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Side by Side Diff: talk/media/webrtc/fakewebrtccall.h

Issue 1414743004: Implement AudioSendStream::GetStats(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: workaround for android build issue Created 5 years, 1 month ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2015 Google Inc. 3 * Copyright 2015 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 35 matching lines...) Expand 10 before | Expand all | Expand 10 after
46 #include "webrtc/video_receive_stream.h" 46 #include "webrtc/video_receive_stream.h"
47 #include "webrtc/video_send_stream.h" 47 #include "webrtc/video_send_stream.h"
48 48
49 namespace cricket { 49 namespace cricket {
50 50
51 class FakeAudioSendStream : public webrtc::AudioSendStream { 51 class FakeAudioSendStream : public webrtc::AudioSendStream {
52 public: 52 public:
53 explicit FakeAudioSendStream( 53 explicit FakeAudioSendStream(
54 const webrtc::AudioSendStream::Config& config); 54 const webrtc::AudioSendStream::Config& config);
55 55
56 // webrtc::AudioSendStream implementation.
57 webrtc::AudioSendStream::Stats GetStats() const override;
58
59 const webrtc::AudioSendStream::Config& GetConfig() const; 56 const webrtc::AudioSendStream::Config& GetConfig() const;
57 void SetStats(const webrtc::AudioSendStream::Stats& stats);
60 58
61 private: 59 private:
62 // webrtc::SendStream implementation. 60 // webrtc::SendStream implementation.
63 void Start() override {} 61 void Start() override {}
64 void Stop() override {} 62 void Stop() override {}
65 void SignalNetworkState(webrtc::NetworkState state) override {} 63 void SignalNetworkState(webrtc::NetworkState state) override {}
66 bool DeliverRtcp(const uint8_t* packet, size_t length) override { 64 bool DeliverRtcp(const uint8_t* packet, size_t length) override {
67 return true; 65 return true;
68 } 66 }
69 67
68 // webrtc::AudioSendStream implementation.
69 webrtc::AudioSendStream::Stats GetStats() const override;
70
70 webrtc::AudioSendStream::Config config_; 71 webrtc::AudioSendStream::Config config_;
72 webrtc::AudioSendStream::Stats stats_;
71 }; 73 };
72 74
73 class FakeAudioReceiveStream : public webrtc::AudioReceiveStream { 75 class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
74 public: 76 public:
75 explicit FakeAudioReceiveStream( 77 explicit FakeAudioReceiveStream(
76 const webrtc::AudioReceiveStream::Config& config); 78 const webrtc::AudioReceiveStream::Config& config);
77 79
78 const webrtc::AudioReceiveStream::Config& GetConfig() const; 80 const webrtc::AudioReceiveStream::Config& GetConfig() const;
79 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); 81 void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
80 int received_packets() const { return received_packets_; } 82 int received_packets() const { return received_packets_; }
81 void IncrementReceivedPackets(); 83 void IncrementReceivedPackets();
82 84
83 private: 85 private:
84 // webrtc::ReceiveStream implementation. 86 // webrtc::ReceiveStream implementation.
85 void Start() override {} 87 void Start() override {}
86 void Stop() override {} 88 void Stop() override {}
87 void SignalNetworkState(webrtc::NetworkState state) override {} 89 void SignalNetworkState(webrtc::NetworkState state) override {}
88 bool DeliverRtcp(const uint8_t* packet, size_t length) override { 90 bool DeliverRtcp(const uint8_t* packet, size_t length) override {
89 return true; 91 return true;
90 } 92 }
91 bool DeliverRtp(const uint8_t* packet, 93 bool DeliverRtp(const uint8_t* packet,
92 size_t length, 94 size_t length,
93 const webrtc::PacketTime& packet_time) override { 95 const webrtc::PacketTime& packet_time) override {
94 return true; 96 return true;
95 } 97 }
96 98
97 // webrtc::AudioReceiveStream implementation. 99 // webrtc::AudioReceiveStream implementation.
98 webrtc::AudioReceiveStream::Stats GetStats() const override { 100 webrtc::AudioReceiveStream::Stats GetStats() const override;
99 return stats_;
100 }
101 101
102 webrtc::AudioReceiveStream::Config config_; 102 webrtc::AudioReceiveStream::Config config_;
103 webrtc::AudioReceiveStream::Stats stats_; 103 webrtc::AudioReceiveStream::Stats stats_;
104 int received_packets_; 104 int received_packets_;
105 }; 105 };
106 106
107 class FakeVideoSendStream : public webrtc::VideoSendStream, 107 class FakeVideoSendStream : public webrtc::VideoSendStream,
108 public webrtc::VideoCaptureInput { 108 public webrtc::VideoCaptureInput {
109 public: 109 public:
110 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, 110 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
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249 std::vector<FakeAudioSendStream*> audio_send_streams_; 249 std::vector<FakeAudioSendStream*> audio_send_streams_;
250 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 250 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
251 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 251 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
252 252
253 int num_created_send_streams_; 253 int num_created_send_streams_;
254 int num_created_receive_streams_; 254 int num_created_receive_streams_;
255 }; 255 };
256 256
257 } // namespace cricket 257 } // namespace cricket
258 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 258 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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