I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
142 lines
4.8 KiB
C++
142 lines
4.8 KiB
C++
/*
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* libjingle
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* Copyright 2012 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
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#define WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
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#include <string>
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#include <vector>
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#include "webrtc/base/timing.h"
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#include "webrtc/media/base/constants.h"
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#include "webrtc/media/base/mediachannel.h"
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#include "webrtc/media/base/mediaengine.h"
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namespace cricket {
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struct DataCodec;
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class RtpDataEngine : public DataEngineInterface {
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public:
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RtpDataEngine();
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virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);
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virtual const std::vector<DataCodec>& data_codecs() {
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return data_codecs_;
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}
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// Mostly for testing with a fake clock. Ownership is passed in.
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void SetTiming(rtc::Timing* timing) {
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timing_.reset(timing);
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}
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private:
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std::vector<DataCodec> data_codecs_;
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rtc::scoped_ptr<rtc::Timing> timing_;
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};
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// Keep track of sequence number and timestamp of an RTP stream. The
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// sequence number starts with a "random" value and increments. The
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// timestamp starts with a "random" value and increases monotonically
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// according to the clockrate.
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class RtpClock {
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public:
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RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
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: clockrate_(clockrate),
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last_seq_num_(first_seq_num),
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timestamp_offset_(timestamp_offset) {}
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// Given the current time (in number of seconds which must be
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// monotonically increasing), Return the next sequence number and
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// timestamp.
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void Tick(double now, int* seq_num, uint32_t* timestamp);
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private:
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int clockrate_;
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uint16_t last_seq_num_;
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uint32_t timestamp_offset_;
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};
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class RtpDataMediaChannel : public DataMediaChannel {
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public:
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// Timing* Used for the RtpClock
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explicit RtpDataMediaChannel(rtc::Timing* timing);
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// Sets Timing == NULL, so you'll need to call set_timer() before
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// using it. This is needed by FakeMediaEngine.
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RtpDataMediaChannel();
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virtual ~RtpDataMediaChannel();
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void set_timing(rtc::Timing* timing) {
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timing_ = timing;
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}
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virtual bool SetSendParameters(const DataSendParameters& params);
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virtual bool SetRecvParameters(const DataRecvParameters& params);
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virtual bool AddSendStream(const StreamParams& sp);
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virtual bool RemoveSendStream(uint32_t ssrc);
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virtual bool AddRecvStream(const StreamParams& sp);
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virtual bool RemoveRecvStream(uint32_t ssrc);
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virtual bool SetSend(bool send) {
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sending_ = send;
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return true;
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}
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virtual bool SetReceive(bool receive) {
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receiving_ = receive;
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return true;
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}
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virtual void OnPacketReceived(rtc::Buffer* packet,
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const rtc::PacketTime& packet_time);
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virtual void OnRtcpReceived(rtc::Buffer* packet,
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const rtc::PacketTime& packet_time) {}
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virtual void OnReadyToSend(bool ready) {}
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virtual bool SendData(
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const SendDataParams& params,
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const rtc::Buffer& payload,
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SendDataResult* result);
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private:
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void Construct(rtc::Timing* timing);
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bool SetMaxSendBandwidth(int bps);
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bool SetSendCodecs(const std::vector<DataCodec>& codecs);
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bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
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bool sending_;
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bool receiving_;
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rtc::Timing* timing_;
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std::vector<DataCodec> send_codecs_;
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std::vector<DataCodec> recv_codecs_;
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std::vector<StreamParams> send_streams_;
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std::vector<StreamParams> recv_streams_;
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std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
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rtc::scoped_ptr<rtc::RateLimiter> send_limiter_;
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};
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} // namespace cricket
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#endif // WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
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