated; }
- virtual int id() const { return config_.id; }
- virtual uint64_t buffered_amount() const;
- virtual void Close();
- virtual DataState state() const { return state_; }
- virtual uint32_t messages_sent() const { return messages_sent_; }
- virtual uint64_t bytes_sent() const { return bytes_sent_; }
- virtual uint32_t messages_received() const { return messages_received_; }
- virtual uint64_t bytes_received() const { return bytes_received_; }
- virtual bool Send(const DataBuffer& buffer);
-
- // rtc::MessageHandler override.
- virtual void OnMessage(rtc::Message* msg);
-
- // Called when the channel's ready to use. That can happen when the
- // underlying DataMediaChannel becomes ready, or when this channel is a new
- // stream on an existing DataMediaChannel, and we've finished negotiation.
- void OnChannelReady(bool writable);
-
- // Slots for provider to connect signals to.
- void OnDataReceived(const cricket::ReceiveDataParams& params,
- const rtc::CopyOnWriteBuffer& payload);
- void OnStreamClosedRemotely(int sid);
-
- // The remote peer request that this channel should be closed.
- void RemotePeerRequestClose();
-
- // The following methods are for SCTP only.
-
- // Sets the SCTP sid and adds to transport layer if not set yet. Should only
- // be called once.
- void SetSctpSid(int sid);
- // Called when the transport channel is created.
- // Only needs to be called for SCTP data channels.
- void OnTransportChannelCreated();
- // Called when the transport channel is destroyed.
- // This method makes sure the DataChannel is disconnected and changes state
- // to kClosed.
- void OnTransportChannelDestroyed();
-
- // The following methods are for RTP only.
-
- // Set the SSRC this channel should use to send data on the
- // underlying data engine. |send_ssrc| == 0 means that the channel is no
- // longer part of the session negotiation.
- void SetSendSsrc(uint32_t send_ssrc);
- // Set the SSRC this channel should use to receive data from the
- // underlying data engine.
- void SetReceiveSsrc(uint32_t receive_ssrc);
-
- cricket::DataChannelType data_channel_type() const {
- return data_channel_type_;
- }
-
- // Emitted when state transitions to kOpen.
- sigslot::signal1<DataChannel*> SignalOpened;
- // Emitted when state transitions to kClosed.
- // In the case of SCTP channels, this signal can be used to tell when the
- // channel's sid is free.
- sigslot::signal1<DataChannel*> SignalClosed;
-
- protected:
- DataChannel(DataChannelProviderInterface* client,
- cricket::DataChannelType dct,
- const std::string& label);
- virtual ~DataChannel();
-
- private:
- // A packet queue which tracks the total queued bytes. Queued packets are
- // owned by this class.
- class PacketQueue {
- public:
- PacketQueue();
- ~PacketQueue();
-
- size_t byte_count() const {
- return byte_count_;
- }
-
- bool Empty() const;
-
- DataBuffer* Front();
-
- void Pop();
-
- void Push(DataBuffer* packet);
-
- void Clear();
-
- void Swap(PacketQueue* other);
-
- private:
- std::deque<DataBuffer*> packets_;
- size_t byte_count_;
- };
-
- // The OPEN(_ACK) signaling state.
- enum HandshakeState {
- kHandshakeInit,
- kHandshakeShouldSendOpen,
- kHandshakeShouldSendAck,
- kHandshakeWaitingForAck,
- kHandshakeReady
- };
-
- bool Init(const InternalDataChannelInit& config);
- void DoClose();
- void UpdateState();
- void SetState(DataState state);
- void DisconnectFromProvider();
-
- void DeliverQueuedReceivedData();
-
- void SendQueuedDataMessages();
- bool SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked);
- bool QueueSendDataMessage(const DataBuffer& buffer);
-
- void SendQueuedControlMessages();
- void QueueControlMessage(const rtc::CopyOnWriteBuffer& buffer);
- bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer);
-
- std::string label_;
- InternalDataChannelInit config_;
- DataChannelObserver* observer_;
- DataState state_;
- uint32_t messages_sent_;
- uint64_t bytes_sent_;
- uint32_t messages_received_;
- uint64_t bytes_received_;
- cricket::DataChannelType data_channel_type_;
- DataChannelProviderInterface* provider_;
- HandshakeState handshake_state_;
- bool connected_to_provider_;
- bool send_ssrc_set_;
- bool receive_ssrc_set_;
- bool writable_;
- uint32_t send_ssrc_;
- uint32_t receive_ssrc_;
- // Control messages that always have to get sent out before any queued
- // data.
- PacketQueue queued_control_data_;
- PacketQueue queued_received_data_;
- PacketQueue queued_send_data_;
-};
-
-// Define proxy for DataChannelInterface.
-BEGIN_SIGNALING_PROXY_MAP(DataChannel)
- PROXY_SIGNALING_THREAD_DESTRUCTOR()
- PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
- PROXY_METHOD0(void, UnregisterObserver)
- PROXY_CONSTMETHOD0(std::string, label)
- PROXY_CONSTMETHOD0(bool, reliable)
- PROXY_CONSTMETHOD0(bool, ordered)
- PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime)
- PROXY_CONSTMETHOD0(uint16_t, maxRetransmits)
- PROXY_CONSTMETHOD0(std::string, protocol)
- PROXY_CONSTMETHOD0(bool, negotiated)
- PROXY_CONSTMETHOD0(int, id)
- PROXY_CONSTMETHOD0(DataState, state)
- PROXY_CONSTMETHOD0(uint32_t, messages_sent)
- PROXY_CONSTMETHOD0(uint64_t, bytes_sent)
- PROXY_CONSTMETHOD0(uint32_t, messages_received)
- PROXY_CONSTMETHOD0(uint64_t, bytes_received)
- PROXY_CONSTMETHOD0(uint64_t, buffered_amount)
- PROXY_METHOD0(void, Close)
- PROXY_METHOD1(bool, Send, const DataBuffer&)
-END_PROXY_MAP()
-
-} // namespace webrtc
+// Including this file is deprecated. It is no longer part of the public API.
+// This only includes the file in its new location for backwards compatibility.
+#include "webrtc/pc/datachannel.h"
#endif // WEBRTC_API_DATACHANNEL_H_
diff --git a/webrtc/api/mediacontroller.h b/webrtc/api/mediacontroller.h
index 5ada20d..9c2b708 100644
--- a/webrtc/api/mediacontroller.h
+++ b/webrtc/api/mediacontroller.h
@@ -11,34 +11,8 @@
#ifndef WEBRTC_API_MEDIACONTROLLER_H_
#define WEBRTC_API_MEDIACONTROLLER_H_
-#include "webrtc/base/thread.h"
-
-namespace cricket {
-class ChannelManager;
-struct MediaConfig;
-} // namespace cricket
-
-namespace webrtc {
-class Call;
-class VoiceEngine;
-class RtcEventLog;
-
-// The MediaController currently owns shared state between media channels, but
-// in the future will create and own RtpSenders and RtpReceivers.
-class MediaControllerInterface {
- public:
- static MediaControllerInterface* Create(
- const cricket::MediaConfig& config,
- rtc::Thread* worker_thread,
- cricket::ChannelManager* channel_manager,
- webrtc::RtcEventLog* event_log);
-
- virtual ~MediaControllerInterface() {}
- virtual void Close() = 0;
- virtual webrtc::Call* call_w() = 0;
- virtual cricket::ChannelManager* channel_manager() const = 0;
- virtual const cricket::MediaConfig& config() const = 0;
-};
-} // namespace webrtc
+// Including this file is deprecated. It is no longer part of the public API.
+// This only includes the file in its new location for backwards compatibility.
+#include "webrtc/pc/mediacontroller.h"
#endif // WEBRTC_API_MEDIACONTROLLER_H_
diff --git a/webrtc/api/mediastream.h b/webrtc/api/mediastream.h
index 1f80f25..e6556c5 100644
--- a/webrtc/api/mediastream.h
+++ b/webrtc/api/mediastream.h
@@ -1,5 +1,5 @@
/*
- * Copyright 2011 The WebRTC project authors. All Rights Reserved.
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -8,51 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// This file contains the implementation of MediaStreamInterface interface.
-
#ifndef WEBRTC_API_MEDIASTREAM_H_
#define WEBRTC_API_MEDIASTREAM_H_
-#include <string>
-#include <vector>
-
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/notifier.h"
-
-namespace webrtc {
-
-class MediaStream : public Notifier<MediaStreamInterface> {
- public:
- static rtc::scoped_refptr<MediaStream> Create(const std::string& label);
-
- std::string label() const override { return label_; }
-
- bool AddTrack(AudioTrackInterface* track) override;
- bool AddTrack(VideoTrackInterface* track) override;
- bool RemoveTrack(AudioTrackInterface* track) override;
- bool RemoveTrack(VideoTrackInterface* track) override;
- rtc::scoped_refptr<AudioTrackInterface>
- FindAudioTrack(const std::string& track_id) override;
- rtc::scoped_refptr<VideoTrackInterface>
- FindVideoTrack(const std::string& track_id) override;
-
- AudioTrackVector GetAudioTracks() override { return audio_tracks_; }
- VideoTrackVector GetVideoTracks() override { return video_tracks_; }
-
- protected:
- explicit MediaStream(const std::string& label);
-
- private:
- template <typename TrackVector, typename Track>
- bool AddTrack(TrackVector* Tracks, Track* track);
- template <typename TrackVector>
- bool RemoveTrack(TrackVector* Tracks, MediaStreamTrackInterface* track);
-
- std::string label_;
- AudioTrackVector audio_tracks_;
- VideoTrackVector video_tracks_;
-};
-
-} // namespace webrtc
+// Including this file is deprecated. It is no longer part of the public API.
+// This only includes the file in its new location for backwards compatibility.
+#include "webrtc/pc/mediastream.h"
#endif // WEBRTC_API_MEDIASTREAM_H_
diff --git a/webrtc/api/test/androidtestinitializer.h b/webrtc/api/mediastreaminterface.cc
similarity index 60%
copy from webrtc/api/test/androidtestinitializer.h
copy to webrtc/api/mediastreaminterface.cc
index b56783e..d4fe5cb 100644
--- a/webrtc/api/test/androidtestinitializer.h
+++ b/webrtc/api/mediastreaminterface.cc
@@ -1,5 +1,5 @@
/*
- * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ * Copyright 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -8,13 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_
-#define WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_
+#include "webrtc/api/mediastreaminterface.h"
namespace webrtc {
-void InitializeAndroidObjects();
+const char MediaStreamTrackInterface::kVideoKind[] = "video";
+const char MediaStreamTrackInterface::kAudioKind[] = "audio";
} // namespace webrtc
-
-#endif // WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_
diff --git a/webrtc/api/mediastreamtrack.h b/webrtc/api/mediastreamtrack.h
index 9ba0420..6075abe 100644
--- a/webrtc/api/mediastreamtrack.h
+++ b/webrtc/api/mediastreamtrack.h
@@ -11,52 +11,8 @@
#ifndef WEBRTC_API_MEDIASTREAMTRACK_H_
#define WEBRTC_API_MEDIASTREAMTRACK_H_
-#include <string>
-
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/notifier.h"
-
-namespace webrtc {
-
-// MediaTrack implements the interface common to AudioTrackInterface and
-// VideoTrackInterface.
-template <typename T>
-class MediaStreamTrack : public Notifier<T> {
- public:
- typedef typename T::TrackState TypedTrackState;
-
- std::string id() const override { return id_; }
- MediaStreamTrackInterface::TrackState state() const override {
- return state_;
- }
- bool enabled() const override { return enabled_; }
- bool set_enabled(bool enable) override {
- bool fire_on_change = (enable != enabled_);
- enabled_ = enable;
- if (fire_on_change) {
- Notifier<T>::FireOnChanged();
- }
- return fire_on_change;
- }
-
- protected:
- explicit MediaStreamTrack(const std::string& id)
- : enabled_(true), id_(id), state_(MediaStreamTrackInterface::kLive) {}
-
- bool set_state(MediaStreamTrackInterface::TrackState new_state) {
- bool fire_on_change = (state_ != new_state);
- state_ = new_state;
- if (fire_on_change)
- Notifier<T>::FireOnChanged();
- return true;
- }
-
- private:
- bool enabled_;
- std::string id_;
- MediaStreamTrackInterface::TrackState state_;
-};
-
-} // namespace webrtc
+// Including this file is deprecated. It is no longer part of the public API.
+// This only includes the file in its new location for backwards compatibility.
+#include "webrtc/pc/mediastreamtrack.h"
#endif // WEBRTC_API_MEDIASTREAMTRACK_H_
diff --git a/webrtc/api/mediatypes.cc b/webrtc/api/mediatypes.cc
new file mode 100644
index 0000000..97b0189
--- /dev/null
+++ b/webrtc/api/mediatypes.cc
@@ -0,0 +1,35 @@
+/*
+ * Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/mediatypes.h"
+#include "webrtc/base/checks.h"
+
+namespace cricket {
+
+std::string MediaTypeToString(MediaType type) {
+ std::string type_str;
+ switch (type) {
+ case MEDIA_TYPE_AUDIO:
+ type_str = "audio";
+ break;
+ case MEDIA_TYPE_VIDEO:
+ type_str = "video";
+ break;
+ case MEDIA_TYPE_DATA:
+ type_str = "data";
+ break;
+ default:
+ RTC_NOTREACHED();
+ break;
+ }
+ return type_str;
+}
+
+} // namespace cricket
diff --git a/webrtc/api/mediatypes.h b/webrtc/api/mediatypes.h
new file mode 100644
index 0000000..19acf4b
--- /dev/null
+++ b/webrtc/api/mediatypes.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_MEDIATYPES_H_
+#define WEBRTC_API_MEDIATYPES_H_
+
+#include <string>
+
+namespace cricket {
+
+enum MediaType {
+ MEDIA_TYPE_AUDIO,
+ MEDIA_TYPE_VIDEO,
+ MEDIA_TYPE_DATA
+};
+
+std::string MediaTypeToString(MediaType type);
+
+} // namespace cricket
+
+#endif // WEBRTC_API_MEDIATYPES_H_
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index e622e52..e42c60f 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -61,7 +61,7 @@
#include "webrtc/api/dtmfsenderinterface.h"
#include "webrtc/api/jsep.h"
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/rtcstatscollector.h"
+#include "webrtc/api/stats/rtcstatscollectorcallback.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/api/statstypes.h"
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
index 103df47..4e47c9a 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -16,12 +16,12 @@
#include <string>
+#include "webrtc/api/mediatypes.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/pc/mediasession.h"
namespace webrtc {
diff --git a/webrtc/api/rtpsender.h b/webrtc/api/rtpsender.h
index 6c05921..3b7faec 100644
--- a/webrtc/api/rtpsender.h
+++ b/webrtc/api/rtpsender.h
@@ -8,228 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// This file contains classes that implement RtpSenderInterface.
-// An RtpSender associates a MediaStreamTrackInterface with an underlying
-// transport (provided by AudioProviderInterface/VideoProviderInterface)
-
#ifndef WEBRTC_API_RTPSENDER_H_
#define WEBRTC_API_RTPSENDER_H_
-#include <memory>
-#include <string>
-
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/rtpsenderinterface.h"
-#include "webrtc/api/statscollector.h"
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/media/base/audiosource.h"
-#include "webrtc/pc/channel.h"
-
-namespace webrtc {
-
-// Internal interface used by PeerConnection.
-class RtpSenderInternal : public RtpSenderInterface {
- public:
- // Used to set the SSRC of the sender, once a local description has been set.
- // If |ssrc| is 0, this indiates that the sender should disconnect from the
- // underlying transport (this occurs if the sender isn't seen in a local
- // description).
- virtual void SetSsrc(uint32_t ssrc) = 0;
-
- // TODO(deadbeef): Support one sender having multiple stream ids.
- virtual void set_stream_id(const std::string& stream_id) = 0;
- virtual std::string stream_id() const = 0;
-
- virtual void Stop() = 0;
-};
-
-// LocalAudioSinkAdapter receives data callback as a sink to the local
-// AudioTrack, and passes the data to the sink of AudioSource.
-class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
- public cricket::AudioSource {
- public:
- LocalAudioSinkAdapter();
- virtual ~LocalAudioSinkAdapter();
-
- private:
- // AudioSinkInterface implementation.
- void OnData(const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- size_t number_of_channels,
- size_t number_of_frames) override;
-
- // cricket::AudioSource implementation.
- void SetSink(cricket::AudioSource::Sink* sink) override;
-
- cricket::AudioSource::Sink* sink_;
- // Critical section protecting |sink_|.
- rtc::CriticalSection lock_;
-};
-
-class AudioRtpSender : public ObserverInterface,
- public rtc::RefCountedObject<RtpSenderInternal> {
- public:
- // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
- // at the appropriate times.
- // |channel| can be null if one does not exist yet.
- AudioRtpSender(AudioTrackInterface* track,
- const std::string& stream_id,
- cricket::VoiceChannel* channel,
- StatsCollector* stats);
-
- // Randomly generates stream_id.
- // |channel| can be null if one does not exist yet.
- AudioRtpSender(AudioTrackInterface* track,
- cricket::VoiceChannel* channel,
- StatsCollector* stats);
-
- // Randomly generates id and stream_id.
- // |channel| can be null if one does not exist yet.
- AudioRtpSender(cricket::VoiceChannel* channel, StatsCollector* stats);
-
- virtual ~AudioRtpSender();
-
- // ObserverInterface implementation
- void OnChanged() override;
-
- // RtpSenderInterface implementation
- bool SetTrack(MediaStreamTrackInterface* track) override;
- rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
- return track_;
- }
-
- uint32_t ssrc() const override { return ssrc_; }
-
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_AUDIO;
- }
-
- std::string id() const override { return id_; }
-
- std::vector<std::string> stream_ids() const override {
- std::vector<std::string> ret = {stream_id_};
- return ret;
- }
-
- RtpParameters GetParameters() const override;
- bool SetParameters(const RtpParameters& parameters) override;
-
- // RtpSenderInternal implementation.
- void SetSsrc(uint32_t ssrc) override;
-
- void set_stream_id(const std::string& stream_id) override {
- stream_id_ = stream_id;
- }
- std::string stream_id() const override { return stream_id_; }
-
- void Stop() override;
-
- // Does not take ownership.
- // Should call SetChannel(nullptr) before |channel| is destroyed.
- void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; }
-
- private:
- // TODO(nisse): Since SSRC == 0 is technically valid, figure out
- // some other way to test if we have a valid SSRC.
- bool can_send_track() const { return track_ && ssrc_; }
- // Helper function to construct options for
- // AudioProviderInterface::SetAudioSend.
- void SetAudioSend();
- // Helper function to call SetAudioSend with "stop sending" parameters.
- void ClearAudioSend();
-
- std::string id_;
- std::string stream_id_;
- cricket::VoiceChannel* channel_ = nullptr;
- StatsCollector* stats_;
- rtc::scoped_refptr<AudioTrackInterface> track_;
- uint32_t ssrc_ = 0;
- bool cached_track_enabled_ = false;
- bool stopped_ = false;
-
- // Used to pass the data callback from the |track_| to the other end of
- // cricket::AudioSource.
- std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
-};
-
-class VideoRtpSender : public ObserverInterface,
- public rtc::RefCountedObject<RtpSenderInternal> {
- public:
- // |channel| can be null if one does not exist yet.
- VideoRtpSender(VideoTrackInterface* track,
- const std::string& stream_id,
- cricket::VideoChannel* channel);
-
- // Randomly generates stream_id.
- // |channel| can be null if one does not exist yet.
- VideoRtpSender(VideoTrackInterface* track, cricket::VideoChannel* channel);
-
- // Randomly generates id and stream_id.
- // |channel| can be null if one does not exist yet.
- explicit VideoRtpSender(cricket::VideoChannel* channel);
-
- virtual ~VideoRtpSender();
-
- // ObserverInterface implementation
- void OnChanged() override;
-
- // RtpSenderInterface implementation
- bool SetTrack(MediaStreamTrackInterface* track) override;
- rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
- return track_;
- }
-
- uint32_t ssrc() const override { return ssrc_; }
-
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_VIDEO;
- }
-
- std::string id() const override { return id_; }
-
- std::vector<std::string> stream_ids() const override {
- std::vector<std::string> ret = {stream_id_};
- return ret;
- }
-
- RtpParameters GetParameters() const override;
- bool SetParameters(const RtpParameters& parameters) override;
-
- // RtpSenderInternal implementation.
- void SetSsrc(uint32_t ssrc) override;
-
- void set_stream_id(const std::string& stream_id) override {
- stream_id_ = stream_id;
- }
- std::string stream_id() const override { return stream_id_; }
-
- void Stop() override;
-
- // Does not take ownership.
- // Should call SetChannel(nullptr) before |channel| is destroyed.
- void SetChannel(cricket::VideoChannel* channel) { channel_ = channel; }
-
- private:
- bool can_send_track() const { return track_ && ssrc_; }
- // Helper function to construct options for
- // VideoProviderInterface::SetVideoSend.
- void SetVideoSend();
- // Helper function to call SetVideoSend with "stop sending" parameters.
- void ClearVideoSend();
-
- std::string id_;
- std::string stream_id_;
- cricket::VideoChannel* channel_ = nullptr;
- rtc::scoped_refptr<VideoTrackInterface> track_;
- uint32_t ssrc_ = 0;
- bool cached_track_enabled_ = false;
- VideoTrackInterface::ContentHint cached_track_content_hint_ =
- VideoTrackInterface::ContentHint::kNone;
- bool stopped_ = false;
-};
-
-} // namespace webrtc
+// Including this file is deprecated. It is no longer part of the public API.
+// This only includes the file in its new location for backwards compatibility.
+#include "webrtc/pc/rtpsender.h"
#endif // WEBRTC_API_RTPSENDER_H_
diff --git a/webrtc/api/rtpsenderinterface.h b/webrtc/api/rtpsenderinterface.h
index 2e6b742..7129376 100644
--- a/webrtc/api/rtpsenderinterface.h
+++ b/webrtc/api/rtpsenderinterface.h
@@ -17,12 +17,12 @@
#include <string>
#include <vector>
+#include "webrtc/api/mediatypes.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/pc/mediasession.h"
namespace webrtc {
diff --git a/webrtc/api/stats/rtcstatscollectorcallback.h b/webrtc/api/stats/rtcstatscollectorcallback.h
new file mode 100644
index 0000000..2493e49
--- /dev/null
+++ b/webrtc/api/stats/rtcstatscollectorcallback.h
@@ -0,0 +1,30 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_STATS_RTCSTATSCOLLECTORCALLBACK_H_
+#define WEBRTC_API_STATS_RTCSTATSCOLLECTORCALLBACK_H_
+
+#include "webrtc/api/stats/rtcstatsreport.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class RTCStatsCollectorCallback : public virtual rtc::RefCountInterface {
+ public:
+ virtual ~RTCStatsCollectorCallback() {}
+
+ virtual void OnStatsDelivered(
+ const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_STATS_RTCSTATSCOLLECTORCALLBACK_H_
diff --git a/webrtc/api/statstypes.cc b/webrtc/api/statstypes.cc
index 80b0d76..300bb4a 100644
--- a/webrtc/api/statstypes.cc
+++ b/webrtc/api/statstypes.cc
@@ -632,6 +632,8 @@
RTC_DCHECK(id_.get());
}
+StatsReport::~StatsReport() = default;
+
// static
StatsReport::Id StatsReport::NewBandwidthEstimationId() {
return Id(new RefCountedObject<BandwidthEstimationId>());
diff --git a/webrtc/api/statstypes.h b/webrtc/api/statstypes.h
index 788c2c4..26e2ac4 100644
--- a/webrtc/api/statstypes.h
+++ b/webrtc/api/statstypes.h
@@ -342,6 +342,7 @@
// Ownership of |id| is passed to |this|.
explicit StatsReport(const Id& id);
+ ~StatsReport();
// Factory functions for various types of stats IDs.
static Id NewBandwidthEstimationId();
diff --git a/webrtc/api/streamcollection.h b/webrtc/api/streamcollection.h
index bedad83..0f847f1 100644
--- a/webrtc/api/streamcollection.h
+++ b/webrtc/api/streamcollection.h
@@ -11,98 +11,8 @@
#ifndef WEBRTC_API_STREAMCOLLECTION_H_
#define WEBRTC_API_STREAMCOLLECTION_H_
-#include <string>
-#include <vector>
-
-#include "webrtc/api/peerconnectioninterface.h"
-
-namespace webrtc {
-
-// Implementation of StreamCollection.
-class StreamCollection : public StreamCollectionInterface {
- public:
- static rtc::scoped_refptr<StreamCollection> Create() {
- rtc::RefCountedObject<StreamCollection>* implementation =
- new rtc::RefCountedObject<StreamCollection>();
- return implementation;
- }
-
- static rtc::scoped_refptr<StreamCollection> Create(
- StreamCollection* streams) {
- rtc::RefCountedObject<StreamCollection>* implementation =
- new rtc::RefCountedObject<StreamCollection>(streams);
- return implementation;
- }
-
- virtual size_t count() {
- return media_streams_.size();
- }
-
- virtual MediaStreamInterface* at(size_t index) {
- return media_streams_.at(index);
- }
-
- virtual MediaStreamInterface* find(const std::string& label) {
- for (StreamVector::iterator it = media_streams_.begin();
- it != media_streams_.end(); ++it) {
- if ((*it)->label().compare(label) == 0) {
- return (*it);
- }
- }
- return NULL;
- }
-
- virtual MediaStreamTrackInterface* FindAudioTrack(
- const std::string& id) {
- for (size_t i = 0; i < media_streams_.size(); ++i) {
- MediaStreamTrackInterface* track = media_streams_[i]->FindAudioTrack(id);
- if (track) {
- return track;
- }
- }
- return NULL;
- }
-
- virtual MediaStreamTrackInterface* FindVideoTrack(
- const std::string& id) {
- for (size_t i = 0; i < media_streams_.size(); ++i) {
- MediaStreamTrackInterface* track = media_streams_[i]->FindVideoTrack(id);
- if (track) {
- return track;
- }
- }
- return NULL;
- }
-
- void AddStream(MediaStreamInterface* stream) {
- for (StreamVector::iterator it = media_streams_.begin();
- it != media_streams_.end(); ++it) {
- if ((*it)->label().compare(stream->label()) == 0)
- return;
- }
- media_streams_.push_back(stream);
- }
-
- void RemoveStream(MediaStreamInterface* remove_stream) {
- for (StreamVector::iterator it = media_streams_.begin();
- it != media_streams_.end(); ++it) {
- if ((*it)->label().compare(remove_stream->label()) == 0) {
- media_streams_.erase(it);
- break;
- }
- }
- }
-
- protected:
- StreamCollection() {}
- explicit StreamCollection(StreamCollection* original)
- : media_streams_(original->media_streams_) {
- }
- typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> >
- StreamVector;
- StreamVector media_streams_;
-};
-
-} // namespace webrtc
+// Including this file is deprecated. It is no longer part of the public API.
+// This only includes the file in its new location for backwards compatibility.
+#include "webrtc/pc/streamcollection.h"
#endif // WEBRTC_API_STREAMCOLLECTION_H_
diff --git a/webrtc/api/videotracksource.h b/webrtc/api/videotracksource.h
index 330af23..45366cd 100644
--- a/webrtc/api/videotracksource.h
+++ b/webrtc/api/videotracksource.h
@@ -11,45 +11,8 @@
#ifndef WEBRTC_API_VIDEOTRACKSOURCE_H_
#define WEBRTC_API_VIDEOTRACKSOURCE_H_
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/notifier.h"
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/media/base/mediachannel.h"
-#include "webrtc/media/base/videosinkinterface.h"
-
-// VideoTrackSource implements VideoTrackSourceInterface.
-namespace webrtc {
-
-class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
- public:
- VideoTrackSource(rtc::VideoSourceInterface<VideoFrame>* source, bool remote);
- void SetState(SourceState new_state);
- // OnSourceDestroyed clears this instance pointer to |source_|. It is useful
- // when the underlying rtc::VideoSourceInterface is destroyed before the
- // reference counted VideoTrackSource.
- void OnSourceDestroyed();
-
- SourceState state() const override { return state_; }
- bool remote() const override { return remote_; }
-
- bool is_screencast() const override { return false; }
- rtc::Optional<bool> needs_denoising() const override {
- return rtc::Optional<bool>(); }
-
- bool GetStats(Stats* stats) override { return false; }
-
- void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
- const rtc::VideoSinkWants& wants) override;
- void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override;
-
- private:
- rtc::ThreadChecker worker_thread_checker_;
- rtc::VideoSourceInterface<VideoFrame>* source_;
- cricket::VideoOptions options_;
- SourceState state_;
- const bool remote_;
-};
-
-} // namespace webrtc
+// Including this file is deprecated. It is no longer part of the public API.
+// This only includes the file in its new location for backwards compatibility.
+#include "webrtc/pc/videotracksource.h"
#endif // WEBRTC_API_VIDEOTRACKSOURCE_H_
diff --git a/webrtc/build/gn_isolate_map.pyl b/webrtc/build/gn_isolate_map.pyl
index ed56777..a6e357e 100644
--- a/webrtc/build/gn_isolate_map.pyl
+++ b/webrtc/build/gn_isolate_map.pyl
@@ -52,7 +52,7 @@
"type": "windowed_test_launcher",
},
"peerconnection_unittests": {
- "label": "//webrtc/api:peerconnection_unittests",
+ "label": "//webrtc/pc:peerconnection_unittests",
"type": "console_test_launcher",
},
"rtc_media_unittests": {
diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn
index eaa3bc9..6f7eaa7 100644
--- a/webrtc/examples/BUILD.gn
+++ b/webrtc/examples/BUILD.gn
@@ -505,7 +505,7 @@
}
deps = [
"//third_party/libyuv",
- "//webrtc/api:libjingle_peerconnection",
+ "//webrtc/pc:libjingle_peerconnection",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
]
diff --git a/webrtc/examples/DEPS b/webrtc/examples/DEPS
index f065c89..524aa26 100644
--- a/webrtc/examples/DEPS
+++ b/webrtc/examples/DEPS
@@ -6,4 +6,5 @@
"+webrtc/modules/audio_device",
"+webrtc/modules/video_capture",
"+webrtc/p2p",
+ "+webrtc/pc",
]
diff --git a/webrtc/p2p/base/p2ptransportchannel_unittest.cc b/webrtc/p2p/base/p2ptransportchannel_unittest.cc
index e5fa8a1..b35bc1b 100644
--- a/webrtc/p2p/base/p2ptransportchannel_unittest.cc
+++ b/webrtc/p2p/base/p2ptransportchannel_unittest.cc
@@ -12,13 +12,6 @@
#include <memory>
#include "webrtc/api/fakemetricsobserver.h"
-#include "webrtc/p2p/base/fakeportallocator.h"
-#include "webrtc/p2p/base/packettransportinterface.h"
-#include "webrtc/p2p/base/p2ptransportchannel.h"
-#include "webrtc/p2p/base/testrelayserver.h"
-#include "webrtc/p2p/base/teststunserver.h"
-#include "webrtc/p2p/base/testturnserver.h"
-#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/fakeclock.h"
@@ -36,6 +29,13 @@
#include "webrtc/base/thread.h"
#include "webrtc/base/virtualsocketserver.h"
#include "webrtc/p2p/base/icetransportinternal.h"
+#include "webrtc/p2p/base/fakeportallocator.h"
+#include "webrtc/p2p/base/p2ptransportchannel.h"
+#include "webrtc/p2p/base/packettransportinterface.h"
+#include "webrtc/p2p/base/testrelayserver.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/client/basicportallocator.h"
namespace {
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index f2bdbc3..5c4d87c 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -7,6 +7,10 @@
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
+if (is_android) {
+ import("//build/config/android/config.gni")
+ import("//build/config/android/rules.gni")
+}
group("pc") {
public_deps = [
@@ -69,6 +73,100 @@
}
}
+config("libjingle_peerconnection_warnings_config") {
+ # GN orders flags on a target before flags from configs. The default config
+ # adds these flags so to cancel them out they need to come from a config and
+ # cannot be on the target directly.
+ if (!is_win && !is_clang) {
+ cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
+ }
+}
+
+rtc_static_library("libjingle_peerconnection") {
+ check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+ cflags = []
+ sources = [
+ "audiotrack.cc",
+ "audiotrack.h",
+ "datachannel.cc",
+ "datachannel.h",
+ "dtmfsender.cc",
+ "dtmfsender.h",
+ "jsepicecandidate.cc",
+ "jsepsessiondescription.cc",
+ "localaudiosource.cc",
+ "localaudiosource.h",
+ "mediacontroller.cc",
+ "mediacontroller.h",
+ "mediastream.cc",
+ "mediastream.h",
+ "mediastreamobserver.cc",
+ "mediastreamobserver.h",
+ "mediastreamtrack.h",
+ "ortcfactory.cc",
+ "ortcfactory.h",
+ "peerconnection.cc",
+ "peerconnection.h",
+ "peerconnectionfactory.cc",
+ "peerconnectionfactory.h",
+ "remoteaudiosource.cc",
+ "remoteaudiosource.h",
+ "rtcstatscollector.cc",
+ "rtcstatscollector.h",
+ "rtpreceiver.cc",
+ "rtpreceiver.h",
+ "rtpsender.cc",
+ "rtpsender.h",
+ "sctputils.cc",
+ "sctputils.h",
+ "statscollector.cc",
+ "statscollector.h",
+ "streamcollection.h",
+ "videocapturertracksource.cc",
+ "videocapturertracksource.h",
+ "videotrack.cc",
+ "videotrack.h",
+ "videotracksource.cc",
+ "videotracksource.h",
+ "webrtcsdp.cc",
+ "webrtcsdp.h",
+ "webrtcsession.cc",
+ "webrtcsession.h",
+ "webrtcsessiondescriptionfactory.cc",
+ "webrtcsessiondescriptionfactory.h",
+ ]
+
+ configs += [ ":libjingle_peerconnection_warnings_config" ]
+
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+
+ deps = [
+ ":rtc_pc",
+ "../api:call_api",
+ "../api:libjingle_peerconnection_api",
+ "../api:rtc_stats_api",
+ "../call",
+ "../media",
+ "../stats",
+ ]
+
+ if (rtc_use_quic) {
+ sources += [
+ "quicdatachannel.cc",
+ "quicdatachannel.h",
+ "quicdatatransport.cc",
+ "quicdatatransport.h",
+ ]
+ deps += [ "//third_party/libquic" ]
+ public_deps = [
+ "//third_party/libquic",
+ ]
+ }
+}
+
if (rtc_include_tests) {
config("rtc_pc_unittests_config") {
# GN orders flags on a target before flags from configs. The default config
@@ -106,8 +204,8 @@
}
deps = [
+ ":libjingle_peerconnection",
":rtc_pc",
- "../api:libjingle_peerconnection",
"../base:rtc_base_tests_utils",
"../media:rtc_unittest_main",
"../system_wrappers:metrics_default",
@@ -121,4 +219,130 @@
deps += [ "//testing/android/native_test:native_test_support" ]
}
}
+
+ config("peerconnection_unittests_config") {
+ # The warnings below are enabled by default. Since GN orders compiler flags
+ # for a target before flags from configs, the only way to disable such
+ # warnings is by having them in a separate config, loaded from the target.
+ # TODO(kjellander): Make the code compile without disabling these flags.
+ # See https://bugs.webrtc.org/3307.
+ if (is_clang && is_win) {
+ cflags = [
+ # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
+ # for -Wno-sign-compare
+ "-Wno-sign-compare",
+ "-Wno-unused-function",
+ ]
+ }
+
+ if (!is_win) {
+ cflags = [ "-Wno-sign-compare" ]
+ }
+ }
+
+ rtc_test("peerconnection_unittests") {
+ check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+ testonly = true
+ sources = [
+ "datachannel_unittest.cc",
+ "dtmfsender_unittest.cc",
+ "fakemediacontroller.h",
+ "jsepsessiondescription_unittest.cc",
+ "localaudiosource_unittest.cc",
+ "mediaconstraintsinterface_unittest.cc",
+ "mediastream_unittest.cc",
+ "ortcfactory_unittest.cc",
+ "peerconnection_unittest.cc",
+ "peerconnectionendtoend_unittest.cc",
+ "peerconnectionfactory_unittest.cc",
+ "peerconnectioninterface_unittest.cc",
+ "proxy_unittest.cc",
+ "rtcstats_integrationtest.cc",
+ "rtcstatscollector_unittest.cc",
+ "rtpsenderreceiver_unittest.cc",
+ "sctputils_unittest.cc",
+ "statscollector_unittest.cc",
+ "test/fakeaudiocapturemodule.cc",
+ "test/fakeaudiocapturemodule.h",
+ "test/fakeaudiocapturemodule_unittest.cc",
+ "test/fakedatachannelprovider.h",
+ "test/fakeperiodicvideocapturer.h",
+ "test/fakertccertificategenerator.h",
+ "test/fakevideotrackrenderer.h",
+ "test/fakevideotracksource.h",
+ "test/mock_datachannel.h",
+ "test/mock_peerconnection.h",
+ "test/mock_webrtcsession.h",
+ "test/mockpeerconnectionobservers.h",
+ "test/peerconnectiontestwrapper.cc",
+ "test/peerconnectiontestwrapper.h",
+ "test/rtcstatsobtainer.h",
+ "test/testsdpstrings.h",
+ "videocapturertracksource_unittest.cc",
+ "videotrack_unittest.cc",
+ "webrtcsdp_unittest.cc",
+ "webrtcsession_unittest.cc",
+ ]
+
+ if (rtc_enable_sctp) {
+ defines = [ "HAVE_SCTP" ]
+ }
+
+ configs += [ ":peerconnection_unittests_config" ]
+
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+
+ # TODO(jschuh): Bug 1348: fix this warning.
+ configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
+
+ if (is_win) {
+ cflags = [
+ "/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
+ "/wd4389", # signed/unsigned mismatch.
+ ]
+ }
+
+ if (rtc_use_quic) {
+ public_deps = [
+ "//third_party/libquic",
+ ]
+ sources += [
+ "quicdatachannel_unittest.cc",
+ "quicdatatransport_unittest.cc",
+ ]
+ }
+
+ deps = []
+ if (is_android) {
+ sources += [
+ "test/androidtestinitializer.cc",
+ "test/androidtestinitializer.h",
+ ]
+ deps += [
+ "//testing/android/native_test:native_test_support",
+ "//webrtc/sdk/android:libjingle_peerconnection_java",
+ "//webrtc/sdk/android:libjingle_peerconnection_jni",
+ ]
+ }
+
+ deps += [
+ ":libjingle_peerconnection",
+ "..:webrtc_common",
+ "../api:fakemetricsobserver",
+ "../base:rtc_base_tests_utils",
+ "../media:rtc_unittest_main",
+ "../pc:rtc_pc",
+ "../system_wrappers:metrics_default",
+ "//testing/gmock",
+ ]
+
+ if (is_android) {
+ deps += [ "//testing/android/native_test:native_test_support" ]
+
+ shard_timeout = 900
+ }
+ }
}
diff --git a/webrtc/pc/DEPS b/webrtc/pc/DEPS
index 2bb6253..5da778c 100644
--- a/webrtc/pc/DEPS
+++ b/webrtc/pc/DEPS
@@ -1,15 +1,29 @@
include_rules = [
+ "+third_party/libsrtp"
"+webrtc/api",
"+webrtc/base",
+ "+webrtc/call",
"+webrtc/common_video/h264",
"+webrtc/logging/rtc_event_log",
+ "+webrtc/logging/rtc_event_log",
"+webrtc/media",
+ "+webrtc/modules/audio_device",
+ "+webrtc/modules/rtp_rtcp",
+ "+webrtc/modules/video_coding",
+ "+webrtc/modules/video_render",
"+webrtc/p2p",
- "+third_party/libsrtp"
+ "+webrtc/system_wrappers",
]
specific_include_rules = {
+ "androidtestinitializer\.cc": [
+ "+base/android", # Allowed only for Android tests.
+ "+webrtc/voice_engine",
+ ],
"srtpfilter_unittest\.cc": [
"+crypto",
],
+ "peerconnectionfactory\.cc": [
+ "+webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h",
+ ],
}
diff --git a/webrtc/api/audiotrack.cc b/webrtc/pc/audiotrack.cc
similarity index 95%
rename from webrtc/api/audiotrack.cc
rename to webrtc/pc/audiotrack.cc
index c2c9557..951acbf 100644
--- a/webrtc/api/audiotrack.cc
+++ b/webrtc/pc/audiotrack.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/audiotrack.h"
+#include "webrtc/pc/audiotrack.h"
#include "webrtc/base/checks.h"
@@ -16,8 +16,6 @@
namespace webrtc {
-const char MediaStreamTrackInterface::kAudioKind[] = "audio";
-
// static
scoped_refptr<AudioTrack> AudioTrack::Create(
const std::string& id,
diff --git a/webrtc/api/audiotrack.h b/webrtc/pc/audiotrack.h
similarity index 91%
rename from webrtc/api/audiotrack.h
rename to webrtc/pc/audiotrack.h
index 096caf9..33f4b1f 100644
--- a/webrtc/api/audiotrack.h
+++ b/webrtc/pc/audiotrack.h
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_AUDIOTRACK_H_
-#define WEBRTC_API_AUDIOTRACK_H_
+#ifndef WEBRTC_PC_AUDIOTRACK_H_
+#define WEBRTC_PC_AUDIOTRACK_H_
#include <string>
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/mediastreamtrack.h"
#include "webrtc/api/notifier.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread_checker.h"
+#include "webrtc/pc/mediastreamtrack.h"
namespace webrtc {
@@ -56,4 +56,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_AUDIOTRACK_H_
+#endif // WEBRTC_PC_AUDIOTRACK_H_
diff --git a/webrtc/pc/channelmanager.cc b/webrtc/pc/channelmanager.cc
index 0570267..f667900 100644
--- a/webrtc/pc/channelmanager.cc
+++ b/webrtc/pc/channelmanager.cc
@@ -12,7 +12,6 @@
#include <algorithm>
-#include "webrtc/api/mediacontroller.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
@@ -23,6 +22,7 @@
#include "webrtc/media/base/device.h"
#include "webrtc/media/base/rtpdataengine.h"
#include "webrtc/pc/srtpfilter.h"
+#include "webrtc/pc/mediacontroller.h"
namespace cricket {
diff --git a/webrtc/pc/channelmanager_unittest.cc b/webrtc/pc/channelmanager_unittest.cc
index cf54b91..74c3590 100644
--- a/webrtc/pc/channelmanager_unittest.cc
+++ b/webrtc/pc/channelmanager_unittest.cc
@@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/fakemediacontroller.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
@@ -19,6 +18,7 @@
#include "webrtc/media/engine/fakewebrtccall.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/channelmanager.h"
+#include "webrtc/pc/fakemediacontroller.h"
namespace cricket {
const bool kDefaultRtcpMuxRequired = true;
diff --git a/webrtc/api/datachannel.cc b/webrtc/pc/datachannel.cc
similarity index 99%
rename from webrtc/api/datachannel.cc
rename to webrtc/pc/datachannel.cc
index c0bc3dc..5606a64 100644
--- a/webrtc/api/datachannel.cc
+++ b/webrtc/pc/datachannel.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/datachannel.h"
+#include "webrtc/pc/datachannel.h"
#include <memory>
#include <string>
-#include "webrtc/api/sctputils.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/refcount.h"
#include "webrtc/media/sctp/sctptransportinternal.h"
+#include "webrtc/pc/sctputils.h"
namespace webrtc {
diff --git a/webrtc/pc/datachannel.h b/webrtc/pc/datachannel.h
new file mode 100644
index 0000000..2619a85
--- /dev/null
+++ b/webrtc/pc/datachannel.h
@@ -0,0 +1,298 @@
+/*
+ * Copyright 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_PC_DATACHANNEL_H_
+#define WEBRTC_PC_DATACHANNEL_H_
+
+#include <deque>
+#include <set>
+#include <string>
+
+#include "webrtc/api/datachannelinterface.h"
+#include "webrtc/api/proxy.h"
+#include "webrtc/base/messagehandler.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/pc/channel.h"
+
+namespace webrtc {
+
+class DataChannel;
+
+class DataChannelProviderInterface {
+ public:
+ // Sends the data to the transport.
+ virtual bool SendData(const cricket::SendDataParams& params,
+ const rtc::CopyOnWriteBuffer& payload,
+ cricket::SendDataResult* result) = 0;
+ // Connects to the transport signals.
+ virtual bool ConnectDataChannel(DataChannel* data_channel) = 0;
+ // Disconnects from the transport signals.
+ virtual void DisconnectDataChannel(DataChannel* data_channel) = 0;
+ // Adds the data channel SID to the transport for SCTP.
+ virtual void AddSctpDataStream(int sid) = 0;
+ // Removes the data channel SID from the transport for SCTP.
+ virtual void RemoveSctpDataStream(int sid) = 0;
+ // Returns true if the transport channel is ready to send data.
+ virtual bool ReadyToSendData() const = 0;
+
+ protected:
+ virtual ~DataChannelProviderInterface() {}
+};
+
+struct InternalDataChannelInit : public DataChannelInit {
+ enum OpenHandshakeRole {
+ kOpener,
+ kAcker,
+ kNone
+ };
+ // The default role is kOpener because the default |negotiated| is false.
+ InternalDataChannelInit() : open_handshake_role(kOpener) {}
+ explicit InternalDataChannelInit(const DataChannelInit& base)
+ : DataChannelInit(base), open_handshake_role(kOpener) {
+ // If the channel is externally negotiated, do not send the OPEN message.
+ if (base.negotiated) {
+ open_handshake_role = kNone;
+ }
+ }
+
+ OpenHandshakeRole open_handshake_role;
+};
+
+// Helper class to allocate unique IDs for SCTP DataChannels
+class SctpSidAllocator {
+ public:
+ // Gets the first unused odd/even id based on the DTLS role. If |role| is
+ // SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
+ // otherwise, the id starts from 1 and takes odd numbers.
+ // Returns false if no id can be allocated.
+ bool AllocateSid(rtc::SSLRole role, int* sid);
+
+ // Attempts to reserve a specific sid. Returns false if it's unavailable.
+ bool ReserveSid(int sid);
+
+ // Indicates that |sid| isn't in use any more, and is thus available again.
+ void ReleaseSid(int sid);
+
+ private:
+ // Checks if |sid| is available to be assigned to a new SCTP data channel.
+ bool IsSidAvailable(int sid) const;
+
+ std::set<int> used_sids_;
+};
+
+// DataChannel is a an implementation of the DataChannelInterface based on
+// libjingle's data engine. It provides an implementation of unreliable or
+// reliabledata channels. Currently this class is specifically designed to use
+// both RtpDataEngine and SctpDataEngine.
+
+// DataChannel states:
+// kConnecting: The channel has been created the transport might not yet be
+// ready.
+// kOpen: The channel have a local SSRC set by a call to UpdateSendSsrc
+// and a remote SSRC set by call to UpdateReceiveSsrc and the transport
+// has been writable once.
+// kClosing: DataChannelInterface::Close has been called or UpdateReceiveSsrc
+// has been called with SSRC==0
+// kClosed: Both UpdateReceiveSsrc and UpdateSendSsrc has been called with
+// SSRC==0.
+class DataChannel : public DataChannelInterface,
+ public sigslot::has_slots<>,
+ public rtc::MessageHandler {
+ public:
+ static rtc::scoped_refptr<DataChannel> Create(
+ DataChannelProviderInterface* provider,
+ cricket::DataChannelType dct,
+ const std::string& label,
+ const InternalDataChannelInit& config);
+
+ virtual void RegisterObserver(DataChannelObserver* observer);
+ virtual void UnregisterObserver();
+
+ virtual std::string label() const { return label_; }
+ virtual bool reliable() const;
+ virtual bool ordered() const { return config_.ordered; }
+ virtual uint16_t maxRetransmitTime() const {
+ return config_.maxRetransmitTime;
+ }
+ virtual uint16_t maxRetransmits() const { return config_.maxRetransmits; }
+ virtual std::string protocol() const { return config_.protocol; }
+ virtual bool negotiated() const { return config_.negotiated; }
+ virtual int id() const { return config_.id; }
+ virtual uint64_t buffered_amount() const;
+ virtual void Close();
+ virtual DataState state() const { return state_; }
+ virtual uint32_t messages_sent() const { return messages_sent_; }
+ virtual uint64_t bytes_sent() const { return bytes_sent_; }
+ virtual uint32_t messages_received() const { return messages_received_; }
+ virtual uint64_t bytes_received() const { return bytes_received_; }
+ virtual bool Send(const DataBuffer& buffer);
+
+ // rtc::MessageHandler override.
+ virtual void OnMessage(rtc::Message* msg);
+
+ // Called when the channel's ready to use. That can happen when the
+ // underlying DataMediaChannel becomes ready, or when this channel is a new
+ // stream on an existing DataMediaChannel, and we've finished negotiation.
+ void OnChannelReady(bool writable);
+
+ // Slots for provider to connect signals to.
+ void OnDataReceived(const cricket::ReceiveDataParams& params,
+ const rtc::CopyOnWriteBuffer& payload);
+ void OnStreamClosedRemotely(int sid);
+
+ // The remote peer request that this channel should be closed.
+ void RemotePeerRequestClose();
+
+ // The following methods are for SCTP only.
+
+ // Sets the SCTP sid and adds to transport layer if not set yet. Should only
+ // be called once.
+ void SetSctpSid(int sid);
+ // Called when the transport channel is created.
+ // Only needs to be called for SCTP data channels.
+ void OnTransportChannelCreated();
+ // Called when the transport channel is destroyed.
+ // This method makes sure the DataChannel is disconnected and changes state
+ // to kClosed.
+ void OnTransportChannelDestroyed();
+
+ // The following methods are for RTP only.
+
+ // Set the SSRC this channel should use to send data on the
+ // underlying data engine. |send_ssrc| == 0 means that the channel is no
+ // longer part of the session negotiation.
+ void SetSendSsrc(uint32_t send_ssrc);
+ // Set the SSRC this channel should use to receive data from the
+ // underlying data engine.
+ void SetReceiveSsrc(uint32_t receive_ssrc);
+
+ cricket::DataChannelType data_channel_type() const {
+ return data_channel_type_;
+ }
+
+ // Emitted when state transitions to kOpen.
+ sigslot::signal1<DataChannel*> SignalOpened;
+ // Emitted when state transitions to kClosed.
+ // In the case of SCTP channels, this signal can be used to tell when the
+ // channel's sid is free.
+ sigslot::signal1<DataChannel*> SignalClosed;
+
+ protected:
+ DataChannel(DataChannelProviderInterface* client,
+ cricket::DataChannelType dct,
+ const std::string& label);
+ virtual ~DataChannel();
+
+ private:
+ // A packet queue which tracks the total queued bytes. Queued packets are
+ // owned by this class.
+ class PacketQueue {
+ public:
+ PacketQueue();
+ ~PacketQueue();
+
+ size_t byte_count() const {
+ return byte_count_;
+ }
+
+ bool Empty() const;
+
+ DataBuffer* Front();
+
+ void Pop();
+
+ void Push(DataBuffer* packet);
+
+ void Clear();
+
+ void Swap(PacketQueue* other);
+
+ private:
+ std::deque<DataBuffer*> packets_;
+ size_t byte_count_;
+ };
+
+ // The OPEN(_ACK) signaling state.
+ enum HandshakeState {
+ kHandshakeInit,
+ kHandshakeShouldSendOpen,
+ kHandshakeShouldSendAck,
+ kHandshakeWaitingForAck,
+ kHandshakeReady
+ };
+
+ bool Init(const InternalDataChannelInit& config);
+ void DoClose();
+ void UpdateState();
+ void SetState(DataState state);
+ void DisconnectFromProvider();
+
+ void DeliverQueuedReceivedData();
+
+ void SendQueuedDataMessages();
+ bool SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked);
+ bool QueueSendDataMessage(const DataBuffer& buffer);
+
+ void SendQueuedControlMessages();
+ void QueueControlMessage(const rtc::CopyOnWriteBuffer& buffer);
+ bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer);
+
+ std::string label_;
+ InternalDataChannelInit config_;
+ DataChannelObserver* observer_;
+ DataState state_;
+ uint32_t messages_sent_;
+ uint64_t bytes_sent_;
+ uint32_t messages_received_;
+ uint64_t bytes_received_;
+ cricket::DataChannelType data_channel_type_;
+ DataChannelProviderInterface* provider_;
+ HandshakeState handshake_state_;
+ bool connected_to_provider_;
+ bool send_ssrc_set_;
+ bool receive_ssrc_set_;
+ bool writable_;
+ uint32_t send_ssrc_;
+ uint32_t receive_ssrc_;
+ // Control messages that always have to get sent out before any queued
+ // data.
+ PacketQueue queued_control_data_;
+ PacketQueue queued_received_data_;
+ PacketQueue queued_send_data_;
+};
+
+// Define proxy for DataChannelInterface.
+BEGIN_SIGNALING_PROXY_MAP(DataChannel)
+ PROXY_SIGNALING_THREAD_DESTRUCTOR()
+ PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
+ PROXY_METHOD0(void, UnregisterObserver)
+ PROXY_CONSTMETHOD0(std::string, label)
+ PROXY_CONSTMETHOD0(bool, reliable)
+ PROXY_CONSTMETHOD0(bool, ordered)
+ PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime)
+ PROXY_CONSTMETHOD0(uint16_t, maxRetransmits)
+ PROXY_CONSTMETHOD0(std::string, protocol)
+ PROXY_CONSTMETHOD0(bool, negotiated)
+ PROXY_CONSTMETHOD0(int, id)
+ PROXY_CONSTMETHOD0(DataState, state)
+ PROXY_CONSTMETHOD0(uint32_t, messages_sent)
+ PROXY_CONSTMETHOD0(uint64_t, bytes_sent)
+ PROXY_CONSTMETHOD0(uint32_t, messages_received)
+ PROXY_CONSTMETHOD0(uint64_t, bytes_received)
+ PROXY_CONSTMETHOD0(uint64_t, buffered_amount)
+ PROXY_METHOD0(void, Close)
+ PROXY_METHOD1(bool, Send, const DataBuffer&)
+END_PROXY_MAP()
+
+} // namespace webrtc
+
+#endif // WEBRTC_PC_DATACHANNEL_H_
diff --git a/webrtc/api/datachannel_unittest.cc b/webrtc/pc/datachannel_unittest.cc
similarity index 99%
rename from webrtc/api/datachannel_unittest.cc
rename to webrtc/pc/datachannel_unittest.cc
index 4ce1be5..13ab63c 100644
--- a/webrtc/api/datachannel_unittest.cc
+++ b/webrtc/pc/datachannel_unittest.cc
@@ -10,10 +10,10 @@
#include <memory>
-#include "webrtc/api/datachannel.h"
-#include "webrtc/api/sctputils.h"
-#include "webrtc/api/test/fakedatachannelprovider.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/pc/datachannel.h"
+#include "webrtc/pc/sctputils.h"
+#include "webrtc/pc/test/fakedatachannelprovider.h"
using webrtc::DataChannel;
using webrtc::SctpSidAllocator;
diff --git a/webrtc/api/dtmfsender.cc b/webrtc/pc/dtmfsender.cc
similarity index 99%
rename from webrtc/api/dtmfsender.cc
rename to webrtc/pc/dtmfsender.cc
index c715d3d..2ef921b 100644
--- a/webrtc/api/dtmfsender.cc
+++ b/webrtc/pc/dtmfsender.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/dtmfsender.h"
+#include "webrtc/pc/dtmfsender.h"
#include <ctype.h>
diff --git a/webrtc/api/dtmfsender.h b/webrtc/pc/dtmfsender.h
similarity index 97%
rename from webrtc/api/dtmfsender.h
rename to webrtc/pc/dtmfsender.h
index d4e8f06..1163adf 100644
--- a/webrtc/api/dtmfsender.h
+++ b/webrtc/pc/dtmfsender.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_DTMFSENDER_H_
-#define WEBRTC_API_DTMFSENDER_H_
+#ifndef WEBRTC_PC_DTMFSENDER_H_
+#define WEBRTC_PC_DTMFSENDER_H_
#include <string>
@@ -121,4 +121,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_DTMFSENDER_H_
+#endif // WEBRTC_PC_DTMFSENDER_H_
diff --git a/webrtc/api/dtmfsender_unittest.cc b/webrtc/pc/dtmfsender_unittest.cc
similarity index 99%
rename from webrtc/api/dtmfsender_unittest.cc
rename to webrtc/pc/dtmfsender_unittest.cc
index 54f3a7b..a58c1ec 100644
--- a/webrtc/api/dtmfsender_unittest.cc
+++ b/webrtc/pc/dtmfsender_unittest.cc
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/dtmfsender.h"
+#include "webrtc/pc/dtmfsender.h"
#include <memory>
#include <set>
#include <string>
#include <vector>
-#include "webrtc/api/audiotrack.h"
#include "webrtc/base/fakeclock.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/timeutils.h"
+#include "webrtc/pc/audiotrack.h"
using webrtc::AudioTrackInterface;
using webrtc::AudioTrack;
diff --git a/webrtc/api/fakemediacontroller.h b/webrtc/pc/fakemediacontroller.h
similarity index 88%
rename from webrtc/api/fakemediacontroller.h
rename to webrtc/pc/fakemediacontroller.h
index bc3f3e2..6d639ed 100644
--- a/webrtc/api/fakemediacontroller.h
+++ b/webrtc/pc/fakemediacontroller.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_FAKEMEDIACONTROLLER_H_
-#define WEBRTC_API_FAKEMEDIACONTROLLER_H_
+#ifndef WEBRTC_PC_FAKEMEDIACONTROLLER_H_
+#define WEBRTC_PC_FAKEMEDIACONTROLLER_H_
-#include "webrtc/api/mediacontroller.h"
#include "webrtc/base/checks.h"
#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/pc/mediacontroller.h"
namespace cricket {
@@ -39,4 +39,4 @@
webrtc::Call* call_;
};
} // namespace cricket
-#endif // WEBRTC_API_FAKEMEDIACONTROLLER_H_
+#endif // WEBRTC_PC_FAKEMEDIACONTROLLER_H_
diff --git a/webrtc/api/jsepicecandidate.cc b/webrtc/pc/jsepicecandidate.cc
similarity index 98%
rename from webrtc/api/jsepicecandidate.cc
rename to webrtc/pc/jsepicecandidate.cc
index cced1b4..51faa07 100644
--- a/webrtc/api/jsepicecandidate.cc
+++ b/webrtc/pc/jsepicecandidate.cc
@@ -12,8 +12,8 @@
#include <vector>
-#include "webrtc/api/webrtcsdp.h"
#include "webrtc/base/stringencode.h"
+#include "webrtc/pc/webrtcsdp.h"
namespace webrtc {
diff --git a/webrtc/api/jsepsessiondescription.cc b/webrtc/pc/jsepsessiondescription.cc
similarity index 99%
rename from webrtc/api/jsepsessiondescription.cc
rename to webrtc/pc/jsepsessiondescription.cc
index 6f73a1e..02919b6 100644
--- a/webrtc/api/jsepsessiondescription.cc
+++ b/webrtc/pc/jsepsessiondescription.cc
@@ -12,10 +12,10 @@
#include <memory>
-#include "webrtc/api/webrtcsdp.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/pc/mediasession.h"
+#include "webrtc/pc/webrtcsdp.h"
using cricket::SessionDescription;
diff --git a/webrtc/api/jsepsessiondescription_unittest.cc b/webrtc/pc/jsepsessiondescription_unittest.cc
similarity index 100%
rename from webrtc/api/jsepsessiondescription_unittest.cc
rename to webrtc/pc/jsepsessiondescription_unittest.cc
diff --git a/webrtc/api/localaudiosource.cc b/webrtc/pc/localaudiosource.cc
similarity index 97%
rename from webrtc/api/localaudiosource.cc
rename to webrtc/pc/localaudiosource.cc
index 57cfdf8..3d79da9 100644
--- a/webrtc/api/localaudiosource.cc
+++ b/webrtc/pc/localaudiosource.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/localaudiosource.h"
+#include "webrtc/pc/localaudiosource.h"
#include <vector>
diff --git a/webrtc/api/localaudiosource.h b/webrtc/pc/localaudiosource.h
similarity index 93%
rename from webrtc/api/localaudiosource.h
rename to webrtc/pc/localaudiosource.h
index e1c023e..58e90af 100644
--- a/webrtc/api/localaudiosource.h
+++ b/webrtc/pc/localaudiosource.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_LOCALAUDIOSOURCE_H_
-#define WEBRTC_API_LOCALAUDIOSOURCE_H_
+#ifndef WEBRTC_PC_LOCALAUDIOSOURCE_H_
+#define WEBRTC_PC_LOCALAUDIOSOURCE_H_
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/notifier.h"
@@ -57,4 +57,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_LOCALAUDIOSOURCE_H_
+#endif // WEBRTC_PC_LOCALAUDIOSOURCE_H_
diff --git a/webrtc/api/localaudiosource_unittest.cc b/webrtc/pc/localaudiosource_unittest.cc
similarity index 98%
rename from webrtc/api/localaudiosource_unittest.cc
rename to webrtc/pc/localaudiosource_unittest.cc
index 12f46b7..394a2ea 100644
--- a/webrtc/api/localaudiosource_unittest.cc
+++ b/webrtc/pc/localaudiosource_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/localaudiosource.h"
+#include "webrtc/pc/localaudiosource.h"
#include <string>
#include <vector>
diff --git a/webrtc/api/mediaconstraintsinterface_unittest.cc b/webrtc/pc/mediaconstraintsinterface_unittest.cc
similarity index 100%
rename from webrtc/api/mediaconstraintsinterface_unittest.cc
rename to webrtc/pc/mediaconstraintsinterface_unittest.cc
diff --git a/webrtc/api/mediacontroller.cc b/webrtc/pc/mediacontroller.cc
similarity index 98%
rename from webrtc/api/mediacontroller.cc
rename to webrtc/pc/mediacontroller.cc
index c64ff01..fa8acad 100644
--- a/webrtc/api/mediacontroller.cc
+++ b/webrtc/pc/mediacontroller.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/mediacontroller.h"
+#include "webrtc/pc/mediacontroller.h"
#include <memory>
diff --git a/webrtc/pc/mediacontroller.h b/webrtc/pc/mediacontroller.h
new file mode 100644
index 0000000..85617af
--- /dev/null
+++ b/webrtc/pc/mediacontroller.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_PC_MEDIACONTROLLER_H_
+#define WEBRTC_PC_MEDIACONTROLLER_H_
+
+#include "webrtc/base/thread.h"
+
+namespace cricket {
+class ChannelManager;
+struct MediaConfig;
+} // namespace cricket
+
+namespace webrtc {
+class Call;
+class VoiceEngine;
+class RtcEventLog;
+
+// The MediaController currently owns shared state between media channels, but
+// in the future will create and own RtpSenders and RtpReceivers.
+class MediaControllerInterface {
+ public:
+ static MediaControllerInterface* Create(
+ const cricket::MediaConfig& config,
+ rtc::Thread* worker_thread,
+ cricket::ChannelManager* channel_manager,
+ webrtc::RtcEventLog* event_log);
+
+ virtual ~MediaControllerInterface() {}
+ virtual void Close() = 0;
+ virtual webrtc::Call* call_w() = 0;
+ virtual cricket::ChannelManager* channel_manager() const = 0;
+ virtual const cricket::MediaConfig& config() const = 0;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_PC_MEDIACONTROLLER_H_
diff --git a/webrtc/pc/mediasession.cc b/webrtc/pc/mediasession.cc
index 5e69423..910c72b 100644
--- a/webrtc/pc/mediasession.cc
+++ b/webrtc/pc/mediasession.cc
@@ -1193,25 +1193,6 @@
return current_tdesc->secure();
}
-std::string MediaTypeToString(MediaType type) {
- std::string type_str;
- switch (type) {
- case MEDIA_TYPE_AUDIO:
- type_str = "audio";
- break;
- case MEDIA_TYPE_VIDEO:
- type_str = "video";
- break;
- case MEDIA_TYPE_DATA:
- type_str = "data";
- break;
- default:
- RTC_NOTREACHED();
- break;
- }
- return type_str;
-}
-
std::string MediaContentDirectionToString(MediaContentDirection direction) {
std::string dir_str;
switch (direction) {
diff --git a/webrtc/pc/mediasession.h b/webrtc/pc/mediasession.h
index ee2126d..06815ec 100644
--- a/webrtc/pc/mediasession.h
+++ b/webrtc/pc/mediasession.h
@@ -18,6 +18,7 @@
#include <string>
#include <vector>
+#include "webrtc/api/mediatypes.h"
#include "webrtc/media/base/codec.h"
#include "webrtc/media/base/cryptoparams.h"
#include "webrtc/media/base/mediachannel.h"
@@ -37,14 +38,6 @@
typedef std::vector<CryptoParams> CryptoParamsVec;
typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions;
-enum MediaType {
- MEDIA_TYPE_AUDIO,
- MEDIA_TYPE_VIDEO,
- MEDIA_TYPE_DATA
-};
-
-std::string MediaTypeToString(MediaType type);
-
enum MediaContentDirection {
MD_INACTIVE,
MD_SENDONLY,
diff --git a/webrtc/api/mediastream.cc b/webrtc/pc/mediastream.cc
similarity index 98%
rename from webrtc/api/mediastream.cc
rename to webrtc/pc/mediastream.cc
index bef04ae..1830792 100644
--- a/webrtc/api/mediastream.cc
+++ b/webrtc/pc/mediastream.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/mediastream.h"
+#include "webrtc/pc/mediastream.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
diff --git a/webrtc/pc/mediastream.h b/webrtc/pc/mediastream.h
new file mode 100644
index 0000000..56fbed9
--- /dev/null
+++ b/webrtc/pc/mediastream.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains the implementation of MediaStreamInterface interface.
+
+#ifndef WEBRTC_PC_MEDIASTREAM_H_
+#define WEBRTC_PC_MEDIASTREAM_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/notifier.h"
+
+namespace webrtc {
+
+class MediaStream : public Notifier<MediaStreamInterface> {
+ public:
+ static rtc::scoped_refptr<MediaStream> Create(const std::string& label);
+
+ std::string label() const override { return label_; }
+
+ bool AddTrack(AudioTrackInterface* track) override;
+ bool AddTrack(VideoTrackInterface* track) override;
+ bool RemoveTrack(AudioTrackInterface* track) override;
+ bool RemoveTrack(VideoTrackInterface* track) override;
+ rtc::scoped_refptr<AudioTrackInterface>
+ FindAudioTrack(const std::string& track_id) override;
+ rtc::scoped_refptr<VideoTrackInterface>
+ FindVideoTrack(const std::string& track_id) override;
+
+ AudioTrackVector GetAudioTracks() override { return audio_tracks_; }
+ VideoTrackVector GetVideoTracks() override { return video_tracks_; }
+
+ protected:
+ explicit MediaStream(const std::string& label);
+
+ private:
+ template <typename TrackVector, typename Track>
+ bool AddTrack(TrackVector* Tracks, Track* track);
+ template <typename TrackVector>
+ bool RemoveTrack(TrackVector* Tracks, MediaStreamTrackInterface* track);
+
+ std::string label_;
+ AudioTrackVector audio_tracks_;
+ VideoTrackVector video_tracks_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_PC_MEDIASTREAM_H_
diff --git a/webrtc/api/mediastream_unittest.cc b/webrtc/pc/mediastream_unittest.cc
similarity index 96%
rename from webrtc/api/mediastream_unittest.cc
rename to webrtc/pc/mediastream_unittest.cc
index 5df08ce..e2e3985 100644
--- a/webrtc/api/mediastream_unittest.cc
+++ b/webrtc/pc/mediastream_unittest.cc
@@ -10,14 +10,14 @@
#include <string>
-#include "webrtc/api/audiotrack.h"
-#include "webrtc/api/mediastream.h"
-#include "webrtc/api/test/fakevideotracksource.h"
-#include "webrtc/api/videotrack.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/refcount.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
+#include "webrtc/pc/audiotrack.h"
+#include "webrtc/pc/mediastream.h"
+#include "webrtc/pc/test/fakevideotracksource.h"
+#include "webrtc/pc/videotrack.h"
static const char kStreamLabel1[] = "local_stream_1";
static const char kVideoTrackId[] = "dummy_video_cam_1";
diff --git a/webrtc/api/mediastreamobserver.cc b/webrtc/pc/mediastreamobserver.cc
similarity index 98%
rename from webrtc/api/mediastreamobserver.cc
rename to webrtc/pc/mediastreamobserver.cc
index 0cf52b4..bf17582 100644
--- a/webrtc/api/mediastreamobserver.cc
+++ b/webrtc/pc/mediastreamobserver.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/mediastreamobserver.h"
+#include "webrtc/pc/mediastreamobserver.h"
#include <algorithm>
diff --git a/webrtc/api/mediastreamobserver.h b/webrtc/pc/mediastreamobserver.h
similarity index 91%
rename from webrtc/api/mediastreamobserver.h
rename to webrtc/pc/mediastreamobserver.h
index 3179976..5d9012b 100644
--- a/webrtc/api/mediastreamobserver.h
+++ b/webrtc/pc/mediastreamobserver.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_MEDIASTREAMOBSERVER_H_
-#define WEBRTC_API_MEDIASTREAMOBSERVER_H_
+#ifndef WEBRTC_PC_MEDIASTREAMOBSERVER_H_
+#define WEBRTC_PC_MEDIASTREAMOBSERVER_H_
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/base/scoped_ref_ptr.h"
@@ -45,4 +45,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_MEDIASTREAMOBSERVER_H_
+#endif // WEBRTC_PC_MEDIASTREAMOBSERVER_H_
diff --git a/webrtc/pc/mediastreamtrack.h b/webrtc/pc/mediastreamtrack.h
new file mode 100644
index 0000000..0fb2de2
--- /dev/null
+++ b/webrtc/pc/mediastreamtrack.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_PC_MEDIASTREAMTRACK_H_
+#define WEBRTC_PC_MEDIASTREAMTRACK_H_
+
+#include <string>
+
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/notifier.h"
+
+namespace webrtc {
+
+// MediaTrack implements the interface common to AudioTrackInterface and
+// VideoTrackInterface.
+template <typename T>
+class MediaStreamTrack : public Notifier<T> {
+ public:
+ typedef typename T::TrackState TypedTrackState;
+
+ std::string id() const override { return id_; }
+ MediaStreamTrackInterface::TrackState state() const override {
+ return state_;
+ }
+ bool enabled() const override { return enabled_; }
+ bool set_enabled(bool enable) override {
+ bool fire_on_change = (enable != enabled_);
+ enabled_ = enable;
+ if (fire_on_change) {
+ Notifier<T>::FireOnChanged();
+ }
+ return fire_on_change;
+ }
+
+ protected:
+ explicit MediaStreamTrack(const std::string& id)
+ : enabled_(true), id_(id), state_(MediaStreamTrackInterface::kLive) {}
+
+ bool set_state(MediaStreamTrackInterface::TrackState new_state) {
+ bool fire_on_change = (state_ != new_state);
+ state_ = new_state;
+ if (fire_on_change)
+ Notifier<T>::FireOnChanged();
+ return true;
+ }
+
+ private:
+ bool enabled_;
+ std::string id_;
+ MediaStreamTrackInterface::TrackState state_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_PC_MEDIASTREAMTRACK_H_
diff --git a/webrtc/api/ortcfactory.cc b/webrtc/pc/ortcfactory.cc
similarity index 98%
rename from webrtc/api/ortcfactory.cc
rename to webrtc/pc/ortcfactory.cc
index ba41a03..47d39b7 100644
--- a/webrtc/api/ortcfactory.cc
+++ b/webrtc/pc/ortcfactory.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/ortcfactory.h"
+#include "webrtc/pc/ortcfactory.h"
#include <string>
#include <utility> // For std::move.
diff --git a/webrtc/api/ortcfactory.h b/webrtc/pc/ortcfactory.h
similarity index 95%
rename from webrtc/api/ortcfactory.h
rename to webrtc/pc/ortcfactory.h
index dc92774..65fe10f 100644
--- a/webrtc/api/ortcfactory.h
+++ b/webrtc/pc/ortcfactory.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_ORTCFACTORY_H_
-#define WEBRTC_API_ORTCFACTORY_H_
+#ifndef WEBRTC_PC_ORTCFACTORY_H_
+#define WEBRTC_PC_ORTCFACTORY_H_
#include <memory>
@@ -61,4 +61,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_ORTCFACTORY_H_
+#endif // WEBRTC_PC_ORTCFACTORY_H_
diff --git a/webrtc/api/ortcfactory_unittest.cc b/webrtc/pc/ortcfactory_unittest.cc
similarity index 100%
rename from webrtc/api/ortcfactory_unittest.cc
rename to webrtc/pc/ortcfactory_unittest.cc
diff --git a/webrtc/api/peerconnection.cc b/webrtc/pc/peerconnection.cc
similarity index 99%
rename from webrtc/api/peerconnection.cc
rename to webrtc/pc/peerconnection.cc
index 78e6790..cbd0d4b 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/pc/peerconnection.cc
@@ -8,28 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/peerconnection.h"
+#include "webrtc/pc/peerconnection.h"
#include <algorithm>
#include <cctype> // for isdigit
#include <utility>
#include <vector>
-#include "webrtc/api/audiotrack.h"
-#include "webrtc/api/dtmfsender.h"
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/mediaconstraintsinterface.h"
-#include "webrtc/api/mediastream.h"
-#include "webrtc/api/mediastreamobserver.h"
#include "webrtc/api/mediastreamproxy.h"
#include "webrtc/api/mediastreamtrackproxy.h"
-#include "webrtc/api/remoteaudiosource.h"
-#include "webrtc/api/rtpreceiver.h"
-#include "webrtc/api/rtpsender.h"
-#include "webrtc/api/streamcollection.h"
-#include "webrtc/api/videocapturertracksource.h"
-#include "webrtc/api/videotrack.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
@@ -40,7 +30,18 @@
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/sctp/sctptransport.h"
+#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/channelmanager.h"
+#include "webrtc/pc/dtmfsender.h"
+#include "webrtc/pc/mediastream.h"
+#include "webrtc/pc/mediastreamobserver.h"
+#include "webrtc/pc/remoteaudiosource.h"
+#include "webrtc/pc/rtpreceiver.h"
+#include "webrtc/pc/rtpsender.h"
+#include "webrtc/pc/streamcollection.h"
+#include "webrtc/pc/videocapturertracksource.h"
+#include "webrtc/pc/videotrack.h"
+#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace {
diff --git a/webrtc/api/peerconnection.h b/webrtc/pc/peerconnection.h
similarity index 97%
rename from webrtc/api/peerconnection.h
rename to webrtc/pc/peerconnection.h
index 82dc8ed..ac8f33c 100644
--- a/webrtc/api/peerconnection.h
+++ b/webrtc/pc/peerconnection.h
@@ -8,22 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_PEERCONNECTION_H_
-#define WEBRTC_API_PEERCONNECTION_H_
+#ifndef WEBRTC_PC_PEERCONNECTION_H_
+#define WEBRTC_PC_PEERCONNECTION_H_
#include <string>
#include <map>
#include <memory>
#include <vector>
-#include "webrtc/api/peerconnectionfactory.h"
#include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/api/rtcstatscollector.h"
-#include "webrtc/api/rtpreceiver.h"
-#include "webrtc/api/rtpsender.h"
-#include "webrtc/api/statscollector.h"
-#include "webrtc/api/streamcollection.h"
-#include "webrtc/api/webrtcsession.h"
+#include "webrtc/pc/peerconnectionfactory.h"
+#include "webrtc/pc/rtcstatscollector.h"
+#include "webrtc/pc/rtpreceiver.h"
+#include "webrtc/pc/rtpsender.h"
+#include "webrtc/pc/statscollector.h"
+#include "webrtc/pc/streamcollection.h"
+#include "webrtc/pc/webrtcsession.h"
namespace webrtc {
@@ -452,4 +452,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_PEERCONNECTION_H_
+#endif // WEBRTC_PC_PEERCONNECTION_H_
diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/pc/peerconnection_unittest.cc
similarity index 99%
rename from webrtc/api/peerconnection_unittest.cc
rename to webrtc/pc/peerconnection_unittest.cc
index c27e53f..a172a60 100644
--- a/webrtc/api/peerconnection_unittest.cc
+++ b/webrtc/pc/peerconnection_unittest.cc
@@ -17,19 +17,10 @@
#include <utility>
#include <vector>
-#include "webrtc/api/dtmfsender.h"
#include "webrtc/api/fakemetricsobserver.h"
-#include "webrtc/api/localaudiosource.h"
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/peerconnection.h"
-#include "webrtc/api/peerconnectionfactory.h"
#include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/api/test/fakeaudiocapturemodule.h"
#include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/api/test/fakeperiodicvideocapturer.h"
-#include "webrtc/api/test/fakertccertificategenerator.h"
-#include "webrtc/api/test/fakevideotrackrenderer.h"
-#include "webrtc/api/test/mockpeerconnectionobservers.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
@@ -44,7 +35,16 @@
#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/p2p/base/testturnserver.h"
#include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/pc/dtmfsender.h"
+#include "webrtc/pc/localaudiosource.h"
#include "webrtc/pc/mediasession.h"
+#include "webrtc/pc/peerconnection.h"
+#include "webrtc/pc/peerconnectionfactory.h"
+#include "webrtc/pc/test/fakeaudiocapturemodule.h"
+#include "webrtc/pc/test/fakeperiodicvideocapturer.h"
+#include "webrtc/pc/test/fakertccertificategenerator.h"
+#include "webrtc/pc/test/fakevideotrackrenderer.h"
+#include "webrtc/pc/test/mockpeerconnectionobservers.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
diff --git a/webrtc/api/peerconnectionendtoend_unittest.cc b/webrtc/pc/peerconnectionendtoend_unittest.cc
similarity index 98%
rename from webrtc/api/peerconnectionendtoend_unittest.cc
rename to webrtc/pc/peerconnectionendtoend_unittest.cc
index 4110db0..4327e1d 100644
--- a/webrtc/api/peerconnectionendtoend_unittest.cc
+++ b/webrtc/pc/peerconnectionendtoend_unittest.cc
@@ -10,12 +10,6 @@
#include <memory>
-#include "webrtc/api/test/peerconnectiontestwrapper.h"
-// Notice that mockpeerconnectionobservers.h must be included after the above!
-#include "webrtc/api/test/mockpeerconnectionobservers.h"
-#ifdef WEBRTC_ANDROID
-#include "webrtc/api/test/androidtestinitializer.h"
-#endif
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ssladapter.h"
@@ -23,6 +17,12 @@
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
+#ifdef WEBRTC_ANDROID
+#include "webrtc/pc/test/androidtestinitializer.h"
+#endif
+#include "webrtc/pc/test/peerconnectiontestwrapper.h"
+// Notice that mockpeerconnectionobservers.h must be included after the above!
+#include "webrtc/pc/test/mockpeerconnectionobservers.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
diff --git a/webrtc/api/peerconnectionfactory.cc b/webrtc/pc/peerconnectionfactory.cc
similarity index 97%
rename from webrtc/api/peerconnectionfactory.cc
rename to webrtc/pc/peerconnectionfactory.cc
index 32f461e..99a1daa 100644
--- a/webrtc/api/peerconnectionfactory.cc
+++ b/webrtc/pc/peerconnectionfactory.cc
@@ -8,22 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/peerconnectionfactory.h"
+#include "webrtc/pc/peerconnectionfactory.h"
#include <utility>
-#include "webrtc/api/audiotrack.h"
-#include "webrtc/api/localaudiosource.h"
#include "webrtc/api/mediaconstraintsinterface.h"
-#include "webrtc/api/mediastream.h"
#include "webrtc/api/mediastreamproxy.h"
#include "webrtc/api/mediastreamtrackproxy.h"
-#include "webrtc/api/peerconnection.h"
#include "webrtc/api/peerconnectionfactoryproxy.h"
#include "webrtc/api/peerconnectionproxy.h"
-#include "webrtc/api/videocapturertracksource.h"
#include "webrtc/api/videosourceproxy.h"
-#include "webrtc/api/videotrack.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/media/engine/webrtcmediaengine.h"
@@ -33,6 +27,12 @@
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/p2p/base/basicpacketsocketfactory.h"
#include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/pc/audiotrack.h"
+#include "webrtc/pc/localaudiosource.h"
+#include "webrtc/pc/mediastream.h"
+#include "webrtc/pc/peerconnection.h"
+#include "webrtc/pc/videocapturertracksource.h"
+#include "webrtc/pc/videotrack.h"
namespace webrtc {
diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/pc/peerconnectionfactory.h
similarity index 96%
rename from webrtc/api/peerconnectionfactory.h
rename to webrtc/pc/peerconnectionfactory.h
index 4b412ba..6e77d91 100644
--- a/webrtc/api/peerconnectionfactory.h
+++ b/webrtc/pc/peerconnectionfactory.h
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_PEERCONNECTIONFACTORY_H_
-#define WEBRTC_API_PEERCONNECTIONFACTORY_H_
+#ifndef WEBRTC_PC_PEERCONNECTIONFACTORY_H_
+#define WEBRTC_PC_PEERCONNECTIONFACTORY_H_
#include <memory>
#include <string>
-#include "webrtc/api/mediacontroller.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/rtccertificategenerator.h"
#include "webrtc/pc/channelmanager.h"
+#include "webrtc/pc/mediacontroller.h"
namespace rtc {
class BasicNetworkManager;
@@ -143,4 +143,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_
+#endif // WEBRTC_PC_PEERCONNECTIONFACTORY_H_
diff --git a/webrtc/api/peerconnectionfactory_unittest.cc b/webrtc/pc/peerconnectionfactory_unittest.cc
similarity index 98%
rename from webrtc/api/peerconnectionfactory_unittest.cc
rename to webrtc/pc/peerconnectionfactory_unittest.cc
index 8b5b335..a178998 100644
--- a/webrtc/api/peerconnectionfactory_unittest.cc
+++ b/webrtc/pc/peerconnectionfactory_unittest.cc
@@ -13,18 +13,18 @@
#include <utility>
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/peerconnectionfactory.h"
-#ifdef WEBRTC_ANDROID
-#include "webrtc/api/test/androidtestinitializer.h"
-#endif
-#include "webrtc/api/test/fakertccertificategenerator.h"
-#include "webrtc/api/test/fakevideotrackrenderer.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/thread.h"
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/engine/webrtccommon.h"
#include "webrtc/media/engine/webrtcvoe.h"
#include "webrtc/p2p/base/fakeportallocator.h"
+#include "webrtc/pc/peerconnectionfactory.h"
+#ifdef WEBRTC_ANDROID
+#include "webrtc/pc/test/androidtestinitializer.h"
+#endif
+#include "webrtc/pc/test/fakertccertificategenerator.h"
+#include "webrtc/pc/test/fakevideotrackrenderer.h"
using webrtc::DataChannelInterface;
using webrtc::FakeVideoTrackRenderer;
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/pc/peerconnectioninterface_unittest.cc
similarity index 99%
rename from webrtc/api/peerconnectioninterface_unittest.cc
rename to webrtc/pc/peerconnectioninterface_unittest.cc
index be3825c..09395a4 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/pc/peerconnectioninterface_unittest.cc
@@ -13,22 +13,12 @@
#include <string>
#include <utility>
-#include "webrtc/api/audiotrack.h"
#include "webrtc/api/jsepsessiondescription.h"
-#include "webrtc/api/mediastream.h"
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/peerconnection.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/api/rtpsenderinterface.h"
-#include "webrtc/api/streamcollection.h"
#include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/api/test/fakertccertificategenerator.h"
-#include "webrtc/api/test/fakevideotracksource.h"
-#include "webrtc/api/test/mockpeerconnectionobservers.h"
-#include "webrtc/api/test/testsdpstrings.h"
-#include "webrtc/api/videocapturertracksource.h"
-#include "webrtc/api/videotrack.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslstreamadapter.h"
@@ -38,11 +28,21 @@
#include "webrtc/media/sctp/sctptransportinternal.h"
#include "webrtc/p2p/base/fakeportallocator.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
+#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/mediasession.h"
+#include "webrtc/pc/mediastream.h"
+#include "webrtc/pc/peerconnection.h"
+#include "webrtc/pc/streamcollection.h"
+#include "webrtc/pc/test/fakertccertificategenerator.h"
+#include "webrtc/pc/test/fakevideotracksource.h"
+#include "webrtc/pc/test/mockpeerconnectionobservers.h"
+#include "webrtc/pc/test/testsdpstrings.h"
+#include "webrtc/pc/videocapturertracksource.h"
+#include "webrtc/pc/videotrack.h"
#include "webrtc/test/gmock.h"
#ifdef WEBRTC_ANDROID
-#include "webrtc/api/test/androidtestinitializer.h"
+#include "webrtc/pc/test/androidtestinitializer.h"
#endif
static const char kStreamLabel1[] = "local_stream_1";
diff --git a/webrtc/api/proxy_unittest.cc b/webrtc/pc/proxy_unittest.cc
similarity index 100%
rename from webrtc/api/proxy_unittest.cc
rename to webrtc/pc/proxy_unittest.cc
diff --git a/webrtc/api/quicdatachannel.cc b/webrtc/pc/quicdatachannel.cc
similarity index 99%
rename from webrtc/api/quicdatachannel.cc
rename to webrtc/pc/quicdatachannel.cc
index 2ab2a34..8907ce7 100644
--- a/webrtc/api/quicdatachannel.cc
+++ b/webrtc/pc/quicdatachannel.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/quicdatachannel.h"
+#include "webrtc/pc/quicdatachannel.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/bytebuffer.h"
diff --git a/webrtc/api/quicdatachannel.h b/webrtc/pc/quicdatachannel.h
similarity index 98%
rename from webrtc/api/quicdatachannel.h
rename to webrtc/pc/quicdatachannel.h
index 18a10ac..2f6288b 100644
--- a/webrtc/api/quicdatachannel.h
+++ b/webrtc/pc/quicdatachannel.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_QUICDATACHANNEL_H_
-#define WEBRTC_API_QUICDATACHANNEL_H_
+#ifndef WEBRTC_PC_QUICDATACHANNEL_H_
+#define WEBRTC_PC_QUICDATACHANNEL_H_
#include <string>
#include <unordered_map>
@@ -217,4 +217,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_QUICDATACHANNEL_H_
+#endif // WEBRTC_PC_QUICDATACHANNEL_H_
diff --git a/webrtc/api/quicdatachannel_unittest.cc b/webrtc/pc/quicdatachannel_unittest.cc
similarity index 99%
rename from webrtc/api/quicdatachannel_unittest.cc
rename to webrtc/pc/quicdatachannel_unittest.cc
index 7245ccf..649c36f 100644
--- a/webrtc/api/quicdatachannel_unittest.cc
+++ b/webrtc/pc/quicdatachannel_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/quicdatachannel.h"
+#include "webrtc/pc/quicdatachannel.h"
#include <map>
#include <memory>
diff --git a/webrtc/api/quicdatatransport.cc b/webrtc/pc/quicdatatransport.cc
similarity index 99%
rename from webrtc/api/quicdatatransport.cc
rename to webrtc/pc/quicdatatransport.cc
index 44143cc..8b0a93f 100644
--- a/webrtc/api/quicdatatransport.cc
+++ b/webrtc/pc/quicdatatransport.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/quicdatatransport.h"
+#include "webrtc/pc/quicdatatransport.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/logging.h"
diff --git a/webrtc/api/quicdatatransport.h b/webrtc/pc/quicdatatransport.h
similarity index 96%
rename from webrtc/api/quicdatatransport.h
rename to webrtc/pc/quicdatatransport.h
index b8d3f84..6b35a10 100644
--- a/webrtc/api/quicdatatransport.h
+++ b/webrtc/pc/quicdatatransport.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_QUICDATATRANSPORT_H_
-#define WEBRTC_API_QUICDATATRANSPORT_H_
+#ifndef WEBRTC_PC_QUICDATATRANSPORT_H_
+#define WEBRTC_PC_QUICDATATRANSPORT_H_
#include <string>
#include <unordered_map>
#include "webrtc/api/datachannelinterface.h"
-#include "webrtc/api/quicdatachannel.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
#include "webrtc/p2p/base/transportcontroller.h"
+#include "webrtc/pc/quicdatachannel.h"
namespace cricket {
class QuicTransportChannel;
@@ -112,4 +112,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_QUICDATATRANSPORT_H_
+#endif // WEBRTC_PC_QUICDATATRANSPORT_H_
diff --git a/webrtc/api/quicdatatransport_unittest.cc b/webrtc/pc/quicdatatransport_unittest.cc
similarity index 99%
rename from webrtc/api/quicdatatransport_unittest.cc
rename to webrtc/pc/quicdatatransport_unittest.cc
index a9c605f..af28159 100644
--- a/webrtc/api/quicdatatransport_unittest.cc
+++ b/webrtc/pc/quicdatatransport_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/quicdatatransport.h"
+#include "webrtc/pc/quicdatatransport.h"
#include <memory>
#include <set>
@@ -16,12 +16,12 @@
#include <unordered_map>
#include <vector>
-#include "webrtc/api/quicdatachannel.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/gunit.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/p2p/quic/quictransportchannel.h"
#include "webrtc/p2p/quic/reliablequicstream.h"
+#include "webrtc/pc/quicdatachannel.h"
using webrtc::DataBuffer;
using webrtc::DataChannelInit;
diff --git a/webrtc/api/remoteaudiosource.cc b/webrtc/pc/remoteaudiosource.cc
similarity index 98%
rename from webrtc/api/remoteaudiosource.cc
rename to webrtc/pc/remoteaudiosource.cc
index 47213fa..b256819 100644
--- a/webrtc/api/remoteaudiosource.cc
+++ b/webrtc/pc/remoteaudiosource.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/remoteaudiosource.h"
+#include "webrtc/pc/remoteaudiosource.h"
#include <algorithm>
#include <functional>
diff --git a/webrtc/api/remoteaudiosource.h b/webrtc/pc/remoteaudiosource.h
similarity index 94%
rename from webrtc/api/remoteaudiosource.h
rename to webrtc/pc/remoteaudiosource.h
index a67b895..b84a016 100644
--- a/webrtc/api/remoteaudiosource.h
+++ b/webrtc/pc/remoteaudiosource.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_
-#define WEBRTC_API_REMOTEAUDIOSOURCE_H_
+#ifndef WEBRTC_PC_REMOTEAUDIOSOURCE_H_
+#define WEBRTC_PC_REMOTEAUDIOSOURCE_H_
#include <list>
#include <string>
@@ -73,4 +73,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_
+#endif // WEBRTC_PC_REMOTEAUDIOSOURCE_H_
diff --git a/webrtc/api/rtcstats_integrationtest.cc b/webrtc/pc/rtcstats_integrationtest.cc
similarity index 99%
rename from webrtc/api/rtcstats_integrationtest.cc
rename to webrtc/pc/rtcstats_integrationtest.cc
index 6d14c46..6bba2cb 100644
--- a/webrtc/api/rtcstats_integrationtest.cc
+++ b/webrtc/pc/rtcstats_integrationtest.cc
@@ -15,14 +15,14 @@
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/stats/rtcstats_objects.h"
#include "webrtc/api/stats/rtcstatsreport.h"
-#include "webrtc/api/test/peerconnectiontestwrapper.h"
-#include "webrtc/api/test/rtcstatsobtainer.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/refcountedobject.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/pc/test/peerconnectiontestwrapper.h"
+#include "webrtc/pc/test/rtcstatsobtainer.h"
namespace webrtc {
diff --git a/webrtc/api/rtcstatscollector.cc b/webrtc/pc/rtcstatscollector.cc
similarity index 99%
rename from webrtc/api/rtcstatscollector.cc
rename to webrtc/pc/rtcstatscollector.cc
index e7929cb..1a25f32 100644
--- a/webrtc/api/rtcstatscollector.cc
+++ b/webrtc/pc/rtcstatscollector.cc
@@ -8,23 +8,23 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/rtcstatscollector.h"
+#include "webrtc/pc/rtcstatscollector.h"
#include <memory>
#include <sstream>
#include <utility>
#include <vector>
-#include "webrtc/api/peerconnection.h"
-#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/webrtcsession.h"
+#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/p2pconstants.h"
#include "webrtc/p2p/base/port.h"
+#include "webrtc/pc/peerconnection.h"
+#include "webrtc/pc/webrtcsession.h"
namespace webrtc {
diff --git a/webrtc/api/rtcstatscollector.h b/webrtc/pc/rtcstatscollector.h
similarity index 94%
rename from webrtc/api/rtcstatscollector.h
rename to webrtc/pc/rtcstatscollector.h
index cd3c55e..de0af33 100644
--- a/webrtc/api/rtcstatscollector.h
+++ b/webrtc/pc/rtcstatscollector.h
@@ -8,17 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_RTCSTATSCOLLECTOR_H_
-#define WEBRTC_API_RTCSTATSCOLLECTOR_H_
+#ifndef WEBRTC_PC_RTCSTATSCOLLECTOR_H_
+#define WEBRTC_PC_RTCSTATSCOLLECTOR_H_
#include <map>
#include <memory>
#include <set>
#include <vector>
-#include "webrtc/api/datachannel.h"
-#include "webrtc/api/datachannelinterface.h"
#include "webrtc/api/stats/rtcstats_objects.h"
+#include "webrtc/api/stats/rtcstatscollectorcallback.h"
#include "webrtc/api/stats/rtcstatsreport.h"
#include "webrtc/api/trackmediainfomap.h"
#include "webrtc/base/asyncinvoker.h"
@@ -29,6 +28,7 @@
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/pc/datachannel.h"
namespace cricket {
class Candidate;
@@ -44,14 +44,6 @@
struct SessionStats;
struct ChannelNamePairs;
-class RTCStatsCollectorCallback : public virtual rtc::RefCountInterface {
- public:
- virtual ~RTCStatsCollectorCallback() {}
-
- virtual void OnStatsDelivered(
- const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;
-};
-
// All public methods of the collector are to be called on the signaling thread.
// Stats are gathered on the signaling, worker and network threads
// asynchronously. The callback is invoked on the signaling thread. Resulting
@@ -197,4 +189,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_RTCSTATSCOLLECTOR_H_
+#endif // WEBRTC_PC_RTCSTATSCOLLECTOR_H_
diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/pc/rtcstatscollector_unittest.cc
similarity index 99%
rename from webrtc/api/rtcstatscollector_unittest.cc
rename to webrtc/pc/rtcstatscollector_unittest.cc
index c5e8a9c..fd6e1ba 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/pc/rtcstatscollector_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/rtcstatscollector.h"
+#include "webrtc/pc/rtcstatscollector.h"
#include <initializer_list>
#include <memory>
@@ -17,17 +17,11 @@
#include <vector>
#include "webrtc/api/jsepsessiondescription.h"
-#include "webrtc/api/mediastream.h"
-#include "webrtc/api/mediastreamtrack.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/api/stats/rtcstats_objects.h"
#include "webrtc/api/stats/rtcstatsreport.h"
-#include "webrtc/api/test/mock_datachannel.h"
-#include "webrtc/api/test/mock_peerconnection.h"
#include "webrtc/api/test/mock_rtpreceiver.h"
#include "webrtc/api/test/mock_rtpsender.h"
-#include "webrtc/api/test/mock_webrtcsession.h"
-#include "webrtc/api/test/rtcstatsobtainer.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/fakeclock.h"
#include "webrtc/base/fakesslidentity.h"
@@ -42,6 +36,12 @@
#include "webrtc/media/base/test/mock_mediachannel.h"
#include "webrtc/p2p/base/p2pconstants.h"
#include "webrtc/p2p/base/port.h"
+#include "webrtc/pc/mediastream.h"
+#include "webrtc/pc/mediastreamtrack.h"
+#include "webrtc/pc/test/mock_datachannel.h"
+#include "webrtc/pc/test/mock_peerconnection.h"
+#include "webrtc/pc/test/mock_webrtcsession.h"
+#include "webrtc/pc/test/rtcstatsobtainer.h"
using testing::_;
using testing::Invoke;
diff --git a/webrtc/api/rtpreceiver.cc b/webrtc/pc/rtpreceiver.cc
similarity index 98%
rename from webrtc/api/rtpreceiver.cc
rename to webrtc/pc/rtpreceiver.cc
index a5a764c..f57babb 100644
--- a/webrtc/api/rtpreceiver.cc
+++ b/webrtc/pc/rtpreceiver.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/rtpreceiver.h"
+#include "webrtc/pc/rtpreceiver.h"
#include "webrtc/api/mediastreamtrackproxy.h"
-#include "webrtc/api/audiotrack.h"
#include "webrtc/api/videosourceproxy.h"
-#include "webrtc/api/videotrack.h"
#include "webrtc/base/trace_event.h"
+#include "webrtc/pc/audiotrack.h"
+#include "webrtc/pc/videotrack.h"
namespace webrtc {
diff --git a/webrtc/api/rtpreceiver.h b/webrtc/pc/rtpreceiver.h
similarity index 95%
rename from webrtc/api/rtpreceiver.h
rename to webrtc/pc/rtpreceiver.h
index b6807c4..c135f22 100644
--- a/webrtc/api/rtpreceiver.h
+++ b/webrtc/pc/rtpreceiver.h
@@ -12,8 +12,8 @@
// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
// transport (provided by cricket::VoiceChannel/cricket::VideoChannel)
-#ifndef WEBRTC_API_RTPRECEIVER_H_
-#define WEBRTC_API_RTPRECEIVER_H_
+#ifndef WEBRTC_PC_RTPRECEIVER_H_
+#define WEBRTC_PC_RTPRECEIVER_H_
#include <stdint.h>
@@ -21,11 +21,12 @@
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/rtpreceiverinterface.h"
-#include "webrtc/api/remoteaudiosource.h"
-#include "webrtc/api/videotracksource.h"
+#include "webrtc/base/basictypes.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/media/base/videobroadcaster.h"
#include "webrtc/pc/channel.h"
+#include "webrtc/pc/remoteaudiosource.h"
+#include "webrtc/pc/videotracksource.h"
namespace webrtc {
@@ -154,4 +155,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_RTPRECEIVER_H_
+#endif // WEBRTC_PC_RTPRECEIVER_H_
diff --git a/webrtc/api/rtpsender.cc b/webrtc/pc/rtpsender.cc
similarity index 99%
rename from webrtc/api/rtpsender.cc
rename to webrtc/pc/rtpsender.cc
index b2e2461..3e8c7e1 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/pc/rtpsender.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/rtpsender.h"
+#include "webrtc/pc/rtpsender.h"
-#include "webrtc/api/localaudiosource.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/trace_event.h"
+#include "webrtc/pc/localaudiosource.h"
namespace webrtc {
diff --git a/webrtc/pc/rtpsender.h b/webrtc/pc/rtpsender.h
new file mode 100644
index 0000000..ed244b4
--- /dev/null
+++ b/webrtc/pc/rtpsender.h
@@ -0,0 +1,235 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains classes that implement RtpSenderInterface.
+// An RtpSender associates a MediaStreamTrackInterface with an underlying
+// transport (provided by AudioProviderInterface/VideoProviderInterface)
+
+#ifndef WEBRTC_PC_RTPSENDER_H_
+#define WEBRTC_PC_RTPSENDER_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/rtpsenderinterface.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/media/base/audiosource.h"
+#include "webrtc/pc/channel.h"
+#include "webrtc/pc/statscollector.h"
+
+namespace webrtc {
+
+// Internal interface used by PeerConnection.
+class RtpSenderInternal : public RtpSenderInterface {
+ public:
+ // Used to set the SSRC of the sender, once a local description has been set.
+ // If |ssrc| is 0, this indiates that the sender should disconnect from the
+ // underlying transport (this occurs if the sender isn't seen in a local
+ // description).
+ virtual void SetSsrc(uint32_t ssrc) = 0;
+
+ // TODO(deadbeef): Support one sender having multiple stream ids.
+ virtual void set_stream_id(const std::string& stream_id) = 0;
+ virtual std::string stream_id() const = 0;
+
+ virtual void Stop() = 0;
+};
+
+// LocalAudioSinkAdapter receives data callback as a sink to the local
+// AudioTrack, and passes the data to the sink of AudioSource.
+class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
+ public cricket::AudioSource {
+ public:
+ LocalAudioSinkAdapter();
+ virtual ~LocalAudioSinkAdapter();
+
+ private:
+ // AudioSinkInterface implementation.
+ void OnData(const void* audio_data,
+ int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames) override;
+
+ // cricket::AudioSource implementation.
+ void SetSink(cricket::AudioSource::Sink* sink) override;
+
+ cricket::AudioSource::Sink* sink_;
+ // Critical section protecting |sink_|.
+ rtc::CriticalSection lock_;
+};
+
+class AudioRtpSender : public ObserverInterface,
+ public rtc::RefCountedObject<RtpSenderInternal> {
+ public:
+ // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
+ // at the appropriate times.
+ // |channel| can be null if one does not exist yet.
+ AudioRtpSender(AudioTrackInterface* track,
+ const std::string& stream_id,
+ cricket::VoiceChannel* channel,
+ StatsCollector* stats);
+
+ // Randomly generates stream_id.
+ // |channel| can be null if one does not exist yet.
+ AudioRtpSender(AudioTrackInterface* track,
+ cricket::VoiceChannel* channel,
+ StatsCollector* stats);
+
+ // Randomly generates id and stream_id.
+ // |channel| can be null if one does not exist yet.
+ AudioRtpSender(cricket::VoiceChannel* channel, StatsCollector* stats);
+
+ virtual ~AudioRtpSender();
+
+ // ObserverInterface implementation
+ void OnChanged() override;
+
+ // RtpSenderInterface implementation
+ bool SetTrack(MediaStreamTrackInterface* track) override;
+ rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+ return track_;
+ }
+
+ uint32_t ssrc() const override { return ssrc_; }
+
+ cricket::MediaType media_type() const override {
+ return cricket::MEDIA_TYPE_AUDIO;
+ }
+
+ std::string id() const override { return id_; }
+
+ std::vector<std::string> stream_ids() const override {
+ std::vector<std::string> ret = {stream_id_};
+ return ret;
+ }
+
+ RtpParameters GetParameters() const override;
+ bool SetParameters(const RtpParameters& parameters) override;
+
+ // RtpSenderInternal implementation.
+ void SetSsrc(uint32_t ssrc) override;
+
+ void set_stream_id(const std::string& stream_id) override {
+ stream_id_ = stream_id;
+ }
+ std::string stream_id() const override { return stream_id_; }
+
+ void Stop() override;
+
+ // Does not take ownership.
+ // Should call SetChannel(nullptr) before |channel| is destroyed.
+ void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; }
+
+ private:
+ // TODO(nisse): Since SSRC == 0 is technically valid, figure out
+ // some other way to test if we have a valid SSRC.
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // AudioProviderInterface::SetAudioSend.
+ void SetAudioSend();
+ // Helper function to call SetAudioSend with "stop sending" parameters.
+ void ClearAudioSend();
+
+ std::string id_;
+ std::string stream_id_;
+ cricket::VoiceChannel* channel_ = nullptr;
+ StatsCollector* stats_;
+ rtc::scoped_refptr<AudioTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ bool stopped_ = false;
+
+ // Used to pass the data callback from the |track_| to the other end of
+ // cricket::AudioSource.
+ std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
+};
+
+class VideoRtpSender : public ObserverInterface,
+ public rtc::RefCountedObject<RtpSenderInternal> {
+ public:
+ // |channel| can be null if one does not exist yet.
+ VideoRtpSender(VideoTrackInterface* track,
+ const std::string& stream_id,
+ cricket::VideoChannel* channel);
+
+ // Randomly generates stream_id.
+ // |channel| can be null if one does not exist yet.
+ VideoRtpSender(VideoTrackInterface* track, cricket::VideoChannel* channel);
+
+ // Randomly generates id and stream_id.
+ // |channel| can be null if one does not exist yet.
+ explicit VideoRtpSender(cricket::VideoChannel* channel);
+
+ virtual ~VideoRtpSender();
+
+ // ObserverInterface implementation
+ void OnChanged() override;
+
+ // RtpSenderInterface implementation
+ bool SetTrack(MediaStreamTrackInterface* track) override;
+ rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+ return track_;
+ }
+
+ uint32_t ssrc() const override { return ssrc_; }
+
+ cricket::MediaType media_type() const override {
+ return cricket::MEDIA_TYPE_VIDEO;
+ }
+
+ std::string id() const override { return id_; }
+
+ std::vector<std::string> stream_ids() const override {
+ std::vector<std::string> ret = {stream_id_};
+ return ret;
+ }
+
+ RtpParameters GetParameters() const override;
+ bool SetParameters(const RtpParameters& parameters) override;
+
+ // RtpSenderInternal implementation.
+ void SetSsrc(uint32_t ssrc) override;
+
+ void set_stream_id(const std::string& stream_id) override {
+ stream_id_ = stream_id;
+ }
+ std::string stream_id() const override { return stream_id_; }
+
+ void Stop() override;
+
+ // Does not take ownership.
+ // Should call SetChannel(nullptr) before |channel| is destroyed.
+ void SetChannel(cricket::VideoChannel* channel) { channel_ = channel; }
+
+ private:
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // VideoProviderInterface::SetVideoSend.
+ void SetVideoSend();
+ // Helper function to call SetVideoSend with "stop sending" parameters.
+ void ClearVideoSend();
+
+ std::string id_;
+ std::string stream_id_;
+ cricket::VideoChannel* channel_ = nullptr;
+ rtc::scoped_refptr<VideoTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ VideoTrackInterface::ContentHint cached_track_content_hint_ =
+ VideoTrackInterface::ContentHint::kNone;
+ bool stopped_ = false;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_PC_RTPSENDER_H_
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/pc/rtpsenderreceiver_unittest.cc
similarity index 98%
rename from webrtc/api/rtpsenderreceiver_unittest.cc
rename to webrtc/pc/rtpsenderreceiver_unittest.cc
index fc1b748..86c0612 100644
--- a/webrtc/api/rtpsenderreceiver_unittest.cc
+++ b/webrtc/pc/rtpsenderreceiver_unittest.cc
@@ -12,24 +12,24 @@
#include <string>
#include <utility>
-#include "webrtc/api/audiotrack.h"
-#include "webrtc/api/fakemediacontroller.h"
-#include "webrtc/api/localaudiosource.h"
-#include "webrtc/api/mediastream.h"
-#include "webrtc/api/remoteaudiosource.h"
-#include "webrtc/api/rtpreceiver.h"
-#include "webrtc/api/rtpsender.h"
-#include "webrtc/api/streamcollection.h"
-#include "webrtc/api/test/fakevideotracksource.h"
-#include "webrtc/api/videotrack.h"
-#include "webrtc/api/videotracksource.h"
#include "webrtc/base/gunit.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/engine/fakewebrtccall.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
+#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/channelmanager.h"
+#include "webrtc/pc/fakemediacontroller.h"
+#include "webrtc/pc/localaudiosource.h"
+#include "webrtc/pc/mediastream.h"
+#include "webrtc/pc/remoteaudiosource.h"
+#include "webrtc/pc/rtpreceiver.h"
+#include "webrtc/pc/rtpsender.h"
+#include "webrtc/pc/streamcollection.h"
+#include "webrtc/pc/test/fakevideotracksource.h"
+#include "webrtc/pc/videotrack.h"
+#include "webrtc/pc/videotracksource.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
diff --git a/webrtc/api/sctputils.cc b/webrtc/pc/sctputils.cc
similarity index 99%
rename from webrtc/api/sctputils.cc
rename to webrtc/pc/sctputils.cc
index d6ea601..1c2f76f 100644
--- a/webrtc/api/sctputils.cc
+++ b/webrtc/pc/sctputils.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/sctputils.h"
+#include "webrtc/pc/sctputils.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/copyonwritebuffer.h"
diff --git a/webrtc/api/sctputils.h b/webrtc/pc/sctputils.h
similarity index 92%
rename from webrtc/api/sctputils.h
rename to webrtc/pc/sctputils.h
index 2fb1943..940bd36 100644
--- a/webrtc/api/sctputils.h
+++ b/webrtc/pc/sctputils.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_SCTPUTILS_H_
-#define WEBRTC_API_SCTPUTILS_H_
+#ifndef WEBRTC_PC_SCTPUTILS_H_
+#define WEBRTC_PC_SCTPUTILS_H_
#include <string>
@@ -38,4 +38,4 @@
void WriteDataChannelOpenAckMessage(rtc::CopyOnWriteBuffer* payload);
} // namespace webrtc
-#endif // WEBRTC_API_SCTPUTILS_H_
+#endif // WEBRTC_PC_SCTPUTILS_H_
diff --git a/webrtc/api/sctputils_unittest.cc b/webrtc/pc/sctputils_unittest.cc
similarity index 99%
rename from webrtc/api/sctputils_unittest.cc
rename to webrtc/pc/sctputils_unittest.cc
index 45a127d..dc56e94 100644
--- a/webrtc/api/sctputils_unittest.cc
+++ b/webrtc/pc/sctputils_unittest.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/sctputils.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/copyonwritebuffer.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/pc/sctputils.h"
class SctpUtilsTest : public testing::Test {
public:
diff --git a/webrtc/api/statscollector.cc b/webrtc/pc/statscollector.cc
similarity index 99%
rename from webrtc/api/statscollector.cc
rename to webrtc/pc/statscollector.cc
index 7761d10..3e0c2bf 100644
--- a/webrtc/api/statscollector.cc
+++ b/webrtc/pc/statscollector.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/statscollector.h"
+#include "webrtc/pc/statscollector.h"
#include <memory>
#include <utility>
#include <vector>
-#include "webrtc/api/peerconnection.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/checks.h"
#include "webrtc/pc/channel.h"
+#include "webrtc/pc/peerconnection.h"
namespace webrtc {
namespace {
diff --git a/webrtc/api/statscollector.h b/webrtc/pc/statscollector.h
similarity index 97%
rename from webrtc/api/statscollector.h
rename to webrtc/pc/statscollector.h
index 7c19821..bf895ed 100644
--- a/webrtc/api/statscollector.h
+++ b/webrtc/pc/statscollector.h
@@ -11,8 +11,8 @@
// This file contains a class used for gathering statistics from an ongoing
// libjingle PeerConnection.
-#ifndef WEBRTC_API_STATSCOLLECTOR_H_
-#define WEBRTC_API_STATSCOLLECTOR_H_
+#ifndef WEBRTC_PC_STATSCOLLECTOR_H_
+#define WEBRTC_PC_STATSCOLLECTOR_H_
#include <map>
#include <string>
@@ -21,7 +21,7 @@
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/statstypes.h"
-#include "webrtc/api/webrtcsession.h"
+#include "webrtc/pc/webrtcsession.h"
namespace webrtc {
@@ -149,4 +149,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_STATSCOLLECTOR_H_
+#endif // WEBRTC_PC_STATSCOLLECTOR_H_
diff --git a/webrtc/api/statscollector_unittest.cc b/webrtc/pc/statscollector_unittest.cc
similarity index 99%
rename from webrtc/api/statscollector_unittest.cc
rename to webrtc/pc/statscollector_unittest.cc
index f72e355..ef218c00 100644
--- a/webrtc/api/statscollector_unittest.cc
+++ b/webrtc/pc/statscollector_unittest.cc
@@ -13,18 +13,9 @@
#include <algorithm>
#include <memory>
-#include "webrtc/api/statscollector.h"
+#include "webrtc/pc/statscollector.h"
-#include "webrtc/api/mediastream.h"
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/mediastreamtrack.h"
-#include "webrtc/api/peerconnection.h"
-#include "webrtc/api/peerconnectionfactory.h"
-#include "webrtc/api/test/fakedatachannelprovider.h"
-#include "webrtc/api/test/fakevideotracksource.h"
-#include "webrtc/api/test/mock_peerconnection.h"
-#include "webrtc/api/test/mock_webrtcsession.h"
-#include "webrtc/api/videotrack.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
@@ -35,6 +26,15 @@
#include "webrtc/media/base/test/mock_mediachannel.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/channelmanager.h"
+#include "webrtc/pc/mediastream.h"
+#include "webrtc/pc/mediastreamtrack.h"
+#include "webrtc/pc/peerconnection.h"
+#include "webrtc/pc/peerconnectionfactory.h"
+#include "webrtc/pc/test/fakedatachannelprovider.h"
+#include "webrtc/pc/test/fakevideotracksource.h"
+#include "webrtc/pc/test/mock_peerconnection.h"
+#include "webrtc/pc/test/mock_webrtcsession.h"
+#include "webrtc/pc/videotrack.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
diff --git a/webrtc/pc/streamcollection.h b/webrtc/pc/streamcollection.h
new file mode 100644
index 0000000..b4d1f27
--- /dev/null
+++ b/webrtc/pc/streamcollection.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_PC_STREAMCOLLECTION_H_
+#define WEBRTC_PC_STREAMCOLLECTION_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/api/peerconnectioninterface.h"
+
+namespace webrtc {
+
+// Implementation of StreamCollection.
+class StreamCollection : public StreamCollectionInterface {
+ public:
+ static rtc::scoped_refptr<StreamCollection> Create() {
+ rtc::RefCountedObject<StreamCollection>* implementation =
+ new rtc::RefCountedObject<StreamCollection>();
+ return implementation;
+ }
+
+ static rtc::scoped_refptr<StreamCollection> Create(
+ StreamCollection* streams) {
+ rtc::RefCountedObject<StreamCollection>* implementation =
+ new rtc::RefCountedObject<StreamCollection>(streams);
+ return implementation;
+ }
+
+ virtual size_t count() {
+ return media_streams_.size();
+ }
+
+ virtual MediaStreamInterface* at(size_t index) {
+ return media_streams_.at(index);
+ }
+
+ virtual MediaStreamInterface* find(const std::string& label) {
+ for (StreamVector::iterator it = media_streams_.begin();
+ it != media_streams_.end(); ++it) {
+ if ((*it)->label().compare(label) == 0) {
+ return (*it);
+ }
+ }
+ return NULL;
+ }
+
+ virtual MediaStreamTrackInterface* FindAudioTrack(
+ const std::string& id) {
+ for (size_t i = 0; i < media_streams_.size(); ++i) {
+ MediaStreamTrackInterface* track = media_streams_[i]->FindAudioTrack(id);
+ if (track) {
+ return track;
+ }
+ }
+ return NULL;
+ }
+
+ virtual MediaStreamTrackInterface* FindVideoTrack(
+ const std::string& id) {
+ for (size_t i = 0; i < media_streams_.size(); ++i) {
+ MediaStreamTrackInterface* track = media_streams_[i]->FindVideoTrack(id);
+ if (track) {
+ return track;
+ }
+ }
+ return NULL;
+ }
+
+ void AddStream(MediaStreamInterface* stream) {
+ for (StreamVector::iterator it = media_streams_.begin();
+ it != media_streams_.end(); ++it) {
+ if ((*it)->label().compare(stream->label()) == 0)
+ return;
+ }
+ media_streams_.push_back(stream);
+ }
+
+ void RemoveStream(MediaStreamInterface* remove_stream) {
+ for (StreamVector::iterator it = media_streams_.begin();
+ it != media_streams_.end(); ++it) {
+ if ((*it)->label().compare(remove_stream->label()) == 0) {
+ media_streams_.erase(it);
+ break;
+ }
+ }
+ }
+
+ protected:
+ StreamCollection() {}
+ explicit StreamCollection(StreamCollection* original)
+ : media_streams_(original->media_streams_) {
+ }
+ typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> >
+ StreamVector;
+ StreamVector media_streams_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_PC_STREAMCOLLECTION_H_
diff --git a/webrtc/api/test/DEPS b/webrtc/pc/test/DEPS
similarity index 100%
rename from webrtc/api/test/DEPS
rename to webrtc/pc/test/DEPS
diff --git a/webrtc/api/test/androidtestinitializer.cc b/webrtc/pc/test/androidtestinitializer.cc
similarity index 96%
rename from webrtc/api/test/androidtestinitializer.cc
rename to webrtc/pc/test/androidtestinitializer.cc
index 7c1f5f8..c95d7bc 100644
--- a/webrtc/api/test/androidtestinitializer.cc
+++ b/webrtc/pc/test/androidtestinitializer.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/test/androidtestinitializer.h"
+#include "webrtc/pc/test/androidtestinitializer.h"
#include <pthread.h>
diff --git a/webrtc/api/test/androidtestinitializer.h b/webrtc/pc/test/androidtestinitializer.h
similarity index 76%
rename from webrtc/api/test/androidtestinitializer.h
rename to webrtc/pc/test/androidtestinitializer.h
index b56783e..48df9cc 100644
--- a/webrtc/api/test/androidtestinitializer.h
+++ b/webrtc/pc/test/androidtestinitializer.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_
-#define WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_
+#ifndef WEBRTC_PC_TEST_ANDROIDTESTINITIALIZER_H_
+#define WEBRTC_PC_TEST_ANDROIDTESTINITIALIZER_H_
namespace webrtc {
@@ -17,4 +17,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_
+#endif // WEBRTC_PC_TEST_ANDROIDTESTINITIALIZER_H_
diff --git a/webrtc/api/test/fakeaudiocapturemodule.cc b/webrtc/pc/test/fakeaudiocapturemodule.cc
similarity index 99%
rename from webrtc/api/test/fakeaudiocapturemodule.cc
rename to webrtc/pc/test/fakeaudiocapturemodule.cc
index c0b761f..01fd5ee 100644
--- a/webrtc/api/test/fakeaudiocapturemodule.cc
+++ b/webrtc/pc/test/fakeaudiocapturemodule.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/test/fakeaudiocapturemodule.h"
+#include "webrtc/pc/test/fakeaudiocapturemodule.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
diff --git a/webrtc/api/test/fakeaudiocapturemodule.h b/webrtc/pc/test/fakeaudiocapturemodule.h
similarity index 98%
rename from webrtc/api/test/fakeaudiocapturemodule.h
rename to webrtc/pc/test/fakeaudiocapturemodule.h
index 34aa7e0..72735b6 100644
--- a/webrtc/api/test/fakeaudiocapturemodule.h
+++ b/webrtc/pc/test/fakeaudiocapturemodule.h
@@ -17,8 +17,8 @@
// Note P postfix of a function indicates that it should only be called by the
// processing thread.
-#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
-#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#ifndef WEBRTC_PC_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#define WEBRTC_PC_TEST_FAKEAUDIOCAPTUREMODULE_H_
#include <memory>
@@ -278,4 +278,4 @@
rtc::CriticalSection crit_callback_;
};
-#endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#endif // WEBRTC_PC_TEST_FAKEAUDIOCAPTUREMODULE_H_
diff --git a/webrtc/api/test/fakeaudiocapturemodule_unittest.cc b/webrtc/pc/test/fakeaudiocapturemodule_unittest.cc
similarity index 99%
rename from webrtc/api/test/fakeaudiocapturemodule_unittest.cc
rename to webrtc/pc/test/fakeaudiocapturemodule_unittest.cc
index 53fb5f2..aa40af7 100644
--- a/webrtc/api/test/fakeaudiocapturemodule_unittest.cc
+++ b/webrtc/pc/test/fakeaudiocapturemodule_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/test/fakeaudiocapturemodule.h"
+#include "webrtc/pc/test/fakeaudiocapturemodule.h"
#include <algorithm>
diff --git a/webrtc/api/test/fakedatachannelprovider.h b/webrtc/pc/test/fakedatachannelprovider.h
similarity index 95%
rename from webrtc/api/test/fakedatachannelprovider.h
rename to webrtc/pc/test/fakedatachannelprovider.h
index 3e796a3..114f3de 100644
--- a/webrtc/api/test/fakedatachannelprovider.h
+++ b/webrtc/pc/test/fakedatachannelprovider.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_
-#define WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_
+#ifndef WEBRTC_PC_TEST_FAKEDATACHANNELPROVIDER_H_
+#define WEBRTC_PC_TEST_FAKEDATACHANNELPROVIDER_H_
-#include "webrtc/api/datachannel.h"
#include "webrtc/base/checks.h"
+#include "webrtc/pc/datachannel.h"
class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
public:
@@ -144,4 +144,4 @@
std::set<uint32_t> send_ssrcs_;
std::set<uint32_t> recv_ssrcs_;
};
-#endif // WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_
+#endif // WEBRTC_PC_TEST_FAKEDATACHANNELPROVIDER_H_
diff --git a/webrtc/api/test/fakeperiodicvideocapturer.h b/webrtc/pc/test/fakeperiodicvideocapturer.h
similarity index 93%
rename from webrtc/api/test/fakeperiodicvideocapturer.h
rename to webrtc/pc/test/fakeperiodicvideocapturer.h
index 2c16195..551df8b 100644
--- a/webrtc/api/test/fakeperiodicvideocapturer.h
+++ b/webrtc/pc/test/fakeperiodicvideocapturer.h
@@ -11,8 +11,8 @@
// FakePeriodicVideoCapturer implements a fake cricket::VideoCapturer that
// creates video frames periodically after it has been started.
-#ifndef WEBRTC_API_TEST_FAKEPERIODICVIDEOCAPTURER_H_
-#define WEBRTC_API_TEST_FAKEPERIODICVIDEOCAPTURER_H_
+#ifndef WEBRTC_PC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
+#define WEBRTC_PC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
#include <vector>
@@ -71,4 +71,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_FAKEPERIODICVIDEOCAPTURER_H_
+#endif // WEBRTC_PC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
diff --git a/webrtc/api/test/fakertccertificategenerator.h b/webrtc/pc/test/fakertccertificategenerator.h
similarity index 98%
rename from webrtc/api/test/fakertccertificategenerator.h
rename to webrtc/pc/test/fakertccertificategenerator.h
index cd1e4bf..ce36834 100644
--- a/webrtc/api/test/fakertccertificategenerator.h
+++ b/webrtc/pc/test/fakertccertificategenerator.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_FAKERTCCERTIFICATEGENERATOR_H_
-#define WEBRTC_API_TEST_FAKERTCCERTIFICATEGENERATOR_H_
+#ifndef WEBRTC_PC_TEST_FAKERTCCERTIFICATEGENERATOR_H_
+#define WEBRTC_PC_TEST_FAKERTCCERTIFICATEGENERATOR_H_
#include <memory>
#include <string>
@@ -228,4 +228,4 @@
int generated_certificates_ = 0;
};
-#endif // WEBRTC_API_TEST_FAKERTCCERTIFICATEGENERATOR_H_
+#endif // WEBRTC_PC_TEST_FAKERTCCERTIFICATEGENERATOR_H_
diff --git a/webrtc/api/test/fakevideotrackrenderer.h b/webrtc/pc/test/fakevideotrackrenderer.h
similarity index 85%
rename from webrtc/api/test/fakevideotrackrenderer.h
rename to webrtc/pc/test/fakevideotrackrenderer.h
index 0d9b248..9bf0b95 100644
--- a/webrtc/api/test/fakevideotrackrenderer.h
+++ b/webrtc/pc/test/fakevideotrackrenderer.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_FAKEVIDEOTRACKRENDERER_H_
-#define WEBRTC_API_TEST_FAKEVIDEOTRACKRENDERER_H_
+#ifndef WEBRTC_PC_TEST_FAKEVIDEOTRACKRENDERER_H_
+#define WEBRTC_PC_TEST_FAKEVIDEOTRACKRENDERER_H_
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/media/base/fakevideorenderer.h"
@@ -30,4 +30,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_FAKEVIDEOTRACKRENDERER_H_
+#endif // WEBRTC_PC_TEST_FAKEVIDEOTRACKRENDERER_H_
diff --git a/webrtc/api/test/fakevideotracksource.h b/webrtc/pc/test/fakevideotracksource.h
similarity index 87%
rename from webrtc/api/test/fakevideotracksource.h
rename to webrtc/pc/test/fakevideotracksource.h
index 0638a1f..aa5f3f5 100644
--- a/webrtc/api/test/fakevideotracksource.h
+++ b/webrtc/pc/test/fakevideotracksource.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_FAKEVIDEOTRACKSOURCE_H_
-#define WEBRTC_API_TEST_FAKEVIDEOTRACKSOURCE_H_
+#ifndef WEBRTC_PC_TEST_FAKEVIDEOTRACKSOURCE_H_
+#define WEBRTC_PC_TEST_FAKEVIDEOTRACKSOURCE_H_
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/videotracksource.h"
#include "webrtc/media/base/fakevideocapturer.h"
+#include "webrtc/pc/videotracksource.h"
namespace webrtc {
@@ -46,4 +46,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_FAKEVIDEOTRACKSOURCE_H_
+#endif // WEBRTC_PC_TEST_FAKEVIDEOTRACKSOURCE_H_
diff --git a/webrtc/api/test/mock_datachannel.h b/webrtc/pc/test/mock_datachannel.h
similarity index 91%
rename from webrtc/api/test/mock_datachannel.h
rename to webrtc/pc/test/mock_datachannel.h
index 1cb1948..105fcba 100644
--- a/webrtc/api/test/mock_datachannel.h
+++ b/webrtc/pc/test/mock_datachannel.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_MOCK_DATACHANNEL_H_
-#define WEBRTC_API_TEST_MOCK_DATACHANNEL_H_
+#ifndef WEBRTC_PC_TEST_MOCK_DATACHANNEL_H_
+#define WEBRTC_PC_TEST_MOCK_DATACHANNEL_H_
-#include "webrtc/api/datachannel.h"
+#include "webrtc/pc/datachannel.h"
#include "webrtc/test/gmock.h"
namespace webrtc {
@@ -55,4 +55,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_MOCK_DATACHANNEL_H_
+#endif // WEBRTC_PC_TEST_MOCK_DATACHANNEL_H_
diff --git a/webrtc/api/test/mock_peerconnection.h b/webrtc/pc/test/mock_peerconnection.h
similarity index 88%
rename from webrtc/api/test/mock_peerconnection.h
rename to webrtc/pc/test/mock_peerconnection.h
index ca39d2a..4dca0af 100644
--- a/webrtc/api/test/mock_peerconnection.h
+++ b/webrtc/pc/test/mock_peerconnection.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_MOCK_PEERCONNECTION_H_
-#define WEBRTC_API_TEST_MOCK_PEERCONNECTION_H_
+#ifndef WEBRTC_PC_TEST_MOCK_PEERCONNECTION_H_
+#define WEBRTC_PC_TEST_MOCK_PEERCONNECTION_H_
#include <vector>
-#include "webrtc/api/peerconnection.h"
+#include "webrtc/pc/peerconnection.h"
#include "webrtc/test/gmock.h"
namespace webrtc {
@@ -43,4 +43,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_MOCK_PEERCONNECTION_H_
+#endif // WEBRTC_PC_TEST_MOCK_PEERCONNECTION_H_
diff --git a/webrtc/api/test/mock_webrtcsession.h b/webrtc/pc/test/mock_webrtcsession.h
similarity index 93%
rename from webrtc/api/test/mock_webrtcsession.h
rename to webrtc/pc/test/mock_webrtcsession.h
index ae75035..d476377 100644
--- a/webrtc/api/test/mock_webrtcsession.h
+++ b/webrtc/pc/test/mock_webrtcsession.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_MOCK_WEBRTCSESSION_H_
-#define WEBRTC_API_TEST_MOCK_WEBRTCSESSION_H_
+#ifndef WEBRTC_PC_TEST_MOCK_WEBRTCSESSION_H_
+#define WEBRTC_PC_TEST_MOCK_WEBRTCSESSION_H_
#include <memory>
#include <string>
-#include "webrtc/api/webrtcsession.h"
+#include "webrtc/pc/webrtcsession.h"
#include "webrtc/media/sctp/sctptransportinternal.h"
#include "webrtc/test/gmock.h"
@@ -62,4 +62,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_MOCK_WEBRTCSESSION_H_
+#endif // WEBRTC_PC_TEST_MOCK_WEBRTCSESSION_H_
diff --git a/webrtc/api/test/mockpeerconnectionobservers.h b/webrtc/pc/test/mockpeerconnectionobservers.h
similarity index 97%
rename from webrtc/api/test/mockpeerconnectionobservers.h
rename to webrtc/pc/test/mockpeerconnectionobservers.h
index 1f000af..bde59a8 100644
--- a/webrtc/api/test/mockpeerconnectionobservers.h
+++ b/webrtc/pc/test/mockpeerconnectionobservers.h
@@ -10,8 +10,8 @@
// This file contains mock implementations of observers used in PeerConnection.
-#ifndef WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
-#define WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
+#ifndef WEBRTC_PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
+#define WEBRTC_PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
#include <memory>
#include <string>
@@ -227,4 +227,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
+#endif // WEBRTC_PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
diff --git a/webrtc/api/test/peerconnectiontestwrapper.cc b/webrtc/pc/test/peerconnectiontestwrapper.cc
similarity index 97%
rename from webrtc/api/test/peerconnectiontestwrapper.cc
rename to webrtc/pc/test/peerconnectiontestwrapper.cc
index b1eb586..6404e2b 100644
--- a/webrtc/api/test/peerconnectiontestwrapper.cc
+++ b/webrtc/pc/test/peerconnectiontestwrapper.cc
@@ -10,12 +10,12 @@
#include <utility>
-#include "webrtc/api/test/fakeperiodicvideocapturer.h"
-#include "webrtc/api/test/fakertccertificategenerator.h"
-#include "webrtc/api/test/mockpeerconnectionobservers.h"
-#include "webrtc/api/test/peerconnectiontestwrapper.h"
#include "webrtc/base/gunit.h"
#include "webrtc/p2p/base/fakeportallocator.h"
+#include "webrtc/pc/test/fakeperiodicvideocapturer.h"
+#include "webrtc/pc/test/fakertccertificategenerator.h"
+#include "webrtc/pc/test/mockpeerconnectionobservers.h"
+#include "webrtc/pc/test/peerconnectiontestwrapper.h"
static const char kStreamLabelBase[] = "stream_label";
static const char kVideoTrackLabelBase[] = "video_track";
diff --git a/webrtc/api/test/peerconnectiontestwrapper.h b/webrtc/pc/test/peerconnectiontestwrapper.h
similarity index 94%
rename from webrtc/api/test/peerconnectiontestwrapper.h
rename to webrtc/pc/test/peerconnectiontestwrapper.h
index 6433e8f..59e6968 100644
--- a/webrtc/api/test/peerconnectiontestwrapper.h
+++ b/webrtc/pc/test/peerconnectiontestwrapper.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
-#define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
+#ifndef WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_
+#define WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_
#include <memory>
#include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/api/test/fakeaudiocapturemodule.h"
#include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/api/test/fakevideotrackrenderer.h"
#include "webrtc/base/sigslot.h"
+#include "webrtc/pc/test/fakeaudiocapturemodule.h"
+#include "webrtc/pc/test/fakevideotrackrenderer.h"
class PeerConnectionTestWrapper
: public webrtc::PeerConnectionObserver,
@@ -115,4 +115,4 @@
std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
};
-#endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
+#endif // WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_
diff --git a/webrtc/api/test/rtcstatsobtainer.h b/webrtc/pc/test/rtcstatsobtainer.h
similarity index 91%
rename from webrtc/api/test/rtcstatsobtainer.h
rename to webrtc/pc/test/rtcstatsobtainer.h
index aeae87c..d0d4b61 100644
--- a/webrtc/api/test/rtcstatsobtainer.h
+++ b/webrtc/pc/test/rtcstatsobtainer.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_TEST_RTCSTATSOBTAINER_H_
-#define WEBRTC_API_TEST_RTCSTATSOBTAINER_H_
+#ifndef WEBRTC_PC_TEST_RTCSTATSOBTAINER_H_
+#define WEBRTC_PC_TEST_RTCSTATSOBTAINER_H_
#include "webrtc/api/stats/rtcstatsreport.h"
#include "webrtc/base/gunit.h"
@@ -50,4 +50,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_RTCSTATSOBTAINER_H_
+#endif // WEBRTC_PC_TEST_RTCSTATSOBTAINER_H_
diff --git a/webrtc/api/test/testsdpstrings.h b/webrtc/pc/test/testsdpstrings.h
similarity index 97%
rename from webrtc/api/test/testsdpstrings.h
rename to webrtc/pc/test/testsdpstrings.h
index b203aad..ff097b2 100644
--- a/webrtc/api/test/testsdpstrings.h
+++ b/webrtc/pc/test/testsdpstrings.h
@@ -10,8 +10,8 @@
// This file contain SDP strings used for testing.
-#ifndef WEBRTC_API_TEST_TESTSDPSTRINGS_H_
-#define WEBRTC_API_TEST_TESTSDPSTRINGS_H_
+#ifndef WEBRTC_PC_TEST_TESTSDPSTRINGS_H_
+#define WEBRTC_PC_TEST_TESTSDPSTRINGS_H_
namespace webrtc {
@@ -129,4 +129,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_TEST_TESTSDPSTRINGS_H_
+#endif // WEBRTC_PC_TEST_TESTSDPSTRINGS_H_
diff --git a/webrtc/api/videocapturertracksource.cc b/webrtc/pc/videocapturertracksource.cc
similarity index 99%
rename from webrtc/api/videocapturertracksource.cc
rename to webrtc/pc/videocapturertracksource.cc
index 0d6d46d..771429a 100644
--- a/webrtc/api/videocapturertracksource.cc
+++ b/webrtc/pc/videocapturertracksource.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/videocapturertracksource.h"
+#include "webrtc/pc/videocapturertracksource.h"
#include <cstdlib>
#include <string>
diff --git a/webrtc/api/videocapturertracksource.h b/webrtc/pc/videocapturertracksource.h
similarity index 93%
rename from webrtc/api/videocapturertracksource.h
rename to webrtc/pc/videocapturertracksource.h
index b0a2810..30991a1 100644
--- a/webrtc/api/videocapturertracksource.h
+++ b/webrtc/pc/videocapturertracksource.h
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_VIDEOCAPTURERTRACKSOURCE_H_
-#define WEBRTC_API_VIDEOCAPTURERTRACKSOURCE_H_
+#ifndef WEBRTC_PC_VIDEOCAPTURERTRACKSOURCE_H_
+#define WEBRTC_PC_VIDEOCAPTURERTRACKSOURCE_H_
#include <memory>
#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/videotracksource.h"
#include "webrtc/base/asyncinvoker.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/media/base/videocapturer.h"
#include "webrtc/media/base/videocommon.h"
+#include "webrtc/pc/videotracksource.h"
// VideoCapturerTrackSource implements VideoTrackSourceInterface. It owns a
// cricket::VideoCapturer and make sure the camera is started at a resolution
@@ -81,4 +81,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_VIDEOCAPTURERTRACKSOURCE_H_
+#endif // WEBRTC_PC_VIDEOCAPTURERTRACKSOURCE_H_
diff --git a/webrtc/api/videocapturertracksource_unittest.cc b/webrtc/pc/videocapturertracksource_unittest.cc
similarity index 99%
rename from webrtc/api/videocapturertracksource_unittest.cc
rename to webrtc/pc/videocapturertracksource_unittest.cc
index a0350d9..144d4b4 100644
--- a/webrtc/api/videocapturertracksource_unittest.cc
+++ b/webrtc/pc/videocapturertracksource_unittest.cc
@@ -13,11 +13,11 @@
#include <vector>
#include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/api/videocapturertracksource.h"
#include "webrtc/base/gunit.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/base/fakevideorenderer.h"
+#include "webrtc/pc/videocapturertracksource.h"
using webrtc::FakeConstraints;
using webrtc::VideoCapturerTrackSource;
diff --git a/webrtc/api/videotrack.cc b/webrtc/pc/videotrack.cc
similarity index 96%
rename from webrtc/api/videotrack.cc
rename to webrtc/pc/videotrack.cc
index 2fa2ba6..494d728 100644
--- a/webrtc/api/videotrack.cc
+++ b/webrtc/pc/videotrack.cc
@@ -8,14 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/videotrack.h"
+#include "webrtc/pc/videotrack.h"
#include <string>
namespace webrtc {
-const char MediaStreamTrackInterface::kVideoKind[] = "video";
-
VideoTrack::VideoTrack(const std::string& label,
VideoTrackSourceInterface* video_source)
: MediaStreamTrack<VideoTrackInterface>(label),
diff --git a/webrtc/api/videotrack.h b/webrtc/pc/videotrack.h
similarity index 92%
rename from webrtc/api/videotrack.h
rename to webrtc/pc/videotrack.h
index ebea848..7e75e14 100644
--- a/webrtc/api/videotrack.h
+++ b/webrtc/pc/videotrack.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_VIDEOTRACK_H_
-#define WEBRTC_API_VIDEOTRACK_H_
+#ifndef WEBRTC_PC_VIDEOTRACK_H_
+#define WEBRTC_PC_VIDEOTRACK_H_
#include <string>
#include <vector>
-#include "webrtc/api/mediastreamtrack.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/media/base/videosourcebase.h"
+#include "webrtc/pc/mediastreamtrack.h"
namespace webrtc {
@@ -57,4 +57,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_VIDEOTRACK_H_
+#endif // WEBRTC_PC_VIDEOTRACK_H_
diff --git a/webrtc/api/videotrack_unittest.cc b/webrtc/pc/videotrack_unittest.cc
similarity index 95%
rename from webrtc/api/videotrack_unittest.cc
rename to webrtc/pc/videotrack_unittest.cc
index 5320482..6f4330f 100644
--- a/webrtc/api/videotrack_unittest.cc
+++ b/webrtc/pc/videotrack_unittest.cc
@@ -11,12 +11,12 @@
#include <memory>
#include <string>
-#include "webrtc/api/test/fakevideotrackrenderer.h"
-#include "webrtc/api/videocapturertracksource.h"
-#include "webrtc/api/videotrack.h"
#include "webrtc/base/gunit.h"
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/pc/test/fakevideotrackrenderer.h"
+#include "webrtc/pc/videocapturertracksource.h"
+#include "webrtc/pc/videotrack.h"
using webrtc::FakeVideoTrackRenderer;
using webrtc::MediaSourceInterface;
diff --git a/webrtc/api/videotracksource.cc b/webrtc/pc/videotracksource.cc
similarity index 96%
rename from webrtc/api/videotracksource.cc
rename to webrtc/pc/videotracksource.cc
index 83ae493..f3ca8d7 100644
--- a/webrtc/api/videotracksource.cc
+++ b/webrtc/pc/videotracksource.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/videotracksource.h"
+#include "webrtc/pc/videotracksource.h"
#include <string>
diff --git a/webrtc/pc/videotracksource.h b/webrtc/pc/videotracksource.h
new file mode 100644
index 0000000..259b1c3
--- /dev/null
+++ b/webrtc/pc/videotracksource.h
@@ -0,0 +1,55 @@
+/*
+ * Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_PC_VIDEOTRACKSOURCE_H_
+#define WEBRTC_PC_VIDEOTRACKSOURCE_H_
+
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/notifier.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/media/base/videosinkinterface.h"
+
+// VideoTrackSource implements VideoTrackSourceInterface.
+namespace webrtc {
+
+class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
+ public:
+ VideoTrackSource(rtc::VideoSourceInterface<VideoFrame>* source, bool remote);
+ void SetState(SourceState new_state);
+ // OnSourceDestroyed clears this instance pointer to |source_|. It is useful
+ // when the underlying rtc::VideoSourceInterface is destroyed before the
+ // reference counted VideoTrackSource.
+ void OnSourceDestroyed();
+
+ SourceState state() const override { return state_; }
+ bool remote() const override { return remote_; }
+
+ bool is_screencast() const override { return false; }
+ rtc::Optional<bool> needs_denoising() const override {
+ return rtc::Optional<bool>(); }
+
+ bool GetStats(Stats* stats) override { return false; }
+
+ void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
+ const rtc::VideoSinkWants& wants) override;
+ void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override;
+
+ private:
+ rtc::ThreadChecker worker_thread_checker_;
+ rtc::VideoSourceInterface<VideoFrame>* source_;
+ cricket::VideoOptions options_;
+ SourceState state_;
+ const bool remote_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_PC_VIDEOTRACKSOURCE_H_
diff --git a/webrtc/api/webrtcsdp.cc b/webrtc/pc/webrtcsdp.cc
similarity index 99%
rename from webrtc/api/webrtcsdp.cc
rename to webrtc/pc/webrtcsdp.cc
index c79c851..d4003c2 100644
--- a/webrtc/api/webrtcsdp.cc
+++ b/webrtc/pc/webrtcsdp.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/webrtcsdp.h"
+#include "webrtc/pc/webrtcsdp.h"
#include <ctype.h>
#include <limits.h>
diff --git a/webrtc/api/webrtcsdp.h b/webrtc/pc/webrtcsdp.h
similarity index 96%
rename from webrtc/api/webrtcsdp.h
rename to webrtc/pc/webrtcsdp.h
index e7fdb34..beaf0c5 100644
--- a/webrtc/api/webrtcsdp.h
+++ b/webrtc/pc/webrtcsdp.h
@@ -17,8 +17,8 @@
// * draft-lennox-mmusic-sdp-source-selection-02 -
// Mechanisms for Media Source Selection in SDP
-#ifndef WEBRTC_API_WEBRTCSDP_H_
-#define WEBRTC_API_WEBRTCSDP_H_
+#ifndef WEBRTC_PC_WEBRTCSDP_H_
+#define WEBRTC_PC_WEBRTCSDP_H_
#include <string>
@@ -84,4 +84,4 @@
} // namespace webrtc
-#endif // WEBRTC_API_WEBRTCSDP_H_
+#endif // WEBRTC_PC_WEBRTCSDP_H_
diff --git a/webrtc/api/webrtcsdp_unittest.cc b/webrtc/pc/webrtcsdp_unittest.cc
similarity index 99%
rename from webrtc/api/webrtcsdp_unittest.cc
rename to webrtc/pc/webrtcsdp_unittest.cc
index d4accba..54eb22e 100644
--- a/webrtc/api/webrtcsdp_unittest.cc
+++ b/webrtc/pc/webrtcsdp_unittest.cc
@@ -14,10 +14,6 @@
#include <vector>
#include "webrtc/api/jsepsessiondescription.h"
-#ifdef WEBRTC_ANDROID
-#include "webrtc/api/test/androidtestinitializer.h"
-#endif
-#include "webrtc/api/webrtcsdp.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
@@ -30,6 +26,10 @@
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/p2p/base/p2pconstants.h"
#include "webrtc/pc/mediasession.h"
+#ifdef WEBRTC_ANDROID
+#include "webrtc/pc/test/androidtestinitializer.h"
+#endif
+#include "webrtc/pc/webrtcsdp.h"
using cricket::AudioCodec;
using cricket::AudioContentDescription;
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/pc/webrtcsession.cc
similarity index 99%
rename from webrtc/api/webrtcsession.cc
rename to webrtc/pc/webrtcsession.cc
index e98bcb0..def02d4 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/pc/webrtcsession.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/webrtcsession.h"
+#include "webrtc/pc/webrtcsession.h"
#include <limits.h>
@@ -21,8 +21,6 @@
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/api/sctputils.h"
-#include "webrtc/api/webrtcsessiondescriptionfactory.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
@@ -38,6 +36,8 @@
#include "webrtc/pc/channel.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/mediasession.h"
+#include "webrtc/pc/sctputils.h"
+#include "webrtc/pc/webrtcsessiondescriptionfactory.h"
#ifdef HAVE_QUIC
#include "webrtc/p2p/quic/quictransportchannel.h"
diff --git a/webrtc/api/webrtcsession.h b/webrtc/pc/webrtcsession.h
similarity index 98%
rename from webrtc/api/webrtcsession.h
rename to webrtc/pc/webrtcsession.h
index 0030342..c250833 100644
--- a/webrtc/api/webrtcsession.h
+++ b/webrtc/pc/webrtcsession.h
@@ -8,17 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_WEBRTCSESSION_H_
-#define WEBRTC_API_WEBRTCSESSION_H_
+#ifndef WEBRTC_PC_WEBRTCSESSION_H_
+#define WEBRTC_PC_WEBRTCSESSION_H_
#include <memory>
#include <set>
#include <string>
#include <vector>
-#include "webrtc/api/datachannel.h"
-#include "webrtc/api/dtmfsender.h"
-#include "webrtc/api/mediacontroller.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/statstypes.h"
#include "webrtc/base/constructormagic.h"
@@ -29,10 +26,13 @@
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/transportcontroller.h"
+#include "webrtc/pc/datachannel.h"
+#include "webrtc/pc/dtmfsender.h"
+#include "webrtc/pc/mediacontroller.h"
#include "webrtc/pc/mediasession.h"
#ifdef HAVE_QUIC
-#include "webrtc/api/quicdatatransport.h"
+#include "webrtc/pc/quicdatatransport.h"
#endif // HAVE_QUIC
namespace cricket {
@@ -651,4 +651,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_API_WEBRTCSESSION_H_
+#endif // WEBRTC_PC_WEBRTCSESSION_H_
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/pc/webrtcsession_unittest.cc
similarity index 99%
rename from webrtc/api/webrtcsession_unittest.cc
rename to webrtc/pc/webrtcsession_unittest.cc
index 1a7a026..053f64c 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/pc/webrtcsession_unittest.cc
@@ -12,17 +12,9 @@
#include <utility>
#include <vector>
-#include "webrtc/api/audiotrack.h"
-#include "webrtc/api/fakemediacontroller.h"
#include "webrtc/api/fakemetricsobserver.h"
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
-#include "webrtc/api/peerconnection.h"
-#include "webrtc/api/sctputils.h"
-#include "webrtc/api/test/fakertccertificategenerator.h"
-#include "webrtc/api/videotrack.h"
-#include "webrtc/api/webrtcsession.h"
-#include "webrtc/api/webrtcsessiondescriptionfactory.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/firewallsocketserver.h"
@@ -47,8 +39,16 @@
#include "webrtc/p2p/base/teststunserver.h"
#include "webrtc/p2p/base/testturnserver.h"
#include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/channelmanager.h"
+#include "webrtc/pc/fakemediacontroller.h"
#include "webrtc/pc/mediasession.h"
+#include "webrtc/pc/peerconnection.h"
+#include "webrtc/pc/sctputils.h"
+#include "webrtc/pc/test/fakertccertificategenerator.h"
+#include "webrtc/pc/videotrack.h"
+#include "webrtc/pc/webrtcsession.h"
+#include "webrtc/pc/webrtcsessiondescriptionfactory.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
diff --git a/webrtc/api/webrtcsessiondescriptionfactory.cc b/webrtc/pc/webrtcsessiondescriptionfactory.cc
similarity index 99%
rename from webrtc/api/webrtcsessiondescriptionfactory.cc
rename to webrtc/pc/webrtcsessiondescriptionfactory.cc
index 65ee7ef..168e5a2 100644
--- a/webrtc/api/webrtcsessiondescriptionfactory.cc
+++ b/webrtc/pc/webrtcsessiondescriptionfactory.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/webrtcsessiondescriptionfactory.h"
+#include "webrtc/pc/webrtcsessiondescriptionfactory.h"
#include <utility>
#include "webrtc/api/jsep.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/mediaconstraintsinterface.h"
-#include "webrtc/api/webrtcsession.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/sslidentity.h"
+#include "webrtc/pc/webrtcsession.h"
using cricket::MediaSessionOptions;
diff --git a/webrtc/api/webrtcsessiondescriptionfactory.h b/webrtc/pc/webrtcsessiondescriptionfactory.h
similarity index 97%
rename from webrtc/api/webrtcsessiondescriptionfactory.h
rename to webrtc/pc/webrtcsessiondescriptionfactory.h
index 314679d..7572f24 100644
--- a/webrtc/api/webrtcsessiondescriptionfactory.h
+++ b/webrtc/pc/webrtcsessiondescriptionfactory.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
-#define WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
+#ifndef WEBRTC_PC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
+#define WEBRTC_PC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
#include <memory>
@@ -166,4 +166,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
+#endif // WEBRTC_PC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn
index 4ccd530..e610242 100644
--- a/webrtc/sdk/BUILD.gn
+++ b/webrtc/sdk/BUILD.gn
@@ -219,7 +219,7 @@
deps += [
":rtc_sdk_common_objc",
- "../api:libjingle_peerconnection",
+ "../pc:libjingle_peerconnection",
]
if (rtc_build_libyuv) {
diff --git a/webrtc/sdk/android/BUILD.gn b/webrtc/sdk/android/BUILD.gn
index 2adcdb5..1df43e2 100644
--- a/webrtc/sdk/android/BUILD.gn
+++ b/webrtc/sdk/android/BUILD.gn
@@ -67,7 +67,7 @@
}
deps = [
- "//webrtc/api:libjingle_peerconnection",
+ "//webrtc/pc:libjingle_peerconnection",
]
if (rtc_build_libyuv) {
@@ -89,7 +89,7 @@
configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
deps = [
- "//webrtc/api:libjingle_peerconnection",
+ "//webrtc/pc:libjingle_peerconnection",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
]
@@ -105,7 +105,7 @@
deps = [
":libjingle_peerconnection_jni",
":libjingle_peerconnection_metrics_default_jni",
- "//webrtc/api:libjingle_peerconnection",
+ "//webrtc/pc:libjingle_peerconnection",
]
output_extension = "so"
}
diff --git a/webrtc/sdk/android/src/jni/DEPS b/webrtc/sdk/android/src/jni/DEPS
index e803553..95eff5e 100644
--- a/webrtc/sdk/android/src/jni/DEPS
+++ b/webrtc/sdk/android/src/jni/DEPS
@@ -1,5 +1,6 @@
include_rules = [
"+third_party/libyuv",
"+webrtc/common_video/libyuv/include/webrtc_libyuv.h",
+ "+webrtc/pc",
"+webrtc/voice_engine/include/voe_base.h",
]
\ No newline at end of file
diff --git a/webrtc/sdk/android/src/jni/androidvideotracksource_jni.cc b/webrtc/sdk/android/src/jni/androidvideotracksource_jni.cc
index 0b2cdccb..dd6bd1e 100644
--- a/webrtc/sdk/android/src/jni/androidvideotracksource_jni.cc
+++ b/webrtc/sdk/android/src/jni/androidvideotracksource_jni.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/api/videosourceproxy.h"
#include "webrtc/sdk/android/src/jni/androidvideotracksource.h"
#include "webrtc/sdk/android/src/jni/classreferenceholder.h"
-#include "webrtc/api/videosourceproxy.h"
// Identifiers are over 80 characters long so this is needed to fit them on one
// line.
diff --git a/webrtc/sdk/android/src/jni/peerconnection_jni.cc b/webrtc/sdk/android/src/jni/peerconnection_jni.cc
index 8c38a46..212fa43 100644
--- a/webrtc/sdk/android/src/jni/peerconnection_jni.cc
+++ b/webrtc/sdk/android/src/jni/peerconnection_jni.cc
@@ -45,19 +45,11 @@
#include "third_party/libyuv/include/libyuv/convert_from.h"
#include "third_party/libyuv/include/libyuv/scale.h"
-#include "webrtc/sdk/android/src/jni/androidmediadecoder_jni.h"
-#include "webrtc/sdk/android/src/jni/androidmediaencoder_jni.h"
-#include "webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.h"
-#include "webrtc/sdk/android/src/jni/androidvideotracksource.h"
-#include "webrtc/sdk/android/src/jni/classreferenceholder.h"
-#include "webrtc/sdk/android/src/jni/jni_helpers.h"
-#include "webrtc/sdk/android/src/jni/native_handle_impl.h"
#include "webrtc/api/mediaconstraintsinterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/api/videosourceproxy.h"
-#include "webrtc/api/webrtcsdp.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/event_tracer.h"
@@ -72,6 +64,14 @@
#include "webrtc/media/engine/webrtcvideodecoderfactory.h"
#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
#include "webrtc/system_wrappers/include/field_trial.h"
+#include "webrtc/pc/webrtcsdp.h"
+#include "webrtc/sdk/android/src/jni/androidmediadecoder_jni.h"
+#include "webrtc/sdk/android/src/jni/androidmediaencoder_jni.h"
+#include "webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.h"
+#include "webrtc/sdk/android/src/jni/androidvideotracksource.h"
+#include "webrtc/sdk/android/src/jni/classreferenceholder.h"
+#include "webrtc/sdk/android/src/jni/jni_helpers.h"
+#include "webrtc/sdk/android/src/jni/native_handle_impl.h"
#include "webrtc/system_wrappers/include/field_trial_default.h"
#include "webrtc/system_wrappers/include/logcat_trace_context.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/sdk/objc/DEPS b/webrtc/sdk/objc/DEPS
index 137f67a..46aeb8e 100644
--- a/webrtc/sdk/objc/DEPS
+++ b/webrtc/sdk/objc/DEPS
@@ -1,9 +1,10 @@
include_rules = [
"+WebRTC",
"+webrtc/api",
- "+webrtc/common_video/include",
"+webrtc/common_video/h264",
+ "+webrtc/common_video/include",
"+webrtc/media",
"+webrtc/modules/video_coding",
+ "+webrtc/pc",
"+webrtc/system_wrappers",
]
diff --git a/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory+Private.h b/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory+Private.h
index a7e453c..ed18077 100644
--- a/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory+Private.h
+++ b/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory+Private.h
@@ -10,7 +10,7 @@
#import "WebRTC/RTCPeerConnectionFactory.h"
-#include "webrtc/api/peerconnectionfactory.h"
+#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/scoped_ref_ptr.h"
NS_ASSUME_NONNULL_BEGIN
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index 224b212..1cb12d5 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -266,7 +266,7 @@
"sdp_parser_fuzzer.cc",
]
deps = [
- "../../api:libjingle_peerconnection",
+ "../../pc:libjingle_peerconnection",
]
seed_corpus = "corpora/sdp-corpus"
}
