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Side by Side 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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1 /*
2 * libjingle
3 * Copyright 2012 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28 #ifndef TALK_MEDIA_BASE_RTPDATAENGINE_H_
29 #define TALK_MEDIA_BASE_RTPDATAENGINE_H_
30
31 #include <string>
32 #include <vector>
33
34 #include "talk/media/base/constants.h"
35 #include "talk/media/base/mediachannel.h"
36 #include "talk/media/base/mediaengine.h"
37 #include "webrtc/base/timing.h"
38
39 namespace cricket {
40
41 struct DataCodec;
42
43 class RtpDataEngine : public DataEngineInterface {
44 public:
45 RtpDataEngine();
46
47 virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);
48
49 virtual const std::vector<DataCodec>& data_codecs() {
50 return data_codecs_;
51 }
52
53 // Mostly for testing with a fake clock. Ownership is passed in.
54 void SetTiming(rtc::Timing* timing) {
55 timing_.reset(timing);
56 }
57
58 private:
59 std::vector<DataCodec> data_codecs_;
60 rtc::scoped_ptr<rtc::Timing> timing_;
61 };
62
63 // Keep track of sequence number and timestamp of an RTP stream. The
64 // sequence number starts with a "random" value and increments. The
65 // timestamp starts with a "random" value and increases monotonically
66 // according to the clockrate.
67 class RtpClock {
68 public:
69 RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
70 : clockrate_(clockrate),
71 last_seq_num_(first_seq_num),
72 timestamp_offset_(timestamp_offset) {}
73
74 // Given the current time (in number of seconds which must be
75 // monotonically increasing), Return the next sequence number and
76 // timestamp.
77 void Tick(double now, int* seq_num, uint32_t* timestamp);
78
79 private:
80 int clockrate_;
81 uint16_t last_seq_num_;
82 uint32_t timestamp_offset_;
83 };
84
85 class RtpDataMediaChannel : public DataMediaChannel {
86 public:
87 // Timing* Used for the RtpClock
88 explicit RtpDataMediaChannel(rtc::Timing* timing);
89 // Sets Timing == NULL, so you'll need to call set_timer() before
90 // using it. This is needed by FakeMediaEngine.
91 RtpDataMediaChannel();
92 virtual ~RtpDataMediaChannel();
93
94 void set_timing(rtc::Timing* timing) {
95 timing_ = timing;
96 }
97
98 virtual bool SetSendParameters(const DataSendParameters& params);
99 virtual bool SetRecvParameters(const DataRecvParameters& params);
100 virtual bool AddSendStream(const StreamParams& sp);
101 virtual bool RemoveSendStream(uint32_t ssrc);
102 virtual bool AddRecvStream(const StreamParams& sp);
103 virtual bool RemoveRecvStream(uint32_t ssrc);
104 virtual bool SetSend(bool send) {
105 sending_ = send;
106 return true;
107 }
108 virtual bool SetReceive(bool receive) {
109 receiving_ = receive;
110 return true;
111 }
112 virtual void OnPacketReceived(rtc::Buffer* packet,
113 const rtc::PacketTime& packet_time);
114 virtual void OnRtcpReceived(rtc::Buffer* packet,
115 const rtc::PacketTime& packet_time) {}
116 virtual void OnReadyToSend(bool ready) {}
117 virtual bool SendData(
118 const SendDataParams& params,
119 const rtc::Buffer& payload,
120 SendDataResult* result);
121
122 private:
123 void Construct(rtc::Timing* timing);
124 bool SetMaxSendBandwidth(int bps);
125 bool SetSendCodecs(const std::vector<DataCodec>& codecs);
126 bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
127
128 bool sending_;
129 bool receiving_;
130 rtc::Timing* timing_;
131 std::vector<DataCodec> send_codecs_;
132 std::vector<DataCodec> recv_codecs_;
133 std::vector<StreamParams> send_streams_;
134 std::vector<StreamParams> recv_streams_;
135 std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
136 rtc::scoped_ptr<rtc::RateLimiter> send_limiter_;
137 };
138
139 } // namespace cricket
140
141 #endif // TALK_MEDIA_BASE_RTPDATAENGINE_H_
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