diff options
Diffstat (limited to 'content')
50 files changed, 296 insertions, 295 deletions
diff --git a/content/BUILD.gn b/content/BUILD.gn index a49a887..a2bd489 100644 --- a/content/BUILD.gn +++ b/content/BUILD.gn @@ -81,17 +81,18 @@ config("libjingle_stub_config") { ] if (is_mac) { - defines += [ "OSX" ] + defines += [ "OSX", "WEBRTC_MAC" ] } else if (is_linux) { - defines += [ "LINUX" ] + defines += [ "LINUX", "WEBRTC_LINUX" ] } else if (is_android) { - defines += [ "ANDROID" ] + defines += [ "ANDROID", "WEBRTC_LINUX", "WEBRTC_ANDROID" ] } else if (is_win) { libs = [ "secur32.lib", "crypt32.lib", "iphlpapi.lib" ] + defines += [ "WEBRTC_WIN" ] } if (is_posix) { - defines += [ "POSIX" ] + defines += [ "POSIX", "WEBRTC_POSIX" ] } if (is_chromeos) { defines += [ "CHROMEOS" ] diff --git a/content/browser/renderer_host/p2p/socket_dispatcher_host.cc b/content/browser/renderer_host/p2p/socket_dispatcher_host.cc index bb8c1fb..8bab6af 100644 --- a/content/browser/renderer_host/p2p/socket_dispatcher_host.cc +++ b/content/browser/renderer_host/p2p/socket_dispatcher_host.cc @@ -269,7 +269,7 @@ void P2PSocketDispatcherHost::OnAcceptIncomingTcpConnection( void P2PSocketDispatcherHost::OnSend(int socket_id, const net::IPEndPoint& socket_address, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) { P2PSocketHost* socket = LookupSocket(socket_id); if (!socket) { diff --git a/content/browser/renderer_host/p2p/socket_dispatcher_host.h b/content/browser/renderer_host/p2p/socket_dispatcher_host.h index 21376bd..11045f56 100644 --- a/content/browser/renderer_host/p2p/socket_dispatcher_host.h +++ b/content/browser/renderer_host/p2p/socket_dispatcher_host.h @@ -22,7 +22,7 @@ namespace net { class URLRequestContextGetter; } -namespace talk_base { +namespace rtc { struct PacketOptions; } @@ -84,7 +84,7 @@ class P2PSocketDispatcherHost void OnSend(int socket_id, const net::IPEndPoint& socket_address, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id); void OnSetOption(int socket_id, P2PSocketOption option, int value); void OnDestroySocket(int socket_id); diff --git a/content/browser/renderer_host/p2p/socket_host.cc b/content/browser/renderer_host/p2p/socket_host.cc index 5fc3818..fd26298 100644 --- a/content/browser/renderer_host/p2p/socket_host.cc +++ b/content/browser/renderer_host/p2p/socket_host.cc @@ -11,10 +11,10 @@ #include "content/browser/renderer_host/render_process_host_impl.h" #include "content/public/browser/browser_thread.h" #include "crypto/hmac.h" -#include "third_party/libjingle/source/talk/base/asyncpacketsocket.h" -#include "third_party/libjingle/source/talk/base/byteorder.h" -#include "third_party/libjingle/source/talk/base/messagedigest.h" #include "third_party/libjingle/source/talk/p2p/base/stun.h" +#include "third_party/webrtc/base/asyncpacketsocket.h" +#include "third_party/webrtc/base/byteorder.h" +#include "third_party/webrtc/base/messagedigest.h" namespace { @@ -48,7 +48,7 @@ bool IsRtcpPacket(const char* data) { } bool IsTurnSendIndicationPacket(const char* data) { - uint16 type = talk_base::GetBE16(data); + uint16 type = rtc::GetBE16(data); return (type == cricket::TURN_SEND_INDICATION); } @@ -80,7 +80,7 @@ bool ValidateRtpHeader(const char* rtp, int length, size_t* header_length) { // Getting extension profile length. // Length is in 32 bit words. - uint16 extn_length = talk_base::GetBE16(rtp + 2) * 4; + uint16 extn_length = rtc::GetBE16(rtp + 2) * 4; // Verify input length against total header size. if (rtp_hdr_len_without_extn + kRtpExtnHdrLen + extn_length > length) { @@ -129,7 +129,7 @@ void UpdateAbsSendTimeExtnValue(char* extn_data, int len, // Assumes |len| is actual packet length + tag length. Updates HMAC at end of // the RTP packet. void UpdateRtpAuthTag(char* rtp, int len, - const talk_base::PacketOptions& options) { + const rtc::PacketOptions& options) { // If there is no key, return. if (options.packet_time_params.srtp_auth_key.empty()) return; @@ -176,7 +176,7 @@ namespace content { namespace packet_processing_helpers { bool ApplyPacketOptions(char* data, int length, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint32 abs_send_time) { DCHECK(data != NULL); DCHECK(length > 0); @@ -239,7 +239,7 @@ bool GetRtpPacketStartPositionAndLength(const char* packet, } rtp_begin = kTurnChannelHdrLen; - rtp_length = talk_base::GetBE16(&packet[2]); + rtp_length = rtc::GetBE16(&packet[2]); if (length < rtp_length + kTurnChannelHdrLen) { return false; } @@ -249,7 +249,7 @@ bool GetRtpPacketStartPositionAndLength(const char* packet, return false; } // Validate STUN message length. - int stun_msg_len = talk_base::GetBE16(&packet[2]); + int stun_msg_len = rtc::GetBE16(&packet[2]); if (stun_msg_len + P2PSocketHost::kStunHeaderSize != length) { return false; } @@ -275,8 +275,8 @@ bool GetRtpPacketStartPositionAndLength(const char* packet, // padding bits are ignored, and may be any value. uint16 attr_type, attr_length; // Getting attribute type and length. - attr_type = talk_base::GetBE16(&packet[rtp_begin]); - attr_length = talk_base::GetBE16( + attr_type = rtc::GetBE16(&packet[rtp_begin]); + attr_length = rtc::GetBE16( &packet[rtp_begin + sizeof(attr_type)]); // Checking for bogus attribute length. if (length < attr_length + rtp_begin) { @@ -353,9 +353,9 @@ bool UpdateRtpAbsSendTimeExtn(char* rtp, int length, rtp += rtp_hdr_len_without_extn; // Getting extension profile ID and length. - uint16 profile_id = talk_base::GetBE16(rtp); + uint16 profile_id = rtc::GetBE16(rtp); // Length is in 32 bit words. - uint16 extn_length = talk_base::GetBE16(rtp + 2) * 4; + uint16 extn_length = rtc::GetBE16(rtp + 2) * 4; rtp += kRtpExtnHdrLen; // Moving past extn header. diff --git a/content/browser/renderer_host/p2p/socket_host.h b/content/browser/renderer_host/p2p/socket_host.h index bfcbbd2..86eff5d 100644 --- a/content/browser/renderer_host/p2p/socket_host.h +++ b/content/browser/renderer_host/p2p/socket_host.h @@ -20,7 +20,7 @@ namespace net { class URLRequestContextGetter; } -namespace talk_base { +namespace rtc { struct PacketOptions; } @@ -34,7 +34,7 @@ namespace packet_processing_helpers { // if present with current time and 2. update HMAC in RTP packet. // If abs_send_time is 0, ApplyPacketOption will get current time from system. CONTENT_EXPORT bool ApplyPacketOptions(char* data, int length, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint32 abs_send_time); // Helper method which finds RTP ofset and length if the packet is encapsulated @@ -70,7 +70,7 @@ class CONTENT_EXPORT P2PSocketHost { // Sends |data| on the socket to |to|. virtual void Send(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) = 0; virtual P2PSocketHost* AcceptIncomingTcpConnection( diff --git a/content/browser/renderer_host/p2p/socket_host_tcp.cc b/content/browser/renderer_host/p2p/socket_host_tcp.cc index a5aac2f..2955b61 100644 --- a/content/browser/renderer_host/p2p/socket_host_tcp.cc +++ b/content/browser/renderer_host/p2p/socket_host_tcp.cc @@ -18,7 +18,7 @@ #include "net/socket/tcp_client_socket.h" #include "net/url_request/url_request_context.h" #include "net/url_request/url_request_context_getter.h" -#include "third_party/libjingle/source/talk/base/asyncpacketsocket.h" +#include "third_party/webrtc/base/asyncpacketsocket.h" namespace { @@ -330,7 +330,7 @@ void P2PSocketHostTcpBase::OnPacket(const std::vector<char>& data) { // but may be honored in the future. void P2PSocketHostTcpBase::Send(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) { if (!socket_) { // The Send message may be sent after the an OnError message was @@ -490,7 +490,7 @@ int P2PSocketHostTcp::ProcessInput(char* input, int input_len) { void P2PSocketHostTcp::DoSend(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options) { + const rtc::PacketOptions& options) { int size = kPacketHeaderSize + data.size(); scoped_refptr<net::DrainableIOBuffer> buffer = new net::DrainableIOBuffer(new net::IOBuffer(size), size); @@ -543,7 +543,7 @@ int P2PSocketHostStunTcp::ProcessInput(char* input, int input_len) { void P2PSocketHostStunTcp::DoSend(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options) { + const rtc::PacketOptions& options) { // Each packet is expected to have header (STUN/TURN ChannelData), where // header contains message type and and length of message. if (data.size() < kPacketHeaderSize + kPacketLengthOffset) { diff --git a/content/browser/renderer_host/p2p/socket_host_tcp.h b/content/browser/renderer_host/p2p/socket_host_tcp.h index f5ff8633..005bebb 100644 --- a/content/browser/renderer_host/p2p/socket_host_tcp.h +++ b/content/browser/renderer_host/p2p/socket_host_tcp.h @@ -42,7 +42,7 @@ class CONTENT_EXPORT P2PSocketHostTcpBase : public P2PSocketHost { const P2PHostAndIPEndPoint& remote_address) OVERRIDE; virtual void Send(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) OVERRIDE; virtual P2PSocketHost* AcceptIncomingTcpConnection( const net::IPEndPoint& remote_address, int id) OVERRIDE; @@ -53,7 +53,7 @@ class CONTENT_EXPORT P2PSocketHostTcpBase : public P2PSocketHost { virtual int ProcessInput(char* input, int input_len) = 0; virtual void DoSend(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options) = 0; + const rtc::PacketOptions& options) = 0; void WriteOrQueue(scoped_refptr<net::DrainableIOBuffer>& buffer); void OnPacket(const std::vector<char>& data); @@ -110,7 +110,7 @@ class CONTENT_EXPORT P2PSocketHostTcp : public P2PSocketHostTcpBase { virtual int ProcessInput(char* input, int input_len) OVERRIDE; virtual void DoSend(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options) OVERRIDE; + const rtc::PacketOptions& options) OVERRIDE; private: DISALLOW_COPY_AND_ASSIGN(P2PSocketHostTcp); }; @@ -132,7 +132,7 @@ class CONTENT_EXPORT P2PSocketHostStunTcp : public P2PSocketHostTcpBase { virtual int ProcessInput(char* input, int input_len) OVERRIDE; virtual void DoSend(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options) OVERRIDE; + const rtc::PacketOptions& options) OVERRIDE; private: int GetExpectedPacketSize(const char* data, int len, int* pad_bytes); diff --git a/content/browser/renderer_host/p2p/socket_host_tcp_server.cc b/content/browser/renderer_host/p2p/socket_host_tcp_server.cc index f3c4290..1017828 100644 --- a/content/browser/renderer_host/p2p/socket_host_tcp_server.cc +++ b/content/browser/renderer_host/p2p/socket_host_tcp_server.cc @@ -119,7 +119,7 @@ void P2PSocketHostTcpServer::OnAccepted(int result) { void P2PSocketHostTcpServer::Send(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) { NOTREACHED(); OnError(); diff --git a/content/browser/renderer_host/p2p/socket_host_tcp_server.h b/content/browser/renderer_host/p2p/socket_host_tcp_server.h index e050b00..72ce6443 100644 --- a/content/browser/renderer_host/p2p/socket_host_tcp_server.h +++ b/content/browser/renderer_host/p2p/socket_host_tcp_server.h @@ -38,7 +38,7 @@ class CONTENT_EXPORT P2PSocketHostTcpServer : public P2PSocketHost { const P2PHostAndIPEndPoint& remote_address) OVERRIDE; virtual void Send(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) OVERRIDE; virtual P2PSocketHost* AcceptIncomingTcpConnection( const net::IPEndPoint& remote_address, int id) OVERRIDE; diff --git a/content/browser/renderer_host/p2p/socket_host_tcp_unittest.cc b/content/browser/renderer_host/p2p/socket_host_tcp_unittest.cc index 24b77c3..f2d6146 100644 --- a/content/browser/renderer_host/p2p/socket_host_tcp_unittest.cc +++ b/content/browser/renderer_host/p2p/socket_host_tcp_unittest.cc @@ -89,7 +89,7 @@ TEST_F(P2PSocketHostTcpTest, SendStunNoAuth) { .Times(3) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); socket_host_->Send(dest_.ip_address, packet1, options, 0); @@ -121,7 +121,7 @@ TEST_F(P2PSocketHostTcpTest, ReceiveStun) { .Times(3) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); socket_host_->Send(dest_.ip_address, packet1, options, 0); @@ -168,7 +168,7 @@ TEST_F(P2PSocketHostTcpTest, SendDataNoAuth) { MatchMessage(static_cast<uint32>(P2PMsg_OnError::ID)))) .WillOnce(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet; CreateRandomPacket(&packet); socket_host_->Send(dest_.ip_address, packet, options, 0); @@ -194,7 +194,7 @@ TEST_F(P2PSocketHostTcpTest, SendAfterStunRequest) { .WillOnce(DoAll(DeleteArg<0>(), Return(true))); socket_->AppendInputData(&received_data[0], received_data.size()); - talk_base::PacketOptions options; + rtc::PacketOptions options; // Now we should be able to send any data to |dest_|. std::vector<char> packet; CreateRandomPacket(&packet); @@ -218,7 +218,7 @@ TEST_F(P2PSocketHostTcpTest, AsyncWrites) { .Times(2) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); @@ -254,7 +254,7 @@ TEST_F(P2PSocketHostTcpTest, SendDataWithPacketOptions) { .WillOnce(DoAll(DeleteArg<0>(), Return(true))); socket_->AppendInputData(&received_data[0], received_data.size()); - talk_base::PacketOptions options; + rtc::PacketOptions options; options.packet_time_params.rtp_sendtime_extension_id = 3; // Now we should be able to send any data to |dest_|. std::vector<char> packet; @@ -278,7 +278,7 @@ TEST_F(P2PSocketHostStunTcpTest, SendStunNoAuth) { .Times(3) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); socket_host_->Send(dest_.ip_address, packet1, options, 0); @@ -307,7 +307,7 @@ TEST_F(P2PSocketHostStunTcpTest, ReceiveStun) { .Times(3) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); socket_host_->Send(dest_.ip_address, packet1, options, 0); @@ -351,7 +351,7 @@ TEST_F(P2PSocketHostStunTcpTest, SendDataNoAuth) { MatchMessage(static_cast<uint32>(P2PMsg_OnError::ID)))) .WillOnce(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet; CreateRandomPacket(&packet); socket_host_->Send(dest_.ip_address, packet, options, 0); @@ -370,7 +370,7 @@ TEST_F(P2PSocketHostStunTcpTest, AsyncWrites) { .Times(2) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); socket_host_->Send(dest_.ip_address, packet1, options, 0); diff --git a/content/browser/renderer_host/p2p/socket_host_throttler.cc b/content/browser/renderer_host/p2p/socket_host_throttler.cc index 50a4dd0..0ef92eb 100644 --- a/content/browser/renderer_host/p2p/socket_host_throttler.cc +++ b/content/browser/renderer_host/p2p/socket_host_throttler.cc @@ -3,8 +3,8 @@ // found in the LICENSE file. #include "content/browser/renderer_host/p2p/socket_host_throttler.h" -#include "third_party/libjingle/source/talk/base/ratelimiter.h" -#include "third_party/libjingle/source/talk/base/timing.h" +#include "third_party/webrtc/base/ratelimiter.h" +#include "third_party/webrtc/base/timing.h" namespace content { @@ -16,19 +16,19 @@ const int kMaxIceMessageBandwidth = 256 * 1024; P2PMessageThrottler::P2PMessageThrottler() - : timing_(new talk_base::Timing()), - rate_limiter_(new talk_base::RateLimiter(kMaxIceMessageBandwidth, 1.0)) { + : timing_(new rtc::Timing()), + rate_limiter_(new rtc::RateLimiter(kMaxIceMessageBandwidth, 1.0)) { } P2PMessageThrottler::~P2PMessageThrottler() { } -void P2PMessageThrottler::SetTiming(scoped_ptr<talk_base::Timing> timing) { +void P2PMessageThrottler::SetTiming(scoped_ptr<rtc::Timing> timing) { timing_ = timing.Pass(); } void P2PMessageThrottler::SetSendIceBandwidth(int bandwidth_kbps) { - rate_limiter_.reset(new talk_base::RateLimiter(bandwidth_kbps, 1.0)); + rate_limiter_.reset(new rtc::RateLimiter(bandwidth_kbps, 1.0)); } bool P2PMessageThrottler::DropNextPacket(size_t packet_len) { diff --git a/content/browser/renderer_host/p2p/socket_host_throttler.h b/content/browser/renderer_host/p2p/socket_host_throttler.h index 166d300..a28a588 100644 --- a/content/browser/renderer_host/p2p/socket_host_throttler.h +++ b/content/browser/renderer_host/p2p/socket_host_throttler.h @@ -8,7 +8,7 @@ #include "base/memory/scoped_ptr.h" #include "content/common/content_export.h" -namespace talk_base { +namespace rtc { class RateLimiter; class Timing; } @@ -24,13 +24,13 @@ class CONTENT_EXPORT P2PMessageThrottler { P2PMessageThrottler(); virtual ~P2PMessageThrottler(); - void SetTiming(scoped_ptr<talk_base::Timing> timing); + void SetTiming(scoped_ptr<rtc::Timing> timing); bool DropNextPacket(size_t packet_len); void SetSendIceBandwidth(int bandwith_kbps); private: - scoped_ptr<talk_base::Timing> timing_; - scoped_ptr<talk_base::RateLimiter> rate_limiter_; + scoped_ptr<rtc::Timing> timing_; + scoped_ptr<rtc::RateLimiter> rate_limiter_; DISALLOW_COPY_AND_ASSIGN(P2PMessageThrottler); }; diff --git a/content/browser/renderer_host/p2p/socket_host_udp.cc b/content/browser/renderer_host/p2p/socket_host_udp.cc index 2af80aa..3833b62 100644 --- a/content/browser/renderer_host/p2p/socket_host_udp.cc +++ b/content/browser/renderer_host/p2p/socket_host_udp.cc @@ -15,7 +15,7 @@ #include "net/base/io_buffer.h" #include "net/base/net_errors.h" #include "net/base/net_util.h" -#include "third_party/libjingle/source/talk/base/asyncpacketsocket.h" +#include "third_party/webrtc/base/asyncpacketsocket.h" namespace { @@ -52,7 +52,7 @@ namespace content { P2PSocketHostUdp::PendingPacket::PendingPacket( const net::IPEndPoint& to, const std::vector<char>& content, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 id) : to(to), data(new net::IOBuffer(content.size())), @@ -186,7 +186,7 @@ void P2PSocketHostUdp::HandleReadResult(int result) { void P2PSocketHostUdp::Send(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) { if (!socket_) { // The Send message may be sent after the an OnError message was diff --git a/content/browser/renderer_host/p2p/socket_host_udp.h b/content/browser/renderer_host/p2p/socket_host_udp.h index b22795c..761ef1e 100644 --- a/content/browser/renderer_host/p2p/socket_host_udp.h +++ b/content/browser/renderer_host/p2p/socket_host_udp.h @@ -18,7 +18,7 @@ #include "content/common/p2p_socket_type.h" #include "net/base/ip_endpoint.h" #include "net/udp/udp_server_socket.h" -#include "third_party/libjingle/source/talk/base/asyncpacketsocket.h" +#include "third_party/webrtc/base/asyncpacketsocket.h" namespace content { @@ -36,7 +36,7 @@ class CONTENT_EXPORT P2PSocketHostUdp : public P2PSocketHost { const P2PHostAndIPEndPoint& remote_address) OVERRIDE; virtual void Send(const net::IPEndPoint& to, const std::vector<char>& data, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 packet_id) OVERRIDE; virtual P2PSocketHost* AcceptIncomingTcpConnection( const net::IPEndPoint& remote_address, int id) OVERRIDE; @@ -50,13 +50,13 @@ class CONTENT_EXPORT P2PSocketHostUdp : public P2PSocketHost { struct PendingPacket { PendingPacket(const net::IPEndPoint& to, const std::vector<char>& content, - const talk_base::PacketOptions& options, + const rtc::PacketOptions& options, uint64 id); ~PendingPacket(); net::IPEndPoint to; scoped_refptr<net::IOBuffer> data; int size; - talk_base::PacketOptions packet_options; + rtc::PacketOptions packet_options; uint64 id; }; diff --git a/content/browser/renderer_host/p2p/socket_host_udp_unittest.cc b/content/browser/renderer_host/p2p/socket_host_udp_unittest.cc index 6235cb0..2220471 100644 --- a/content/browser/renderer_host/p2p/socket_host_udp_unittest.cc +++ b/content/browser/renderer_host/p2p/socket_host_udp_unittest.cc @@ -17,7 +17,7 @@ #include "net/udp/datagram_server_socket.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "third_party/libjingle/source/talk/base/timing.h" +#include "third_party/webrtc/base/timing.h" using ::testing::_; using ::testing::DeleteArg; @@ -26,7 +26,7 @@ using ::testing::Return; namespace { -class FakeTiming : public talk_base::Timing { +class FakeTiming : public rtc::Timing { public: FakeTiming() : now_(0.0) {} virtual double TimerNow() OVERRIDE { return now_; } @@ -197,7 +197,7 @@ class P2PSocketHostUdpTest : public testing::Test { dest1_ = ParseAddress(kTestIpAddress1, kTestPort1); dest2_ = ParseAddress(kTestIpAddress2, kTestPort2); - scoped_ptr<talk_base::Timing> timing(new FakeTiming()); + scoped_ptr<rtc::Timing> timing(new FakeTiming()); throttler_.SetTiming(timing.Pass()); } @@ -221,7 +221,7 @@ TEST_F(P2PSocketHostUdpTest, SendStunNoAuth) { .Times(3) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); socket_host_->Send(dest1_, packet1, options, 0); @@ -247,7 +247,7 @@ TEST_F(P2PSocketHostUdpTest, SendDataNoAuth) { MatchMessage(static_cast<uint32>(P2PMsg_OnError::ID)))) .WillOnce(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet; CreateRandomPacket(&packet); socket_host_->Send(dest1_, packet, options, 0); @@ -271,7 +271,7 @@ TEST_F(P2PSocketHostUdpTest, SendAfterStunRequest) { MatchMessage(static_cast<uint32>(P2PMsg_OnSendComplete::ID)))) .WillOnce(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet; CreateRandomPacket(&packet); socket_host_->Send(dest1_, packet, options, 0); @@ -296,7 +296,7 @@ TEST_F(P2PSocketHostUdpTest, SendAfterStunResponse) { MatchMessage(static_cast<uint32>(P2PMsg_OnSendComplete::ID)))) .WillOnce(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet; CreateRandomPacket(&packet); socket_host_->Send(dest1_, packet, options, 0); @@ -317,7 +317,7 @@ TEST_F(P2PSocketHostUdpTest, SendAfterStunResponseDifferentHost) { socket_->ReceivePacket(dest1_, request_packet); // Should fail when trying to send the same packet to |dest2_|. - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet; CreateRandomPacket(&packet); EXPECT_CALL(sender_, Send( @@ -334,7 +334,7 @@ TEST_F(P2PSocketHostUdpTest, ThrottleAfterLimit) { .Times(2) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); throttler_.SetSendIceBandwidth(packet1.size() * 2); @@ -363,7 +363,7 @@ TEST_F(P2PSocketHostUdpTest, ThrottleAfterLimitAfterReceive) { .Times(4) .WillRepeatedly(DoAll(DeleteArg<0>(), Return(true))); - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> packet1; CreateStunRequest(&packet1); throttler_.SetSendIceBandwidth(packet1.size()); diff --git a/content/browser/renderer_host/p2p/socket_host_unittest.cc b/content/browser/renderer_host/p2p/socket_host_unittest.cc index 1404ced..fc96e4a 100644 --- a/content/browser/renderer_host/p2p/socket_host_unittest.cc +++ b/content/browser/renderer_host/p2p/socket_host_unittest.cc @@ -297,7 +297,7 @@ TEST(P2PSocketHostTest, TestUpdateAbsSendTimeExtensionInTurnSendIndication) { // without HMAC value in the packet. TEST(P2PSocketHostTest, TestApplyPacketOptionsWithDefaultValues) { unsigned char fake_tag[4] = { 0xba, 0xdd, 0xba, 0xdd }; - talk_base::PacketOptions options; + rtc::PacketOptions options; std::vector<char> rtp_packet; rtp_packet.resize(sizeof(kRtpMsgWithAbsSendTimeExtension) + 4); // tag length memcpy(&rtp_packet[0], kRtpMsgWithAbsSendTimeExtension, @@ -317,7 +317,7 @@ TEST(P2PSocketHostTest, TestApplyPacketOptionsWithDefaultValues) { // Veirfy HMAC is updated when packet option parameters are set. TEST(P2PSocketHostTest, TestApplyPacketOptionsWithAuthParams) { - talk_base::PacketOptions options; + rtc::PacketOptions options; options.packet_time_params.srtp_auth_key.resize(20); options.packet_time_params.srtp_auth_key.assign( kTestKey, kTestKey + sizeof(kTestKey)); @@ -348,7 +348,7 @@ TEST(P2PSocketHostTest, TestUpdateAbsSendTimeExtensionInRtpPacket) { // Verify we update both AbsSendTime extension header and HMAC. TEST(P2PSocketHostTest, TestApplyPacketOptionsWithAuthParamsAndAbsSendTime) { - talk_base::PacketOptions options; + rtc::PacketOptions options; options.packet_time_params.srtp_auth_key.resize(20); options.packet_time_params.srtp_auth_key.assign( kTestKey, kTestKey + sizeof(kTestKey)); diff --git a/content/common/p2p_messages.h b/content/common/p2p_messages.h index 26846dd..d01807f 100644 --- a/content/common/p2p_messages.h +++ b/content/common/p2p_messages.h @@ -10,7 +10,7 @@ #include "content/common/p2p_socket_type.h" #include "ipc/ipc_message_macros.h" #include "net/base/net_util.h" -#include "third_party/libjingle/source/talk/base/asyncpacketsocket.h" +#include "third_party/webrtc/base/asyncpacketsocket.h" #undef IPC_MESSAGE_EXPORT #define IPC_MESSAGE_EXPORT CONTENT_EXPORT @@ -20,9 +20,9 @@ IPC_ENUM_TRAITS_MAX_VALUE(content::P2PSocketType, content::P2P_SOCKET_TYPE_LAST) IPC_ENUM_TRAITS_MAX_VALUE(content::P2PSocketOption, content::P2P_SOCKET_OPT_MAX - 1) -IPC_ENUM_TRAITS_MIN_MAX_VALUE(talk_base::DiffServCodePoint, - talk_base::DSCP_NO_CHANGE, - talk_base::DSCP_CS7) +IPC_ENUM_TRAITS_MIN_MAX_VALUE(rtc::DiffServCodePoint, + rtc::DSCP_NO_CHANGE, + rtc::DSCP_CS7) IPC_STRUCT_TRAITS_BEGIN(net::NetworkInterface) IPC_STRUCT_TRAITS_MEMBER(name) @@ -30,14 +30,14 @@ IPC_STRUCT_TRAITS_BEGIN(net::NetworkInterface) IPC_STRUCT_TRAITS_MEMBER(address) IPC_STRUCT_TRAITS_END() -IPC_STRUCT_TRAITS_BEGIN(talk_base::PacketTimeUpdateParams) +IPC_STRUCT_TRAITS_BEGIN(rtc::PacketTimeUpdateParams) IPC_STRUCT_TRAITS_MEMBER(rtp_sendtime_extension_id) IPC_STRUCT_TRAITS_MEMBER(srtp_auth_key) IPC_STRUCT_TRAITS_MEMBER(srtp_auth_tag_len) IPC_STRUCT_TRAITS_MEMBER(srtp_packet_index) IPC_STRUCT_TRAITS_END() -IPC_STRUCT_TRAITS_BEGIN(talk_base::PacketOptions) +IPC_STRUCT_TRAITS_BEGIN(rtc::PacketOptions) IPC_STRUCT_TRAITS_MEMBER(dscp) IPC_STRUCT_TRAITS_MEMBER(packet_time_params) IPC_STRUCT_TRAITS_END() @@ -104,7 +104,7 @@ IPC_MESSAGE_CONTROL5(P2PHostMsg_Send, int /* socket_id */, net::IPEndPoint /* socket_address */, std::vector<char> /* data */, - talk_base::PacketOptions /* packet options */, + rtc::PacketOptions /* packet options */, uint64 /* packet_id */) IPC_MESSAGE_CONTROL1(P2PHostMsg_DestroySocket, diff --git a/content/renderer/media/media_stream_audio_processor_unittest.cc b/content/renderer/media/media_stream_audio_processor_unittest.cc index 267d1d2..867d48a 100644 --- a/content/renderer/media/media_stream_audio_processor_unittest.cc +++ b/content/renderer/media/media_stream_audio_processor_unittest.cc @@ -162,7 +162,7 @@ TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) { scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); scoped_refptr<MediaStreamAudioProcessor> audio_processor( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get())); EXPECT_FALSE(audio_processor->has_audio_processing()); @@ -182,7 +182,7 @@ TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); scoped_refptr<MediaStreamAudioProcessor> audio_processor( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get())); EXPECT_TRUE(audio_processor->has_audio_processing()); @@ -207,7 +207,7 @@ TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) { tab_constraint_factory.AddMandatory(kMediaStreamSource, tab_string); scoped_refptr<MediaStreamAudioProcessor> audio_processor( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( tab_constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get())); EXPECT_FALSE(audio_processor->has_audio_processing()); @@ -224,7 +224,7 @@ TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) { const std::string system_string = kMediaStreamSourceSystem; system_constraint_factory.AddMandatory(kMediaStreamSource, system_string); - audio_processor = new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + audio_processor = new rtc::RefCountedObject<MediaStreamAudioProcessor>( system_constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get()); EXPECT_FALSE(audio_processor->has_audio_processing()); @@ -241,7 +241,7 @@ TEST_F(MediaStreamAudioProcessorTest, TurnOffDefaultConstraints) { scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); scoped_refptr<MediaStreamAudioProcessor> audio_processor( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get())); EXPECT_FALSE(audio_processor->has_audio_processing()); @@ -357,7 +357,7 @@ TEST_F(MediaStreamAudioProcessorTest, TestAllSampleRates) { scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); scoped_refptr<MediaStreamAudioProcessor> audio_processor( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get())); EXPECT_TRUE(audio_processor->has_audio_processing()); @@ -398,7 +398,7 @@ TEST_F(MediaStreamAudioProcessorTest, GetAecDumpMessageFilter) { scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); scoped_refptr<MediaStreamAudioProcessor> audio_processor( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get())); @@ -418,7 +418,7 @@ TEST_F(MediaStreamAudioProcessorTest, TestStereoAudio) { scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); scoped_refptr<MediaStreamAudioProcessor> audio_processor( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( constraint_factory.CreateWebMediaConstraints(), 0, webrtc_audio_device.get())); EXPECT_FALSE(audio_processor->has_audio_processing()); diff --git a/content/renderer/media/mock_peer_connection_impl.cc b/content/renderer/media/mock_peer_connection_impl.cc index 7a09ea3..831571b 100644 --- a/content/renderer/media/mock_peer_connection_impl.cc +++ b/content/renderer/media/mock_peer_connection_impl.cc @@ -75,7 +75,7 @@ class MockStreamCollection : public webrtc::StreamCollectionInterface { virtual ~MockStreamCollection() {} private: - typedef std::vector<talk_base::scoped_refptr<MediaStreamInterface> > + typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> > StreamVector; StreamVector streams_; }; @@ -194,7 +194,7 @@ class MockDtmfSender : public DtmfSenderInterface { virtual ~MockDtmfSender() {} private: - talk_base::scoped_refptr<AudioTrackInterface> track_; + rtc::scoped_refptr<AudioTrackInterface> track_; DtmfSenderObserverInterface* observer_; std::string tones_; int duration_; @@ -207,8 +207,8 @@ const char MockPeerConnectionImpl::kDummyAnswer[] = "dummy answer"; MockPeerConnectionImpl::MockPeerConnectionImpl( MockPeerConnectionDependencyFactory* factory) : dependency_factory_(factory), - local_streams_(new talk_base::RefCountedObject<MockStreamCollection>), - remote_streams_(new talk_base::RefCountedObject<MockStreamCollection>), + local_streams_(new rtc::RefCountedObject<MockStreamCollection>), + remote_streams_(new rtc::RefCountedObject<MockStreamCollection>), hint_audio_(false), hint_video_(false), getstats_result_(true), @@ -221,12 +221,12 @@ MockPeerConnectionImpl::MockPeerConnectionImpl( MockPeerConnectionImpl::~MockPeerConnectionImpl() {} -talk_base::scoped_refptr<webrtc::StreamCollectionInterface> +rtc::scoped_refptr<webrtc::StreamCollectionInterface> MockPeerConnectionImpl::local_streams() { return local_streams_; } -talk_base::scoped_refptr<webrtc::StreamCollectionInterface> +rtc::scoped_refptr<webrtc::StreamCollectionInterface> MockPeerConnectionImpl::remote_streams() { return remote_streams_; } @@ -247,18 +247,18 @@ void MockPeerConnectionImpl::RemoveStream( local_streams_->RemoveStream(local_stream); } -talk_base::scoped_refptr<DtmfSenderInterface> +rtc::scoped_refptr<DtmfSenderInterface> MockPeerConnectionImpl::CreateDtmfSender(AudioTrackInterface* track) { if (!track) { return NULL; } - return new talk_base::RefCountedObject<MockDtmfSender>(track); + return new rtc::RefCountedObject<MockDtmfSender>(track); } -talk_base::scoped_refptr<webrtc::DataChannelInterface> +rtc::scoped_refptr<webrtc::DataChannelInterface> MockPeerConnectionImpl::CreateDataChannel(const std::string& label, const webrtc::DataChannelInit* config) { - return new talk_base::RefCountedObject<MockDataChannel>(label, config); + return new rtc::RefCountedObject<MockDataChannel>(label, config); } bool MockPeerConnectionImpl::GetStats( diff --git a/content/renderer/media/mock_peer_connection_impl.h b/content/renderer/media/mock_peer_connection_impl.h index d563746..0d7a847 100644 --- a/content/renderer/media/mock_peer_connection_impl.h +++ b/content/renderer/media/mock_peer_connection_impl.h @@ -24,18 +24,18 @@ class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface { explicit MockPeerConnectionImpl(MockPeerConnectionDependencyFactory* factory); // PeerConnectionInterface implementation. - virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface> + virtual rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() OVERRIDE; - virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface> + virtual rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() OVERRIDE; virtual bool AddStream( webrtc::MediaStreamInterface* local_stream, const webrtc::MediaConstraintsInterface* constraints) OVERRIDE; virtual void RemoveStream( webrtc::MediaStreamInterface* local_stream) OVERRIDE; - virtual talk_base::scoped_refptr<webrtc::DtmfSenderInterface> + virtual rtc::scoped_refptr<webrtc::DtmfSenderInterface> CreateDtmfSender(webrtc::AudioTrackInterface* track) OVERRIDE; - virtual talk_base::scoped_refptr<webrtc::DataChannelInterface> + virtual rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(const std::string& label, const webrtc::DataChannelInit* config) OVERRIDE; @@ -124,8 +124,8 @@ class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface { MockPeerConnectionDependencyFactory* dependency_factory_; std::string stream_label_; - talk_base::scoped_refptr<MockStreamCollection> local_streams_; - talk_base::scoped_refptr<MockStreamCollection> remote_streams_; + rtc::scoped_refptr<MockStreamCollection> local_streams_; + rtc::scoped_refptr<MockStreamCollection> remote_streams_; scoped_ptr<webrtc::SessionDescriptionInterface> local_desc_; scoped_ptr<webrtc::SessionDescriptionInterface> remote_desc_; scoped_ptr<webrtc::SessionDescriptionInterface> created_sessiondescription_; diff --git a/content/renderer/media/peer_connection_identity_service.h b/content/renderer/media/peer_connection_identity_service.h index b68cafa..a72cf47 100644 --- a/content/renderer/media/peer_connection_identity_service.h +++ b/content/renderer/media/peer_connection_identity_service.h @@ -38,7 +38,7 @@ class PeerConnectionIdentityService // The origin of the DTLS connection. GURL origin_; - talk_base::scoped_refptr<webrtc::DTLSIdentityRequestObserver> + rtc::scoped_refptr<webrtc::DTLSIdentityRequestObserver> pending_observer_; int pending_request_id_; diff --git a/content/renderer/media/peer_connection_tracker.cc b/content/renderer/media/peer_connection_tracker.cc index dc9be52..bead220 100644 --- a/content/renderer/media/peer_connection_tracker.cc +++ b/content/renderer/media/peer_connection_tracker.cc @@ -287,8 +287,8 @@ void PeerConnectionTracker::OnGetAllStats() { for (PeerConnectionIdMap::iterator it = peer_connection_id_map_.begin(); it != peer_connection_id_map_.end(); ++it) { - talk_base::scoped_refptr<InternalStatsObserver> observer( - new talk_base::RefCountedObject<InternalStatsObserver>(it->second)); + rtc::scoped_refptr<InternalStatsObserver> observer( + new rtc::RefCountedObject<InternalStatsObserver>(it->second)); it->first->GetStats( observer, diff --git a/content/renderer/media/rtc_data_channel_handler.cc b/content/renderer/media/rtc_data_channel_handler.cc index 1fe4de1..004cc46 100644 --- a/content/renderer/media/rtc_data_channel_handler.cc +++ b/content/renderer/media/rtc_data_channel_handler.cc @@ -103,14 +103,14 @@ unsigned long RtcDataChannelHandler::bufferedAmount() { bool RtcDataChannelHandler::sendStringData(const blink::WebString& data) { std::string utf8_buffer = base::UTF16ToUTF8(data); - talk_base::Buffer buffer(utf8_buffer.c_str(), utf8_buffer.length()); + rtc::Buffer buffer(utf8_buffer.c_str(), utf8_buffer.length()); webrtc::DataBuffer data_buffer(buffer, false); RecordMessageSent(data_buffer.size()); return channel_->Send(data_buffer); } bool RtcDataChannelHandler::sendRawData(const char* data, size_t length) { - talk_base::Buffer buffer(data, length); + rtc::Buffer buffer(data, length); webrtc::DataBuffer data_buffer(buffer, true); RecordMessageSent(data_buffer.size()); return channel_->Send(data_buffer); diff --git a/content/renderer/media/rtc_peer_connection_handler.cc b/content/renderer/media/rtc_peer_connection_handler.cc index c5767ae..0f92bb6e 100644 --- a/content/renderer/media/rtc_peer_connection_handler.cc +++ b/content/renderer/media/rtc_peer_connection_handler.cc @@ -292,8 +292,8 @@ class StatsResponse : public webrtc::StatsObserver { blink::WebString::fromUTF8(value)); } - talk_base::scoped_refptr<LocalRTCStatsRequest> request_; - talk_base::scoped_refptr<LocalRTCStatsResponse> response_; + rtc::scoped_refptr<LocalRTCStatsRequest> request_; + rtc::scoped_refptr<LocalRTCStatsResponse> response_; }; // Implementation of LocalRTCStatsRequest. @@ -315,7 +315,7 @@ blink::WebMediaStreamTrack LocalRTCStatsRequest::component() const { scoped_refptr<LocalRTCStatsResponse> LocalRTCStatsRequest::createResponse() { DCHECK(!response_); - response_ = new talk_base::RefCountedObject<LocalRTCStatsResponse>( + response_ = new rtc::RefCountedObject<LocalRTCStatsResponse>( impl_.createResponse()); return response_.get(); } @@ -472,7 +472,7 @@ bool RTCPeerConnectionHandler::initialize( peer_connection_tracker_->RegisterPeerConnection( this, config, constraints, frame_); - uma_observer_ = new talk_base::RefCountedObject<PeerConnectionUMAObserver>(); + uma_observer_ = new rtc::RefCountedObject<PeerConnectionUMAObserver>(); native_peer_connection_->RegisterUMAObserver(uma_observer_.get()); return true; } @@ -500,7 +500,7 @@ void RTCPeerConnectionHandler::createOffer( const blink::WebRTCSessionDescriptionRequest& request, const blink::WebMediaConstraints& options) { scoped_refptr<CreateSessionDescriptionRequest> description_request( - new talk_base::RefCountedObject<CreateSessionDescriptionRequest>( + new rtc::RefCountedObject<CreateSessionDescriptionRequest>( request, this, PeerConnectionTracker::ACTION_CREATE_OFFER)); RTCMediaConstraints constraints(options); native_peer_connection_->CreateOffer(description_request.get(), &constraints); @@ -513,7 +513,7 @@ void RTCPeerConnectionHandler::createOffer( const blink::WebRTCSessionDescriptionRequest& request, const blink::WebRTCOfferOptions& options) { scoped_refptr<CreateSessionDescriptionRequest> description_request( - new talk_base::RefCountedObject<CreateSessionDescriptionRequest>( + new rtc::RefCountedObject<CreateSessionDescriptionRequest>( request, this, PeerConnectionTracker::ACTION_CREATE_OFFER)); RTCMediaConstraints constraints; @@ -528,7 +528,7 @@ void RTCPeerConnectionHandler::createAnswer( const blink::WebRTCSessionDescriptionRequest& request, const blink::WebMediaConstraints& options) { scoped_refptr<CreateSessionDescriptionRequest> description_request( - new talk_base::RefCountedObject<CreateSessionDescriptionRequest>( + new rtc::RefCountedObject<CreateSessionDescriptionRequest>( request, this, PeerConnectionTracker::ACTION_CREATE_ANSWER)); RTCMediaConstraints constraints(options); native_peer_connection_->CreateAnswer(description_request.get(), @@ -558,7 +558,7 @@ void RTCPeerConnectionHandler::setLocalDescription( this, description, PeerConnectionTracker::SOURCE_LOCAL); scoped_refptr<SetSessionDescriptionRequest> set_request( - new talk_base::RefCountedObject<SetSessionDescriptionRequest>( + new rtc::RefCountedObject<SetSessionDescriptionRequest>( request, this, PeerConnectionTracker::ACTION_SET_LOCAL_DESCRIPTION)); native_peer_connection_->SetLocalDescription(set_request.get(), native_desc); } @@ -583,7 +583,7 @@ void RTCPeerConnectionHandler::setRemoteDescription( this, description, PeerConnectionTracker::SOURCE_REMOTE); scoped_refptr<SetSessionDescriptionRequest> set_request( - new talk_base::RefCountedObject<SetSessionDescriptionRequest>( + new rtc::RefCountedObject<SetSessionDescriptionRequest>( request, this, PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION)); native_peer_connection_->SetRemoteDescription(set_request.get(), native_desc); } @@ -728,13 +728,13 @@ void RTCPeerConnectionHandler::removeStream( void RTCPeerConnectionHandler::getStats( const blink::WebRTCStatsRequest& request) { scoped_refptr<LocalRTCStatsRequest> inner_request( - new talk_base::RefCountedObject<LocalRTCStatsRequest>(request)); + new rtc::RefCountedObject<LocalRTCStatsRequest>(request)); getStats(inner_request.get()); } void RTCPeerConnectionHandler::getStats(LocalRTCStatsRequest* request) { - talk_base::scoped_refptr<webrtc::StatsObserver> observer( - new talk_base::RefCountedObject<StatsResponse>(request)); + rtc::scoped_refptr<webrtc::StatsObserver> observer( + new rtc::RefCountedObject<StatsResponse>(request)); webrtc::MediaStreamTrackInterface* track = NULL; if (request->hasSelector()) { blink::WebMediaStreamSource::Type type = @@ -798,7 +798,7 @@ blink::WebRTCDataChannelHandler* RTCPeerConnectionHandler::createDataChannel( config.maxRetransmitTime = init.maxRetransmitTime; config.protocol = base::UTF16ToUTF8(init.protocol); - talk_base::scoped_refptr<webrtc::DataChannelInterface> webrtc_channel( + rtc::scoped_refptr<webrtc::DataChannelInterface> webrtc_channel( native_peer_connection_->CreateDataChannel(base::UTF16ToUTF8(label), &config)); if (!webrtc_channel) { @@ -826,7 +826,7 @@ blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender( } webrtc::AudioTrackInterface* audio_track = native_track->GetAudioAdapter(); - talk_base::scoped_refptr<webrtc::DtmfSenderInterface> sender( + rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender( native_peer_connection_->CreateDtmfSender(audio_track)); if (!sender) { DLOG(ERROR) << "Could not create native DTMF sender."; diff --git a/content/renderer/media/rtc_peer_connection_handler.h b/content/renderer/media/rtc_peer_connection_handler.h index 299585f..ec5f775 100644 --- a/content/renderer/media/rtc_peer_connection_handler.h +++ b/content/renderer/media/rtc_peer_connection_handler.h @@ -33,7 +33,7 @@ class WebRtcMediaStreamAdapter; // Mockable wrapper for blink::WebRTCStatsResponse class CONTENT_EXPORT LocalRTCStatsResponse - : public NON_EXPORTED_BASE(talk_base::RefCountInterface) { + : public NON_EXPORTED_BASE(rtc::RefCountInterface) { public: explicit LocalRTCStatsResponse(const blink::WebRTCStatsResponse& impl) : impl_(impl) { @@ -56,7 +56,7 @@ class CONTENT_EXPORT LocalRTCStatsResponse // Mockable wrapper for blink::WebRTCStatsRequest class CONTENT_EXPORT LocalRTCStatsRequest - : public NON_EXPORTED_BASE(talk_base::RefCountInterface) { + : public NON_EXPORTED_BASE(rtc::RefCountInterface) { public: explicit LocalRTCStatsRequest(blink::WebRTCStatsRequest impl); // Constructor for testing. @@ -72,7 +72,7 @@ class CONTENT_EXPORT LocalRTCStatsRequest private: blink::WebRTCStatsRequest impl_; - talk_base::scoped_refptr<LocalRTCStatsResponse> response_; + rtc::scoped_refptr<LocalRTCStatsResponse> response_; }; // RTCPeerConnectionHandler is a delegate for the RTC PeerConnection API diff --git a/content/renderer/media/rtc_peer_connection_handler_unittest.cc b/content/renderer/media/rtc_peer_connection_handler_unittest.cc index a71c434..2bd05e7 100644 --- a/content/renderer/media/rtc_peer_connection_handler_unittest.cc +++ b/content/renderer/media/rtc_peer_connection_handler_unittest.cc @@ -93,7 +93,7 @@ class MockRTCStatsRequest : public LocalRTCStatsRequest { } virtual scoped_refptr<LocalRTCStatsResponse> createResponse() OVERRIDE { DCHECK(!response_.get()); - response_ = new talk_base::RefCountedObject<MockRTCStatsResponse>(); + response_ = new rtc::RefCountedObject<MockRTCStatsResponse>(); return response_; } @@ -459,7 +459,7 @@ TEST_F(RTCPeerConnectionHandlerTest, addStreamWithStoppedAudioAndVideoTrack) { TEST_F(RTCPeerConnectionHandlerTest, GetStatsNoSelector) { scoped_refptr<MockRTCStatsRequest> request( - new talk_base::RefCountedObject<MockRTCStatsRequest>()); + new rtc::RefCountedObject<MockRTCStatsRequest>()); pc_handler_->getStats(request.get()); // Note that callback gets executed synchronously by mock. ASSERT_TRUE(request->result()); @@ -468,7 +468,7 @@ TEST_F(RTCPeerConnectionHandlerTest, GetStatsNoSelector) { TEST_F(RTCPeerConnectionHandlerTest, GetStatsAfterClose) { scoped_refptr<MockRTCStatsRequest> request( - new talk_base::RefCountedObject<MockRTCStatsRequest>()); + new rtc::RefCountedObject<MockRTCStatsRequest>()); pc_handler_->stop(); pc_handler_->getStats(request.get()); // Note that callback gets executed synchronously by mock. @@ -486,7 +486,7 @@ TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithLocalSelector) { ASSERT_LE(1ul, tracks.size()); scoped_refptr<MockRTCStatsRequest> request( - new talk_base::RefCountedObject<MockRTCStatsRequest>()); + new rtc::RefCountedObject<MockRTCStatsRequest>()); request->setSelector(tracks[0]); pc_handler_->getStats(request.get()); EXPECT_EQ(1, request->result()->report_count()); @@ -503,7 +503,7 @@ TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithRemoteSelector) { ASSERT_LE(1ul, tracks.size()); scoped_refptr<MockRTCStatsRequest> request( - new talk_base::RefCountedObject<MockRTCStatsRequest>()); + new rtc::RefCountedObject<MockRTCStatsRequest>()); request->setSelector(tracks[0]); pc_handler_->getStats(request.get()); EXPECT_EQ(1, request->result()->report_count()); @@ -522,7 +522,7 @@ TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithBadSelector) { mock_peer_connection_->SetGetStatsResult(false); scoped_refptr<MockRTCStatsRequest> request( - new talk_base::RefCountedObject<MockRTCStatsRequest>()); + new rtc::RefCountedObject<MockRTCStatsRequest>()); request->setSelector(component); pc_handler_->getStats(request.get()); EXPECT_EQ(0, request->result()->report_count()); diff --git a/content/renderer/media/webrtc/media_stream_remote_video_source.cc b/content/renderer/media/webrtc/media_stream_remote_video_source.cc index 74dbbb2..933bd5d 100644 --- a/content/renderer/media/webrtc/media_stream_remote_video_source.cc +++ b/content/renderer/media/webrtc/media_stream_remote_video_source.cc @@ -72,7 +72,7 @@ void MediaStreamRemoteVideoSource:: RemoteVideoSourceDelegate::RenderFrame( const cricket::VideoFrame* frame) { base::TimeDelta timestamp = base::TimeDelta::FromMicroseconds( - frame->GetElapsedTime() / talk_base::kNumNanosecsPerMicrosec); + frame->GetElapsedTime() / rtc::kNumNanosecsPerMicrosec); scoped_refptr<media::VideoFrame> video_frame; if (frame->GetNativeHandle() != NULL) { diff --git a/content/renderer/media/webrtc/media_stream_track_metrics.cc b/content/renderer/media/webrtc/media_stream_track_metrics.cc index 736cac3..24feb2f 100644 --- a/content/renderer/media/webrtc/media_stream_track_metrics.cc +++ b/content/renderer/media/webrtc/media_stream_track_metrics.cc @@ -42,9 +42,9 @@ class MediaStreamTrackMetricsObserver : public webrtc::ObserverInterface { virtual void OnChanged() OVERRIDE; template <class T> - IdSet GetTrackIds(const std::vector<talk_base::scoped_refptr<T> >& tracks) { + IdSet GetTrackIds(const std::vector<rtc::scoped_refptr<T> >& tracks) { IdSet track_ids; - typename std::vector<talk_base::scoped_refptr<T> >::const_iterator it = + typename std::vector<rtc::scoped_refptr<T> >::const_iterator it = tracks.begin(); for (; it != tracks.end(); ++it) { track_ids.insert((*it)->id()); @@ -72,7 +72,7 @@ class MediaStreamTrackMetricsObserver : public webrtc::ObserverInterface { IdSet video_track_ids_; MediaStreamTrackMetrics::StreamType stream_type_; - talk_base::scoped_refptr<MediaStreamInterface> stream_; + rtc::scoped_refptr<MediaStreamInterface> stream_; // Non-owning. MediaStreamTrackMetrics* owner_; diff --git a/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc b/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc index 343ab30..382fcba 100644 --- a/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc +++ b/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc @@ -80,7 +80,7 @@ class MediaStreamTrackMetricsTest : public testing::Test { public: virtual void SetUp() OVERRIDE { metrics_.reset(new MockMediaStreamTrackMetrics()); - stream_ = new talk_base::RefCountedObject<MockMediaStream>("stream"); + stream_ = new rtc::RefCountedObject<MockMediaStream>("stream"); } virtual void TearDown() OVERRIDE { @@ -89,11 +89,11 @@ class MediaStreamTrackMetricsTest : public testing::Test { } scoped_refptr<MockAudioTrackInterface> MakeAudioTrack(std::string id) { - return new talk_base::RefCountedObject<MockAudioTrackInterface>(id); + return new rtc::RefCountedObject<MockAudioTrackInterface>(id); } scoped_refptr<MockVideoTrackInterface> MakeVideoTrack(std::string id) { - return new talk_base::RefCountedObject<MockVideoTrackInterface>(id); + return new rtc::RefCountedObject<MockVideoTrackInterface>(id); } scoped_ptr<MockMediaStreamTrackMetrics> metrics_; diff --git a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc index a11272f..6d64c4c 100644 --- a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc +++ b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc @@ -14,8 +14,8 @@ #include "content/renderer/media/webrtc_local_audio_track.h" #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" -#include "third_party/libjingle/source/talk/base/scoped_ref_ptr.h" #include "third_party/libjingle/source/talk/media/base/videocapturer.h" +#include "third_party/webrtc/base/scoped_ref_ptr.h" using webrtc::AudioSourceInterface; using webrtc::AudioTrackInterface; @@ -90,13 +90,13 @@ VideoTrackVector MockMediaStream::GetVideoTracks() { return video_track_vector_; } -talk_base::scoped_refptr<AudioTrackInterface> MockMediaStream::FindAudioTrack( +rtc::scoped_refptr<AudioTrackInterface> MockMediaStream::FindAudioTrack( const std::string& track_id) { AudioTrackVector::iterator it = FindTrack(&audio_track_vector_, track_id); return it == audio_track_vector_.end() ? NULL : *it; } -talk_base::scoped_refptr<VideoTrackInterface> MockMediaStream::FindVideoTrack( +rtc::scoped_refptr<VideoTrackInterface> MockMediaStream::FindVideoTrack( const std::string& track_id) { VideoTrackVector::iterator it = FindTrack(&video_track_vector_, track_id); return it == video_track_vector_.end() ? NULL : *it; @@ -443,14 +443,14 @@ MockPeerConnectionDependencyFactory::CreatePeerConnection( const webrtc::MediaConstraintsInterface* constraints, blink::WebFrame* frame, webrtc::PeerConnectionObserver* observer) { - return new talk_base::RefCountedObject<MockPeerConnectionImpl>(this); + return new rtc::RefCountedObject<MockPeerConnectionImpl>(this); } scoped_refptr<webrtc::AudioSourceInterface> MockPeerConnectionDependencyFactory::CreateLocalAudioSource( const webrtc::MediaConstraintsInterface* constraints) { last_audio_source_ = - new talk_base::RefCountedObject<MockAudioSource>(constraints); + new rtc::RefCountedObject<MockAudioSource>(constraints); return last_audio_source_; } @@ -464,7 +464,7 @@ scoped_refptr<webrtc::VideoSourceInterface> MockPeerConnectionDependencyFactory::CreateVideoSource( cricket::VideoCapturer* capturer, const blink::WebMediaConstraints& constraints) { - last_video_source_ = new talk_base::RefCountedObject<MockVideoSource>(); + last_video_source_ = new rtc::RefCountedObject<MockVideoSource>(); last_video_source_->SetVideoCapturer(capturer); return last_video_source_; } @@ -478,7 +478,7 @@ MockPeerConnectionDependencyFactory::CreateWebAudioSource( scoped_refptr<webrtc::MediaStreamInterface> MockPeerConnectionDependencyFactory::CreateLocalMediaStream( const std::string& label) { - return new talk_base::RefCountedObject<MockMediaStream>(label); + return new rtc::RefCountedObject<MockMediaStream>(label); } scoped_refptr<webrtc::VideoTrackInterface> @@ -486,7 +486,7 @@ MockPeerConnectionDependencyFactory::CreateLocalVideoTrack( const std::string& id, webrtc::VideoSourceInterface* source) { scoped_refptr<webrtc::VideoTrackInterface> track( - new talk_base::RefCountedObject<MockWebRtcVideoTrack>( + new rtc::RefCountedObject<MockWebRtcVideoTrack>( id, source)); return track; } @@ -496,11 +496,11 @@ MockPeerConnectionDependencyFactory::CreateLocalVideoTrack( const std::string& id, cricket::VideoCapturer* capturer) { scoped_refptr<MockVideoSource> source = - new talk_base::RefCountedObject<MockVideoSource>(); + new rtc::RefCountedObject<MockVideoSource>(); source->SetVideoCapturer(capturer); return - new talk_base::RefCountedObject<MockWebRtcVideoTrack>(id, source.get()); + new rtc::RefCountedObject<MockWebRtcVideoTrack>(id, source.get()); } SessionDescriptionInterface* diff --git a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h index 2bdda74..871c183 100644 --- a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h +++ b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h @@ -145,9 +145,9 @@ class MockMediaStream : public webrtc::MediaStreamInterface { virtual std::string label() const OVERRIDE; virtual webrtc::AudioTrackVector GetAudioTracks() OVERRIDE; virtual webrtc::VideoTrackVector GetVideoTracks() OVERRIDE; - virtual talk_base::scoped_refptr<webrtc::AudioTrackInterface> + virtual rtc::scoped_refptr<webrtc::AudioTrackInterface> FindAudioTrack(const std::string& track_id) OVERRIDE; - virtual talk_base::scoped_refptr<webrtc::VideoTrackInterface> + virtual rtc::scoped_refptr<webrtc::VideoTrackInterface> FindVideoTrack(const std::string& track_id) OVERRIDE; virtual void RegisterObserver(webrtc::ObserverInterface* observer) OVERRIDE; virtual void UnregisterObserver(webrtc::ObserverInterface* observer) OVERRIDE; diff --git a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc index ed05bb9..9b4fcdb 100644 --- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc +++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc @@ -44,7 +44,7 @@ #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" #if defined(USE_OPENSSL) -#include "third_party/libjingle/source/talk/base/ssladapter.h" +#include "third_party/webrtc/base/ssladapter.h" #else #include "net/socket/nss_ssl_util.h" #endif @@ -115,8 +115,8 @@ class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface { public: P2PPortAllocatorFactory( P2PSocketDispatcher* socket_dispatcher, - talk_base::NetworkManager* network_manager, - talk_base::PacketSocketFactory* socket_factory, + rtc::NetworkManager* network_manager, + rtc::PacketSocketFactory* socket_factory, blink::WebFrame* web_frame) : socket_dispatcher_(socket_dispatcher), network_manager_(network_manager), @@ -163,8 +163,8 @@ class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface { scoped_refptr<P2PSocketDispatcher> socket_dispatcher_; // |network_manager_| and |socket_factory_| are a weak references, owned by // PeerConnectionDependencyFactory. - talk_base::NetworkManager* network_manager_; - talk_base::PacketSocketFactory* socket_factory_; + rtc::NetworkManager* network_manager_; + rtc::PacketSocketFactory* socket_factory_; // Raw ptr to the WebFrame that created the P2PPortAllocatorFactory. blink::WebFrame* web_frame_; }; @@ -309,7 +309,7 @@ void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() { // Init SSL, which will be needed by PeerConnection. #if defined(USE_OPENSSL) - if (!talk_base::InitializeSSL()) { + if (!rtc::InitializeSSL()) { LOG(ERROR) << "Failed on InitializeSSL."; NOTREACHED(); return; @@ -385,7 +385,7 @@ PeerConnectionDependencyFactory::CreatePeerConnection( return NULL; scoped_refptr<P2PPortAllocatorFactory> pa_factory = - new talk_base::RefCountedObject<P2PPortAllocatorFactory>( + new rtc::RefCountedObject<P2PPortAllocatorFactory>( p2p_socket_dispatcher_.get(), network_manager_, socket_factory_.get(), @@ -549,7 +549,7 @@ PeerConnectionDependencyFactory::GetWebRtcAudioDevice() { } void PeerConnectionDependencyFactory::InitializeWorkerThread( - talk_base::Thread** thread, + rtc::Thread** thread, base::WaitableEvent* event) { jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); diff --git a/content/renderer/media/webrtc/peer_connection_dependency_factory.h b/content/renderer/media/webrtc/peer_connection_dependency_factory.h index 37e0b00..a8c96ca 100644 --- a/content/renderer/media/webrtc/peer_connection_dependency_factory.h +++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.h @@ -22,7 +22,7 @@ namespace base { class WaitableEvent; } -namespace talk_base { +namespace rtc { class NetworkManager; class PacketSocketFactory; class Thread; @@ -179,7 +179,7 @@ class CONTENT_EXPORT PeerConnectionDependencyFactory // creating PeerConnection objects. void CreatePeerConnectionFactory(); - void InitializeWorkerThread(talk_base::Thread** thread, + void InitializeWorkerThread(rtc::Thread** thread, base::WaitableEvent* event); void CreateIpcNetworkManagerOnWorkerThread(base::WaitableEvent* event); @@ -206,8 +206,8 @@ class CONTENT_EXPORT PeerConnectionDependencyFactory // PeerConnection threads. signaling_thread_ is created from the // "current" chrome thread. - talk_base::Thread* signaling_thread_; - talk_base::Thread* worker_thread_; + rtc::Thread* signaling_thread_; + rtc::Thread* worker_thread_; base::Thread chrome_worker_thread_; DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory); diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc index d94edb8..96b6837 100644 --- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc +++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc @@ -18,8 +18,8 @@ scoped_refptr<WebRtcLocalAudioTrackAdapter> WebRtcLocalAudioTrackAdapter::Create( const std::string& label, webrtc::AudioSourceInterface* track_source) { - talk_base::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter = - new talk_base::RefCountedObject<WebRtcLocalAudioTrackAdapter>( + rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter = + new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>( label, track_source); return adapter; } @@ -98,7 +98,7 @@ bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) { return true; } -talk_base::scoped_refptr<webrtc::AudioProcessorInterface> +rtc::scoped_refptr<webrtc::AudioProcessorInterface> WebRtcLocalAudioTrackAdapter::GetAudioProcessor() { base::AutoLock auto_lock(lock_); return audio_processor_.get(); diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h index b35ad4a..630af24 100644 --- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h +++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h @@ -67,7 +67,7 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE; virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE; virtual bool GetSignalLevel(int* level) OVERRIDE; - virtual talk_base::scoped_refptr<webrtc::AudioProcessorInterface> + virtual rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() OVERRIDE; // cricket::AudioCapturer implementation. @@ -83,7 +83,7 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter // The source of the audio track which handles the audio constraints. // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. - talk_base::scoped_refptr<webrtc::AudioSourceInterface> track_source_; + rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; // The audio processsor that applies audio processing on the data of audio // track. diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc index e47beea..994f77b 100644 --- a/content/renderer/media/webrtc_audio_capturer.cc +++ b/content/renderer/media/webrtc_audio_capturer.cc @@ -228,7 +228,7 @@ WebRtcAudioCapturer::WebRtcAudioCapturer( MediaStreamAudioSource* audio_source) : constraints_(constraints), audio_processor_( - new talk_base::RefCountedObject<MediaStreamAudioProcessor>( + new rtc::RefCountedObject<MediaStreamAudioProcessor>( constraints, device_info.device.input.effects, audio_device)), running_(false), render_view_id_(render_view_id), diff --git a/content/renderer/media/webrtc_audio_renderer_unittest.cc b/content/renderer/media/webrtc_audio_renderer_unittest.cc index 3cf1b523..914dd52 100644 --- a/content/renderer/media/webrtc_audio_renderer_unittest.cc +++ b/content/renderer/media/webrtc_audio_renderer_unittest.cc @@ -87,7 +87,7 @@ class WebRtcAudioRendererTest : public testing::Test { message_loop_->message_loop_proxy())), factory_(new MockAudioDeviceFactory()), source_(new MockAudioRendererSource()), - stream_(new talk_base::RefCountedObject<MockMediaStream>("label")), + stream_(new rtc::RefCountedObject<MockMediaStream>("label")), renderer_(new WebRtcAudioRenderer(stream_, 1, 1, 1, 44100, 441)) { EXPECT_CALL(*factory_.get(), CreateOutputDevice(1)) .WillOnce(Return(mock_output_device_)); diff --git a/content/renderer/media/webrtc_logging.cc b/content/renderer/media/webrtc_logging.cc index 96868a6..17e5c57 100644 --- a/content/renderer/media/webrtc_logging.cc +++ b/content/renderer/media/webrtc_logging.cc @@ -6,7 +6,7 @@ #include "base/time/time.h" #include "content/public/renderer/webrtc_log_message_delegate.h" -#include "third_party/libjingle/overrides/talk/base/logging.h" +#include "third_party/webrtc/overrides/webrtc/base/logging.h" namespace content { @@ -22,7 +22,7 @@ void InitWebRtcLoggingDelegate(WebRtcLogMessageDelegate* delegate) { void InitWebRtcLogging() { // Log messages from Libjingle should not have timestamps. - talk_base::InitDiagnosticLoggingDelegateFunction(&WebRtcLogMessage); + rtc::InitDiagnosticLoggingDelegateFunction(&WebRtcLogMessage); } void WebRtcLogMessage(const std::string& message) { diff --git a/content/renderer/p2p/host_address_request.cc b/content/renderer/p2p/host_address_request.cc index 72e7c46..e21e0aa 100644 --- a/content/renderer/p2p/host_address_request.cc +++ b/content/renderer/p2p/host_address_request.cc @@ -29,7 +29,7 @@ P2PAsyncAddressResolver::~P2PAsyncAddressResolver() { DCHECK(!registered_); } -void P2PAsyncAddressResolver::Start(const talk_base::SocketAddress& host_name, +void P2PAsyncAddressResolver::Start(const rtc::SocketAddress& host_name, const DoneCallback& done_callback) { DCHECK(delegate_message_loop_->BelongsToCurrentThread()); DCHECK_EQ(STATE_CREATED, state_); @@ -51,7 +51,7 @@ void P2PAsyncAddressResolver::Cancel() { } void P2PAsyncAddressResolver::DoSendRequest( - const talk_base::SocketAddress& host_name, + const rtc::SocketAddress& host_name, const DoneCallback& done_callback) { DCHECK(ipc_message_loop_->BelongsToCurrentThread()); diff --git a/content/renderer/p2p/host_address_request.h b/content/renderer/p2p/host_address_request.h index 7c90b80..ab45486 100644 --- a/content/renderer/p2p/host_address_request.h +++ b/content/renderer/p2p/host_address_request.h @@ -11,7 +11,7 @@ #include "base/memory/ref_counted.h" #include "content/common/content_export.h" #include "net/base/net_util.h" -#include "third_party/libjingle/source/talk/base/asyncresolverinterface.h" +#include "third_party/webrtc/base/asyncresolverinterface.h" namespace base { class MessageLoop; @@ -31,7 +31,7 @@ class P2PAsyncAddressResolver P2PAsyncAddressResolver(P2PSocketDispatcher* dispatcher); // Start address resolve process. - void Start(const talk_base::SocketAddress& addr, + void Start(const rtc::SocketAddress& addr, const DoneCallback& done_callback); // Clients must unregister before exiting for cleanup. void Cancel(); @@ -49,7 +49,7 @@ class P2PAsyncAddressResolver virtual ~P2PAsyncAddressResolver(); - void DoSendRequest(const talk_base::SocketAddress& host_name, + void DoSendRequest(const rtc::SocketAddress& host_name, const DoneCallback& done_callback); void DoUnregister(); void OnResponse(const net::IPAddressList& address); @@ -65,7 +65,7 @@ class P2PAsyncAddressResolver // Accessed on the IPC thread only. int32 request_id_; bool registered_; - std::vector<talk_base::IPAddress> addresses_; + std::vector<rtc::IPAddress> addresses_; DoneCallback done_callback_; DISALLOW_COPY_AND_ASSIGN(P2PAsyncAddressResolver); diff --git a/content/renderer/p2p/ipc_network_manager.cc b/content/renderer/p2p/ipc_network_manager.cc index d306787..8995339 100644 --- a/content/renderer/p2p/ipc_network_manager.cc +++ b/content/renderer/p2p/ipc_network_manager.cc @@ -15,21 +15,21 @@ namespace content { namespace { -talk_base::AdapterType ConvertConnectionTypeToAdapterType( +rtc::AdapterType ConvertConnectionTypeToAdapterType( net::NetworkChangeNotifier::ConnectionType type) { switch (type) { case net::NetworkChangeNotifier::CONNECTION_UNKNOWN: - return talk_base::ADAPTER_TYPE_UNKNOWN; + return rtc::ADAPTER_TYPE_UNKNOWN; case net::NetworkChangeNotifier::CONNECTION_ETHERNET: - return talk_base::ADAPTER_TYPE_ETHERNET; + return rtc::ADAPTER_TYPE_ETHERNET; case net::NetworkChangeNotifier::CONNECTION_WIFI: - return talk_base::ADAPTER_TYPE_WIFI; + return rtc::ADAPTER_TYPE_WIFI; case net::NetworkChangeNotifier::CONNECTION_2G: case net::NetworkChangeNotifier::CONNECTION_3G: case net::NetworkChangeNotifier::CONNECTION_4G: - return talk_base::ADAPTER_TYPE_CELLULAR; + return rtc::ADAPTER_TYPE_CELLULAR; default: - return talk_base::ADAPTER_TYPE_UNKNOWN; + return rtc::ADAPTER_TYPE_UNKNOWN; } } @@ -73,9 +73,9 @@ void IpcNetworkManager::OnNetworkListChanged( // Note: 32 and 64 are the arbitrary(kind of) prefix length used to // differentiate IPv4 and IPv6 addresses. - // talk_base::Network uses these prefix_length to compare network + // rtc::Network uses these prefix_length to compare network // interfaces discovered. - std::vector<talk_base::Network*> networks; + std::vector<rtc::Network*> networks; int ipv4_interfaces = 0; int ipv6_interfaces = 0; for (net::NetworkInterfaceList::const_iterator it = list.begin(); @@ -83,19 +83,19 @@ void IpcNetworkManager::OnNetworkListChanged( if (it->address.size() == net::kIPv4AddressSize) { uint32 address; memcpy(&address, &it->address[0], sizeof(uint32)); - address = talk_base::NetworkToHost32(address); - talk_base::Network* network = new talk_base::Network( - it->name, it->name, talk_base::IPAddress(address), 32, + address = rtc::NetworkToHost32(address); + rtc::Network* network = new rtc::Network( + it->name, it->name, rtc::IPAddress(address), 32, ConvertConnectionTypeToAdapterType(it->type)); - network->AddIP(talk_base::IPAddress(address)); + network->AddIP(rtc::IPAddress(address)); networks.push_back(network); ++ipv4_interfaces; } else if (it->address.size() == net::kIPv6AddressSize) { in6_addr address; memcpy(&address, &it->address[0], sizeof(in6_addr)); - talk_base::IPAddress ip6_addr(address); - if (!talk_base::IPIsPrivate(ip6_addr)) { - talk_base::Network* network = new talk_base::Network( + rtc::IPAddress ip6_addr(address); + if (!rtc::IPIsPrivate(ip6_addr)) { + rtc::Network* network = new rtc::Network( it->name, it->name, ip6_addr, 64, ConvertConnectionTypeToAdapterType(it->type)); network->AddIP(ip6_addr); @@ -115,16 +115,16 @@ void IpcNetworkManager::OnNetworkListChanged( if (CommandLine::ForCurrentProcess()->HasSwitch( switches::kAllowLoopbackInPeerConnection)) { std::string name_v4("loopback_ipv4"); - talk_base::IPAddress ip_address_v4(INADDR_LOOPBACK); - talk_base::Network* network_v4 = new talk_base::Network( - name_v4, name_v4, ip_address_v4, 32, talk_base::ADAPTER_TYPE_UNKNOWN); + rtc::IPAddress ip_address_v4(INADDR_LOOPBACK); + rtc::Network* network_v4 = new rtc::Network( + name_v4, name_v4, ip_address_v4, 32, rtc::ADAPTER_TYPE_UNKNOWN); network_v4->AddIP(ip_address_v4); networks.push_back(network_v4); std::string name_v6("loopback_ipv6"); - talk_base::IPAddress ip_address_v6(in6addr_loopback); - talk_base::Network* network_v6 = new talk_base::Network( - name_v6, name_v6, ip_address_v6, 64, talk_base::ADAPTER_TYPE_UNKNOWN); + rtc::IPAddress ip_address_v6(in6addr_loopback); + rtc::Network* network_v6 = new rtc::Network( + name_v6, name_v6, ip_address_v6, 64, rtc::ADAPTER_TYPE_UNKNOWN); network_v6->AddIP(ip_address_v6); networks.push_back(network_v6); } diff --git a/content/renderer/p2p/ipc_network_manager.h b/content/renderer/p2p/ipc_network_manager.h index 06931dc..4ed3c48 100644 --- a/content/renderer/p2p/ipc_network_manager.h +++ b/content/renderer/p2p/ipc_network_manager.h @@ -12,13 +12,13 @@ #include "content/common/content_export.h" #include "content/renderer/p2p/network_list_observer.h" #include "content/renderer/p2p/socket_dispatcher.h" -#include "third_party/libjingle/source/talk/base/network.h" +#include "third_party/webrtc/base/network.h" namespace content { // IpcNetworkManager is a NetworkManager for libjingle that gets a // list of network interfaces from the browser. -class IpcNetworkManager : public talk_base::NetworkManagerBase, +class IpcNetworkManager : public rtc::NetworkManagerBase, public NetworkListObserver { public: // Constructor doesn't take ownership of the |socket_dispatcher|. diff --git a/content/renderer/p2p/ipc_socket_factory.cc b/content/renderer/p2p/ipc_socket_factory.cc index f9dfbe0..62428ad 100644 --- a/content/renderer/p2p/ipc_socket_factory.cc +++ b/content/renderer/p2p/ipc_socket_factory.cc @@ -19,7 +19,7 @@ #include "content/renderer/p2p/socket_client_impl.h" #include "content/renderer/p2p/socket_dispatcher.h" #include "jingle/glue/utils.h" -#include "third_party/libjingle/source/talk/base/asyncpacketsocket.h" +#include "third_party/webrtc/base/asyncpacketsocket.h" namespace content { @@ -36,22 +36,22 @@ bool IsTcpClientSocket(P2PSocketType type) { (type == P2P_SOCKET_STUN_TLS_CLIENT); } -bool JingleSocketOptionToP2PSocketOption(talk_base::Socket::Option option, +bool JingleSocketOptionToP2PSocketOption(rtc::Socket::Option option, P2PSocketOption* ipc_option) { switch (option) { - case talk_base::Socket::OPT_RCVBUF: + case rtc::Socket::OPT_RCVBUF: *ipc_option = P2P_SOCKET_OPT_RCVBUF; break; - case talk_base::Socket::OPT_SNDBUF: + case rtc::Socket::OPT_SNDBUF: *ipc_option = P2P_SOCKET_OPT_SNDBUF; break; - case talk_base::Socket::OPT_DSCP: + case rtc::Socket::OPT_DSCP: *ipc_option = P2P_SOCKET_OPT_DSCP; break; - case talk_base::Socket::OPT_DONTFRAGMENT: - case talk_base::Socket::OPT_NODELAY: - case talk_base::Socket::OPT_IPV6_V6ONLY: - case talk_base::Socket::OPT_RTP_SENDTIME_EXTN_ID: + case rtc::Socket::OPT_DONTFRAGMENT: + case rtc::Socket::OPT_NODELAY: + case rtc::Socket::OPT_IPV6_V6ONLY: + case rtc::Socket::OPT_RTP_SENDTIME_EXTN_ID: return false; // Not supported by the chrome sockets. default: NOTREACHED(); @@ -63,10 +63,10 @@ bool JingleSocketOptionToP2PSocketOption(talk_base::Socket::Option option, // TODO(miu): This needs tuning. http://crbug.com/237960 const size_t kMaximumInFlightBytes = 64 * 1024; // 64 KB -// IpcPacketSocket implements talk_base::AsyncPacketSocket interface +// IpcPacketSocket implements rtc::AsyncPacketSocket interface // using P2PSocketClient that works over IPC-channel. It must be used // on the thread it was created. -class IpcPacketSocket : public talk_base::AsyncPacketSocket, +class IpcPacketSocket : public rtc::AsyncPacketSocket, public P2PSocketClientDelegate { public: IpcPacketSocket(); @@ -74,21 +74,21 @@ class IpcPacketSocket : public talk_base::AsyncPacketSocket, // Always takes ownership of client even if initialization fails. bool Init(P2PSocketType type, P2PSocketClientImpl* client, - const talk_base::SocketAddress& local_address, - const talk_base::SocketAddress& remote_address); + const rtc::SocketAddress& local_address, + const rtc::SocketAddress& remote_address); - // talk_base::AsyncPacketSocket interface. - virtual talk_base::SocketAddress GetLocalAddress() const OVERRIDE; - virtual talk_base::SocketAddress GetRemoteAddress() const OVERRIDE; + // rtc::AsyncPacketSocket interface. + virtual rtc::SocketAddress GetLocalAddress() const OVERRIDE; + virtual rtc::SocketAddress GetRemoteAddress() const OVERRIDE; virtual int Send(const void *pv, size_t cb, - const talk_base::PacketOptions& options) OVERRIDE; + const rtc::PacketOptions& options) OVERRIDE; virtual int SendTo(const void *pv, size_t cb, - const talk_base::SocketAddress& addr, - const talk_base::PacketOptions& options) OVERRIDE; + const rtc::SocketAddress& addr, + const rtc::PacketOptions& options) OVERRIDE; virtual int Close() OVERRIDE; virtual State GetState() const OVERRIDE; - virtual int GetOption(talk_base::Socket::Option option, int* value) OVERRIDE; - virtual int SetOption(talk_base::Socket::Option option, int value) OVERRIDE; + virtual int GetOption(rtc::Socket::Option option, int* value) OVERRIDE; + virtual int SetOption(rtc::Socket::Option option, int value) OVERRIDE; virtual int GetError() const OVERRIDE; virtual void SetError(int error) OVERRIDE; @@ -119,8 +119,8 @@ class IpcPacketSocket : public talk_base::AsyncPacketSocket, void TraceSendThrottlingState() const; void InitAcceptedTcp(P2PSocketClient* client, - const talk_base::SocketAddress& local_address, - const talk_base::SocketAddress& remote_address); + const rtc::SocketAddress& local_address, + const rtc::SocketAddress& remote_address); int DoSetOption(P2PSocketOption option, int value); @@ -135,10 +135,10 @@ class IpcPacketSocket : public talk_base::AsyncPacketSocket, // Local address is allocated by the browser process, and the // renderer side doesn't know the address until it receives OnOpen() // event from the browser. - talk_base::SocketAddress local_address_; + rtc::SocketAddress local_address_; // Remote address for client TCP connections. - talk_base::SocketAddress remote_address_; + rtc::SocketAddress remote_address_; // Current state of the object. InternalState state_; @@ -169,15 +169,15 @@ class IpcPacketSocket : public talk_base::AsyncPacketSocket, // of MT sig slots clients must call disconnect. This class is to make sure // we destruct from the same thread on which is created. class AsyncAddressResolverImpl : public base::NonThreadSafe, - public talk_base::AsyncResolverInterface { + public rtc::AsyncResolverInterface { public: AsyncAddressResolverImpl(P2PSocketDispatcher* dispatcher); virtual ~AsyncAddressResolverImpl(); - // talk_base::AsyncResolverInterface interface. - virtual void Start(const talk_base::SocketAddress& addr) OVERRIDE; + // rtc::AsyncResolverInterface interface. + virtual void Start(const rtc::SocketAddress& addr) OVERRIDE; virtual bool GetResolvedAddress( - int family, talk_base::SocketAddress* addr) const OVERRIDE; + int family, rtc::SocketAddress* addr) const OVERRIDE; virtual int GetError() const OVERRIDE; virtual void Destroy(bool wait) OVERRIDE; @@ -186,7 +186,7 @@ class AsyncAddressResolverImpl : public base::NonThreadSafe, scoped_refptr<P2PAsyncAddressResolver> resolver_; int port_; // Port number in |addr| from Start() method. - std::vector<talk_base::IPAddress> addresses_; // Resolved addresses. + std::vector<rtc::IPAddress> addresses_; // Resolved addresses. }; IpcPacketSocket::IpcPacketSocket() @@ -217,8 +217,8 @@ void IpcPacketSocket::TraceSendThrottlingState() const { bool IpcPacketSocket::Init(P2PSocketType type, P2PSocketClientImpl* client, - const talk_base::SocketAddress& local_address, - const talk_base::SocketAddress& remote_address) { + const rtc::SocketAddress& local_address, + const rtc::SocketAddress& remote_address) { DCHECK_EQ(base::MessageLoop::current(), message_loop_); DCHECK_EQ(state_, IS_UNINITIALIZED); @@ -255,8 +255,8 @@ bool IpcPacketSocket::Init(P2PSocketType type, void IpcPacketSocket::InitAcceptedTcp( P2PSocketClient* client, - const talk_base::SocketAddress& local_address, - const talk_base::SocketAddress& remote_address) { + const rtc::SocketAddress& local_address, + const rtc::SocketAddress& remote_address) { DCHECK_EQ(base::MessageLoop::current(), message_loop_); DCHECK_EQ(state_, IS_UNINITIALIZED); @@ -268,26 +268,26 @@ void IpcPacketSocket::InitAcceptedTcp( client_->SetDelegate(this); } -// talk_base::AsyncPacketSocket interface. -talk_base::SocketAddress IpcPacketSocket::GetLocalAddress() const { +// rtc::AsyncPacketSocket interface. +rtc::SocketAddress IpcPacketSocket::GetLocalAddress() const { DCHECK_EQ(base::MessageLoop::current(), message_loop_); return local_address_; } -talk_base::SocketAddress IpcPacketSocket::GetRemoteAddress() const { +rtc::SocketAddress IpcPacketSocket::GetRemoteAddress() const { DCHECK_EQ(base::MessageLoop::current(), message_loop_); return remote_address_; } int IpcPacketSocket::Send(const void *data, size_t data_size, - const talk_base::PacketOptions& options) { + const rtc::PacketOptions& options) { DCHECK_EQ(base::MessageLoop::current(), message_loop_); return SendTo(data, data_size, remote_address_, options); } int IpcPacketSocket::SendTo(const void *data, size_t data_size, - const talk_base::SocketAddress& address, - const talk_base::PacketOptions& options) { + const rtc::SocketAddress& address, + const rtc::PacketOptions& options) { DCHECK_EQ(base::MessageLoop::current(), message_loop_); switch (state_) { @@ -355,7 +355,7 @@ int IpcPacketSocket::Close() { return 0; } -talk_base::AsyncPacketSocket::State IpcPacketSocket::GetState() const { +rtc::AsyncPacketSocket::State IpcPacketSocket::GetState() const { DCHECK_EQ(base::MessageLoop::current(), message_loop_); switch (state_) { @@ -382,7 +382,7 @@ talk_base::AsyncPacketSocket::State IpcPacketSocket::GetState() const { return STATE_CLOSED; } -int IpcPacketSocket::GetOption(talk_base::Socket::Option option, int* value) { +int IpcPacketSocket::GetOption(rtc::Socket::Option option, int* value) { P2PSocketOption p2p_socket_option = P2P_SOCKET_OPT_MAX; if (!JingleSocketOptionToP2PSocketOption(option, &p2p_socket_option)) { // unsupported option. @@ -393,7 +393,7 @@ int IpcPacketSocket::GetOption(talk_base::Socket::Option option, int* value) { return 0; } -int IpcPacketSocket::SetOption(talk_base::Socket::Option option, int value) { +int IpcPacketSocket::SetOption(rtc::Socket::Option option, int value) { DCHECK_EQ(base::MessageLoop::current(), message_loop_); P2PSocketOption p2p_socket_option = P2P_SOCKET_OPT_MAX; @@ -456,7 +456,7 @@ void IpcPacketSocket::OnOpen(const net::IPEndPoint& local_address, // in the callback. This address will be used while sending the packets // over the network. if (remote_address_.IsUnresolvedIP()) { - talk_base::SocketAddress jingle_socket_address; + rtc::SocketAddress jingle_socket_address; if (!jingle_glue::IPEndPointToSocketAddress( remote_address, &jingle_socket_address)) { NOTREACHED(); @@ -474,7 +474,7 @@ void IpcPacketSocket::OnIncomingTcpConnection( scoped_ptr<IpcPacketSocket> socket(new IpcPacketSocket()); - talk_base::SocketAddress remote_address; + rtc::SocketAddress remote_address; if (!jingle_glue::IPEndPointToSocketAddress(address, &remote_address)) { // Always expect correct IPv4 address to be allocated. NOTREACHED(); @@ -519,7 +519,7 @@ void IpcPacketSocket::OnDataReceived(const net::IPEndPoint& address, const base::TimeTicks& timestamp) { DCHECK_EQ(base::MessageLoop::current(), message_loop_); - talk_base::SocketAddress address_lj; + rtc::SocketAddress address_lj; if (!jingle_glue::IPEndPointToSocketAddress(address, &address_lj)) { // We should always be able to convert address here because we // don't expect IPv6 address on IPv4 connections. @@ -527,7 +527,7 @@ void IpcPacketSocket::OnDataReceived(const net::IPEndPoint& address, return; } - talk_base::PacketTime packet_time(timestamp.ToInternalValue(), 0); + rtc::PacketTime packet_time(timestamp.ToInternalValue(), 0); SignalReadPacket(this, &data[0], data.size(), address_lj, packet_time); } @@ -540,7 +540,7 @@ AsyncAddressResolverImpl::AsyncAddressResolverImpl( AsyncAddressResolverImpl::~AsyncAddressResolverImpl() { } -void AsyncAddressResolverImpl::Start(const talk_base::SocketAddress& addr) { +void AsyncAddressResolverImpl::Start(const rtc::SocketAddress& addr) { DCHECK(CalledOnValidThread()); // Copy port number from |addr|. |port_| must be copied // when resolved address is returned in GetResolvedAddress. @@ -552,7 +552,7 @@ void AsyncAddressResolverImpl::Start(const talk_base::SocketAddress& addr) { } bool AsyncAddressResolverImpl::GetResolvedAddress( - int family, talk_base::SocketAddress* addr) const { + int family, rtc::SocketAddress* addr) const { DCHECK(CalledOnValidThread()); if (addresses_.empty()) @@ -585,7 +585,7 @@ void AsyncAddressResolverImpl::OnAddressResolved( const net::IPAddressList& addresses) { DCHECK(CalledOnValidThread()); for (size_t i = 0; i < addresses.size(); ++i) { - talk_base::SocketAddress socket_address; + rtc::SocketAddress socket_address; if (!jingle_glue::IPEndPointToSocketAddress( net::IPEndPoint(addresses[i], 0), &socket_address)) { NOTREACHED(); @@ -605,54 +605,54 @@ IpcPacketSocketFactory::IpcPacketSocketFactory( IpcPacketSocketFactory::~IpcPacketSocketFactory() { } -talk_base::AsyncPacketSocket* IpcPacketSocketFactory::CreateUdpSocket( - const talk_base::SocketAddress& local_address, int min_port, int max_port) { - talk_base::SocketAddress crome_address; +rtc::AsyncPacketSocket* IpcPacketSocketFactory::CreateUdpSocket( + const rtc::SocketAddress& local_address, int min_port, int max_port) { + rtc::SocketAddress crome_address; P2PSocketClientImpl* socket_client = new P2PSocketClientImpl(socket_dispatcher_); scoped_ptr<IpcPacketSocket> socket(new IpcPacketSocket()); // TODO(sergeyu): Respect local_address and port limits here (need // to pass them over IPC channel to the browser). if (!socket->Init(P2P_SOCKET_UDP, socket_client, - local_address, talk_base::SocketAddress())) { + local_address, rtc::SocketAddress())) { return NULL; } return socket.release(); } -talk_base::AsyncPacketSocket* IpcPacketSocketFactory::CreateServerTcpSocket( - const talk_base::SocketAddress& local_address, int min_port, int max_port, +rtc::AsyncPacketSocket* IpcPacketSocketFactory::CreateServerTcpSocket( + const rtc::SocketAddress& local_address, int min_port, int max_port, int opts) { // TODO(sergeyu): Implement SSL support. - if (opts & talk_base::PacketSocketFactory::OPT_SSLTCP) + if (opts & rtc::PacketSocketFactory::OPT_SSLTCP) return NULL; - P2PSocketType type = (opts & talk_base::PacketSocketFactory::OPT_STUN) ? + P2PSocketType type = (opts & rtc::PacketSocketFactory::OPT_STUN) ? P2P_SOCKET_STUN_TCP_SERVER : P2P_SOCKET_TCP_SERVER; P2PSocketClientImpl* socket_client = new P2PSocketClientImpl(socket_dispatcher_); scoped_ptr<IpcPacketSocket> socket(new IpcPacketSocket()); if (!socket->Init(type, socket_client, local_address, - talk_base::SocketAddress())) { + rtc::SocketAddress())) { return NULL; } return socket.release(); } -talk_base::AsyncPacketSocket* IpcPacketSocketFactory::CreateClientTcpSocket( - const talk_base::SocketAddress& local_address, - const talk_base::SocketAddress& remote_address, - const talk_base::ProxyInfo& proxy_info, +rtc::AsyncPacketSocket* IpcPacketSocketFactory::CreateClientTcpSocket( + const rtc::SocketAddress& local_address, + const rtc::SocketAddress& remote_address, + const rtc::ProxyInfo& proxy_info, const std::string& user_agent, int opts) { P2PSocketType type; - if (opts & talk_base::PacketSocketFactory::OPT_SSLTCP) { - type = (opts & talk_base::PacketSocketFactory::OPT_STUN) ? + if (opts & rtc::PacketSocketFactory::OPT_SSLTCP) { + type = (opts & rtc::PacketSocketFactory::OPT_STUN) ? P2P_SOCKET_STUN_SSLTCP_CLIENT : P2P_SOCKET_SSLTCP_CLIENT; - } else if (opts & talk_base::PacketSocketFactory::OPT_TLS) { - type = (opts & talk_base::PacketSocketFactory::OPT_STUN) ? + } else if (opts & rtc::PacketSocketFactory::OPT_TLS) { + type = (opts & rtc::PacketSocketFactory::OPT_STUN) ? P2P_SOCKET_STUN_TLS_CLIENT : P2P_SOCKET_TLS_CLIENT; } else { - type = (opts & talk_base::PacketSocketFactory::OPT_STUN) ? + type = (opts & rtc::PacketSocketFactory::OPT_STUN) ? P2P_SOCKET_STUN_TCP_CLIENT : P2P_SOCKET_TCP_CLIENT; } P2PSocketClientImpl* socket_client = @@ -663,7 +663,7 @@ talk_base::AsyncPacketSocket* IpcPacketSocketFactory::CreateClientTcpSocket( return socket.release(); } -talk_base::AsyncResolverInterface* +rtc::AsyncResolverInterface* IpcPacketSocketFactory::CreateAsyncResolver() { scoped_ptr<AsyncAddressResolverImpl> resolver( new AsyncAddressResolverImpl(socket_dispatcher_)); diff --git a/content/renderer/p2p/ipc_socket_factory.h b/content/renderer/p2p/ipc_socket_factory.h index cba98b3..53e1edc 100644 --- a/content/renderer/p2p/ipc_socket_factory.h +++ b/content/renderer/p2p/ipc_socket_factory.h @@ -14,33 +14,33 @@ namespace content { class P2PSocketDispatcher; -// IpcPacketSocketFactory implements talk_base::PacketSocketFactory +// IpcPacketSocketFactory implements rtc::PacketSocketFactory // interface for libjingle using IPC-based P2P sockets. The class must // be used on a thread that is a libjingle thread (implements -// talk_base::Thread) and also has associated base::MessageLoop. Each +// rtc::Thread) and also has associated base::MessageLoop. Each // socket created by the factory must be used on the thread it was // created on. -class IpcPacketSocketFactory : public talk_base::PacketSocketFactory { +class IpcPacketSocketFactory : public rtc::PacketSocketFactory { public: CONTENT_EXPORT explicit IpcPacketSocketFactory( P2PSocketDispatcher* socket_dispatcher); virtual ~IpcPacketSocketFactory(); - virtual talk_base::AsyncPacketSocket* CreateUdpSocket( - const talk_base::SocketAddress& local_address, + virtual rtc::AsyncPacketSocket* CreateUdpSocket( + const rtc::SocketAddress& local_address, int min_port, int max_port) OVERRIDE; - virtual talk_base::AsyncPacketSocket* CreateServerTcpSocket( - const talk_base::SocketAddress& local_address, + virtual rtc::AsyncPacketSocket* CreateServerTcpSocket( + const rtc::SocketAddress& local_address, int min_port, int max_port, int opts) OVERRIDE; - virtual talk_base::AsyncPacketSocket* CreateClientTcpSocket( - const talk_base::SocketAddress& local_address, - const talk_base::SocketAddress& remote_address, - const talk_base::ProxyInfo& proxy_info, + virtual rtc::AsyncPacketSocket* CreateClientTcpSocket( + const rtc::SocketAddress& local_address, + const rtc::SocketAddress& remote_address, + const rtc::ProxyInfo& proxy_info, const std::string& user_agent, int opts) OVERRIDE; - virtual talk_base::AsyncResolverInterface* CreateAsyncResolver() OVERRIDE; + virtual rtc::AsyncResolverInterface* CreateAsyncResolver() OVERRIDE; private: P2PSocketDispatcher* socket_dispatcher_; diff --git a/content/renderer/p2p/port_allocator.cc b/content/renderer/p2p/port_allocator.cc index 61cfefd..be6a72e 100644 --- a/content/renderer/p2p/port_allocator.cc +++ b/content/renderer/p2p/port_allocator.cc @@ -69,8 +69,8 @@ P2PPortAllocator::Config::RelayServerConfig::~RelayServerConfig() { P2PPortAllocator::P2PPortAllocator( blink::WebFrame* web_frame, P2PSocketDispatcher* socket_dispatcher, - talk_base::NetworkManager* network_manager, - talk_base::PacketSocketFactory* socket_factory, + rtc::NetworkManager* network_manager, + rtc::PacketSocketFactory* socket_factory, const Config& config) : cricket::BasicPortAllocator(network_manager, socket_factory), web_frame_(web_frame), @@ -259,7 +259,7 @@ void P2PPortAllocatorSession::ParseRelayResponse() { void P2PPortAllocatorSession::AddConfig() { const P2PPortAllocator::Config& config = allocator_->config_; cricket::PortConfiguration* port_config = new cricket::PortConfiguration( - talk_base::SocketAddress(config.stun_server, config.stun_server_port), + rtc::SocketAddress(config.stun_server, config.stun_server_port), std::string(), std::string()); for (size_t i = 0; i < config.relays.size(); ++i) { @@ -278,7 +278,7 @@ void P2PPortAllocatorSession::AddConfig() { } relay_server.ports.push_back(cricket::ProtocolAddress( - talk_base::SocketAddress(config.relays[i].server_address, + rtc::SocketAddress(config.relays[i].server_address, config.relays[i].port), protocol, config.relays[i].secure)); diff --git a/content/renderer/p2p/port_allocator.h b/content/renderer/p2p/port_allocator.h index 8ff20a5..258d785 100644 --- a/content/renderer/p2p/port_allocator.h +++ b/content/renderer/p2p/port_allocator.h @@ -56,8 +56,8 @@ class P2PPortAllocator : public cricket::BasicPortAllocator { P2PPortAllocator(blink::WebFrame* web_frame, P2PSocketDispatcher* socket_dispatcher, - talk_base::NetworkManager* network_manager, - talk_base::PacketSocketFactory* socket_factory, + rtc::NetworkManager* network_manager, + rtc::PacketSocketFactory* socket_factory, const Config& config); virtual ~P2PPortAllocator(); @@ -115,7 +115,7 @@ class P2PPortAllocatorSession : public cricket::BasicPortAllocatorSession, scoped_ptr<blink::WebURLLoader> relay_session_request_; int relay_session_attempts_; std::string relay_session_response_; - talk_base::SocketAddress relay_ip_; + rtc::SocketAddress relay_ip_; int relay_udp_port_; int relay_tcp_port_; int relay_ssltcp_port_; diff --git a/content/renderer/p2p/socket_client.h b/content/renderer/p2p/socket_client.h index 5d1ecb5..d92407c 100644 --- a/content/renderer/p2p/socket_client.h +++ b/content/renderer/p2p/socket_client.h @@ -11,7 +11,7 @@ #include "content/common/p2p_socket_type.h" #include "net/base/ip_endpoint.h" -namespace talk_base { +namespace rtc { struct PacketOptions; }; @@ -44,7 +44,7 @@ class P2PSocketClient : public base::RefCountedThreadSafe<P2PSocketClient> { // |dscp|. virtual void SendWithDscp(const net::IPEndPoint& address, const std::vector<char>& data, - const talk_base::PacketOptions& options) = 0; + const rtc::PacketOptions& options) = 0; virtual void SetOption(P2PSocketOption option, int value) = 0; diff --git a/content/renderer/p2p/socket_client_impl.cc b/content/renderer/p2p/socket_client_impl.cc index cfa898f..1425151 100644 --- a/content/renderer/p2p/socket_client_impl.cc +++ b/content/renderer/p2p/socket_client_impl.cc @@ -72,7 +72,7 @@ void P2PSocketClientImpl::DoInit(P2PSocketType type, void P2PSocketClientImpl::SendWithDscp( const net::IPEndPoint& address, const std::vector<char>& data, - const talk_base::PacketOptions& options) { + const rtc::PacketOptions& options) { if (!ipc_message_loop_->BelongsToCurrentThread()) { ipc_message_loop_->PostTask( FROM_HERE, base::Bind( @@ -92,7 +92,7 @@ void P2PSocketClientImpl::SendWithDscp( void P2PSocketClientImpl::Send(const net::IPEndPoint& address, const std::vector<char>& data) { - talk_base::PacketOptions options(talk_base::DSCP_DEFAULT); + rtc::PacketOptions options(rtc::DSCP_DEFAULT); SendWithDscp(address, data, options); } diff --git a/content/renderer/p2p/socket_client_impl.h b/content/renderer/p2p/socket_client_impl.h index ae19758..89e48c0 100644 --- a/content/renderer/p2p/socket_client_impl.h +++ b/content/renderer/p2p/socket_client_impl.h @@ -47,7 +47,7 @@ class P2PSocketClientImpl : public P2PSocketClient { // |dscp|. virtual void SendWithDscp(const net::IPEndPoint& address, const std::vector<char>& data, - const talk_base::PacketOptions& options) OVERRIDE; + const rtc::PacketOptions& options) OVERRIDE; // Setting socket options. virtual void SetOption(P2PSocketOption option, int value) OVERRIDE; diff --git a/content/renderer/pepper/pepper_media_stream_video_track_host.cc b/content/renderer/pepper/pepper_media_stream_video_track_host.cc index 604d827..318965e 100644 --- a/content/renderer/pepper/pepper_media_stream_video_track_host.cc +++ b/content/renderer/pepper/pepper_media_stream_video_track_host.cc @@ -20,7 +20,7 @@ #include "ppapi/shared_impl/media_stream_buffer.h" // IS_ALIGNED is also defined in -// third_party/libjingle/overrides/talk/base/basictypes.h +// third_party/webrtc/overrides/webrtc/base/basictypes.h // TODO(ronghuawu): Avoid undef. #undef IS_ALIGNED #include "third_party/libyuv/include/libyuv.h" |