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Side by Side Diff: webrtc/video/rampup_tests.h

Issue 1208083002: Remove MimdRateControl and factories for RemoteBitrateEstimor. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated thresholds for TestTimestampGrouping Created 5 years, 6 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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33 class RtpHeaderParser; 33 class RtpHeaderParser;
34 class RTPPayloadRegistry; 34 class RTPPayloadRegistry;
35 class RtpRtcp; 35 class RtpRtcp;
36 36
37 class StreamObserver : public newapi::Transport, public RemoteBitrateObserver { 37 class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
38 public: 38 public:
39 typedef std::map<uint32_t, int> BytesSentMap; 39 typedef std::map<uint32_t, int> BytesSentMap;
40 typedef std::map<uint32_t, uint32_t> SsrcMap; 40 typedef std::map<uint32_t, uint32_t> SsrcMap;
41 StreamObserver(const SsrcMap& rtx_media_ssrcs, 41 StreamObserver(const SsrcMap& rtx_media_ssrcs,
42 newapi::Transport* feedback_transport, 42 newapi::Transport* feedback_transport,
43 Clock* clock, 43 Clock* clock);
44 RemoteBitrateEstimatorFactory* rbe_factory,
45 RateControlType control_type);
46 44
47 void set_expected_bitrate_bps(unsigned int expected_bitrate_bps); 45 void set_expected_bitrate_bps(unsigned int expected_bitrate_bps);
48 46
49 void set_start_bitrate_bps(unsigned int start_bitrate_bps); 47 void set_start_bitrate_bps(unsigned int start_bitrate_bps);
50 48
51 void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, 49 void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
52 unsigned int bitrate) override; 50 unsigned int bitrate) override;
53 51
54 bool SendRtp(const uint8_t* packet, size_t length) override; 52 bool SendRtp(const uint8_t* packet, size_t length) override;
55 53
56 bool SendRtcp(const uint8_t* packet, size_t length) override; 54 bool SendRtcp(const uint8_t* packet, size_t length) override;
57 55
58 EventTypeWrapper Wait(); 56 EventTypeWrapper Wait();
59 57
58 void SetRemoteBitrateEstimator(RemoteBitrateEstimator* rbe);
59
60 private: 60 private:
61 void ReportResult(const std::string& measurement, 61 void ReportResult(const std::string& measurement,
62 size_t value, 62 size_t value,
63 const std::string& units); 63 const std::string& units);
64 void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_); 64 void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_);
65 65
66 Clock* const clock_; 66 Clock* const clock_;
67 const rtc::scoped_ptr<EventWrapper> test_done_; 67 const rtc::scoped_ptr<EventWrapper> test_done_;
68 const rtc::scoped_ptr<RtpHeaderParser> rtp_parser_; 68 const rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
69 rtc::scoped_ptr<RtpRtcp> rtp_rtcp_; 69 rtc::scoped_ptr<RtpRtcp> rtp_rtcp_;
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150 void RunRampUpTest(size_t num_streams, 150 void RunRampUpTest(size_t num_streams,
151 unsigned int start_bitrate_bps, 151 unsigned int start_bitrate_bps,
152 const std::string& extension_type, 152 const std::string& extension_type,
153 bool rtx, 153 bool rtx,
154 bool red); 154 bool red);
155 155
156 void RunRampUpDownUpTest(size_t number_of_streams, bool rtx, bool red); 156 void RunRampUpDownUpTest(size_t number_of_streams, bool rtx, bool red);
157 }; 157 };
158 } // namespace webrtc 158 } // namespace webrtc
159 #endif // WEBRTC_VIDEO_RAMPUP_TESTS_H_ 159 #endif // WEBRTC_VIDEO_RAMPUP_TESTS_H_
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