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

Issue 1706183002: Replace scoped_ptr with unique_ptr in webrtc/audio/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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
11 // This file contains fake implementations, for use in unit tests, of the 11 // This file contains fake implementations, for use in unit tests, of the
12 // following classes: 12 // following classes:
13 // 13 //
14 // webrtc::Call 14 // webrtc::Call
15 // webrtc::AudioSendStream 15 // webrtc::AudioSendStream
16 // webrtc::AudioReceiveStream 16 // webrtc::AudioReceiveStream
17 // webrtc::VideoSendStream 17 // webrtc::VideoSendStream
18 // webrtc::VideoReceiveStream 18 // webrtc::VideoReceiveStream
19 19
20 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ 20 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
21 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ 21 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
22 22
23 #include <memory>
23 #include <vector> 24 #include <vector>
24 25
25 #include "webrtc/audio_receive_stream.h" 26 #include "webrtc/audio_receive_stream.h"
26 #include "webrtc/audio_send_stream.h" 27 #include "webrtc/audio_send_stream.h"
27 #include "webrtc/call.h" 28 #include "webrtc/call.h"
28 #include "webrtc/video_frame.h" 29 #include "webrtc/video_frame.h"
29 #include "webrtc/video_receive_stream.h" 30 #include "webrtc/video_receive_stream.h"
30 #include "webrtc/video_send_stream.h" 31 #include "webrtc/video_send_stream.h"
31 32
32 namespace cricket { 33 namespace cricket {
(...skipping 50 matching lines...) Expand 10 before | Expand all | Expand 10 after
83 return true; 84 return true;
84 } 85 }
85 bool DeliverRtp(const uint8_t* packet, 86 bool DeliverRtp(const uint8_t* packet,
86 size_t length, 87 size_t length,
87 const webrtc::PacketTime& packet_time) override { 88 const webrtc::PacketTime& packet_time) override {
88 return true; 89 return true;
89 } 90 }
90 91
91 // webrtc::AudioReceiveStream implementation. 92 // webrtc::AudioReceiveStream implementation.
92 webrtc::AudioReceiveStream::Stats GetStats() const override; 93 webrtc::AudioReceiveStream::Stats GetStats() const override;
93 void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override; 94 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
94 95
95 webrtc::AudioReceiveStream::Config config_; 96 webrtc::AudioReceiveStream::Config config_;
96 webrtc::AudioReceiveStream::Stats stats_; 97 webrtc::AudioReceiveStream::Stats stats_;
97 int received_packets_; 98 int received_packets_;
98 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; 99 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
the sun 2016/02/18 18:54:44 Change me!
kwiberg-webrtc 2016/02/18 20:02:16 In a later CL. This one changes only webrtc/audio/
99 }; 100 };
100 101
101 class FakeVideoSendStream final : public webrtc::VideoSendStream, 102 class FakeVideoSendStream final : public webrtc::VideoSendStream,
102 public webrtc::VideoCaptureInput { 103 public webrtc::VideoCaptureInput {
103 public: 104 public:
104 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, 105 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
105 const webrtc::VideoEncoderConfig& encoder_config); 106 const webrtc::VideoEncoderConfig& encoder_config);
106 webrtc::VideoSendStream::Config GetConfig() const; 107 webrtc::VideoSendStream::Config GetConfig() const;
107 webrtc::VideoEncoderConfig GetEncoderConfig() const; 108 webrtc::VideoEncoderConfig GetEncoderConfig() const;
108 std::vector<webrtc::VideoStream> GetVideoStreams(); 109 std::vector<webrtc::VideoStream> GetVideoStreams();
(...skipping 134 matching lines...) Expand 10 before | Expand all | Expand 10 after
243 std::vector<FakeAudioSendStream*> audio_send_streams_; 244 std::vector<FakeAudioSendStream*> audio_send_streams_;
244 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 245 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
245 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 246 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
246 247
247 int num_created_send_streams_; 248 int num_created_send_streams_;
248 int num_created_receive_streams_; 249 int num_created_receive_streams_;
249 }; 250 };
250 251
251 } // namespace cricket 252 } // namespace cricket
252 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 253 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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