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Unified Diff: talk/media/base/rtpdataengine.h

Issue 1587193006: Move talk/media to webrtc/media (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased to b647aca12a884a13c1728118586245399b55fa3d (#11493) Created 4 years, 10 months ago
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Index: talk/media/base/rtpdataengine.h
diff --git a/talk/media/base/rtpdataengine.h b/talk/media/base/rtpdataengine.h
deleted file mode 100644
index c0449c9f796ff0d0f171bd75c9161e2901ad19ed..0000000000000000000000000000000000000000
--- a/talk/media/base/rtpdataengine.h
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_MEDIA_BASE_RTPDATAENGINE_H_
-#define TALK_MEDIA_BASE_RTPDATAENGINE_H_
-
-#include <string>
-#include <vector>
-
-#include "talk/media/base/constants.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/mediaengine.h"
-#include "webrtc/base/timing.h"
-
-namespace cricket {
-
-struct DataCodec;
-
-class RtpDataEngine : public DataEngineInterface {
- public:
- RtpDataEngine();
-
- virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);
-
- virtual const std::vector<DataCodec>& data_codecs() {
- return data_codecs_;
- }
-
- // Mostly for testing with a fake clock. Ownership is passed in.
- void SetTiming(rtc::Timing* timing) {
- timing_.reset(timing);
- }
-
- private:
- std::vector<DataCodec> data_codecs_;
- rtc::scoped_ptr<rtc::Timing> timing_;
-};
-
-// Keep track of sequence number and timestamp of an RTP stream. The
-// sequence number starts with a "random" value and increments. The
-// timestamp starts with a "random" value and increases monotonically
-// according to the clockrate.
-class RtpClock {
- public:
- RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
- : clockrate_(clockrate),
- last_seq_num_(first_seq_num),
- timestamp_offset_(timestamp_offset) {}
-
- // Given the current time (in number of seconds which must be
- // monotonically increasing), Return the next sequence number and
- // timestamp.
- void Tick(double now, int* seq_num, uint32_t* timestamp);
-
- private:
- int clockrate_;
- uint16_t last_seq_num_;
- uint32_t timestamp_offset_;
-};
-
-class RtpDataMediaChannel : public DataMediaChannel {
- public:
- // Timing* Used for the RtpClock
- explicit RtpDataMediaChannel(rtc::Timing* timing);
- // Sets Timing == NULL, so you'll need to call set_timer() before
- // using it. This is needed by FakeMediaEngine.
- RtpDataMediaChannel();
- virtual ~RtpDataMediaChannel();
-
- void set_timing(rtc::Timing* timing) {
- timing_ = timing;
- }
-
- virtual bool SetSendParameters(const DataSendParameters& params);
- virtual bool SetRecvParameters(const DataRecvParameters& params);
- virtual bool AddSendStream(const StreamParams& sp);
- virtual bool RemoveSendStream(uint32_t ssrc);
- virtual bool AddRecvStream(const StreamParams& sp);
- virtual bool RemoveRecvStream(uint32_t ssrc);
- virtual bool SetSend(bool send) {
- sending_ = send;
- return true;
- }
- virtual bool SetReceive(bool receive) {
- receiving_ = receive;
- return true;
- }
- virtual void OnPacketReceived(rtc::Buffer* packet,
- const rtc::PacketTime& packet_time);
- virtual void OnRtcpReceived(rtc::Buffer* packet,
- const rtc::PacketTime& packet_time) {}
- virtual void OnReadyToSend(bool ready) {}
- virtual bool SendData(
- const SendDataParams& params,
- const rtc::Buffer& payload,
- SendDataResult* result);
-
- private:
- void Construct(rtc::Timing* timing);
- bool SetMaxSendBandwidth(int bps);
- bool SetSendCodecs(const std::vector<DataCodec>& codecs);
- bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
-
- bool sending_;
- bool receiving_;
- rtc::Timing* timing_;
- std::vector<DataCodec> send_codecs_;
- std::vector<DataCodec> recv_codecs_;
- std::vector<StreamParams> send_streams_;
- std::vector<StreamParams> recv_streams_;
- std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
- rtc::scoped_ptr<rtc::RateLimiter> send_limiter_;
-};
-
-} // namespace cricket
-
-#endif // TALK_MEDIA_BASE_RTPDATAENGINE_H_
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