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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 #include "webrtc/media/engine/fakewebrtccall.h" | 11 #include "webrtc/media/engine/fakewebrtccall.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 #include <utility> | 14 #include <utility> |
15 | 15 |
16 #include "webrtc/api/call/audio_sink.h" | 16 #include "webrtc/api/call/audio_sink.h" |
17 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
18 #include "webrtc/base/platform_file.h" | 18 #include "webrtc/base/platform_file.h" |
19 #include "webrtc/base/gunit.h" | 19 #include "webrtc/base/gunit.h" |
20 #include "webrtc/media/base/rtputils.h" | 20 #include "webrtc/media/base/rtputils.h" |
21 | 21 |
22 namespace cricket { | 22 namespace cricket { |
23 FakeAudioSendStream::FakeAudioSendStream( | 23 FakeAudioSendStream::FakeAudioSendStream( |
24 const webrtc::AudioSendStream::Config& config) : config_(config) { | 24 int id, const webrtc::AudioSendStream::Config& config) |
| 25 : id_(id), config_(config) { |
25 RTC_DCHECK(config.voe_channel_id != -1); | 26 RTC_DCHECK(config.voe_channel_id != -1); |
26 } | 27 } |
27 | 28 |
28 const webrtc::AudioSendStream::Config& | 29 const webrtc::AudioSendStream::Config& |
29 FakeAudioSendStream::GetConfig() const { | 30 FakeAudioSendStream::GetConfig() const { |
30 return config_; | 31 return config_; |
31 } | 32 } |
32 | 33 |
33 void FakeAudioSendStream::SetStats( | 34 void FakeAudioSendStream::SetStats( |
34 const webrtc::AudioSendStream::Stats& stats) { | 35 const webrtc::AudioSendStream::Stats& stats) { |
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52 | 53 |
53 void FakeAudioSendStream::SetMuted(bool muted) { | 54 void FakeAudioSendStream::SetMuted(bool muted) { |
54 muted_ = muted; | 55 muted_ = muted; |
55 } | 56 } |
56 | 57 |
57 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { | 58 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { |
58 return stats_; | 59 return stats_; |
59 } | 60 } |
60 | 61 |
61 FakeAudioReceiveStream::FakeAudioReceiveStream( | 62 FakeAudioReceiveStream::FakeAudioReceiveStream( |
62 const webrtc::AudioReceiveStream::Config& config) | 63 int id, const webrtc::AudioReceiveStream::Config& config) |
63 : config_(config) { | 64 : id_(id), config_(config) { |
64 RTC_DCHECK(config.voe_channel_id != -1); | 65 RTC_DCHECK(config.voe_channel_id != -1); |
65 } | 66 } |
66 | 67 |
67 const webrtc::AudioReceiveStream::Config& | 68 const webrtc::AudioReceiveStream::Config& |
68 FakeAudioReceiveStream::GetConfig() const { | 69 FakeAudioReceiveStream::GetConfig() const { |
69 return config_; | 70 return config_; |
70 } | 71 } |
71 | 72 |
72 void FakeAudioReceiveStream::SetStats( | 73 void FakeAudioReceiveStream::SetStats( |
73 const webrtc::AudioReceiveStream::Stats& stats) { | 74 const webrtc::AudioReceiveStream::Stats& stats) { |
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397 } | 398 } |
398 // Even though all the values for the enum class are listed above,the compiler | 399 // Even though all the values for the enum class are listed above,the compiler |
399 // will emit a warning as the method may be called with a value outside of the | 400 // will emit a warning as the method may be called with a value outside of the |
400 // valid enum range, unless this case is also handled. | 401 // valid enum range, unless this case is also handled. |
401 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; | 402 ADD_FAILURE() << "GetNetworkState called with unknown parameter."; |
402 return webrtc::kNetworkDown; | 403 return webrtc::kNetworkDown; |
403 } | 404 } |
404 | 405 |
405 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( | 406 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( |
406 const webrtc::AudioSendStream::Config& config) { | 407 const webrtc::AudioSendStream::Config& config) { |
407 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config); | 408 FakeAudioSendStream* fake_stream = new FakeAudioSendStream(next_stream_id_++, |
| 409 config); |
408 audio_send_streams_.push_back(fake_stream); | 410 audio_send_streams_.push_back(fake_stream); |
409 ++num_created_send_streams_; | 411 ++num_created_send_streams_; |
410 return fake_stream; | 412 return fake_stream; |
411 } | 413 } |
412 | 414 |
413 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { | 415 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { |
414 auto it = std::find(audio_send_streams_.begin(), | 416 auto it = std::find(audio_send_streams_.begin(), |
415 audio_send_streams_.end(), | 417 audio_send_streams_.end(), |
416 static_cast<FakeAudioSendStream*>(send_stream)); | 418 static_cast<FakeAudioSendStream*>(send_stream)); |
417 if (it == audio_send_streams_.end()) { | 419 if (it == audio_send_streams_.end()) { |
418 ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter."; | 420 ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter."; |
419 } else { | 421 } else { |
420 delete *it; | 422 delete *it; |
421 audio_send_streams_.erase(it); | 423 audio_send_streams_.erase(it); |
422 } | 424 } |
423 } | 425 } |
424 | 426 |
425 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream( | 427 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream( |
426 const webrtc::AudioReceiveStream::Config& config) { | 428 const webrtc::AudioReceiveStream::Config& config) { |
427 audio_receive_streams_.push_back(new FakeAudioReceiveStream(config)); | 429 audio_receive_streams_.push_back(new FakeAudioReceiveStream(next_stream_id_++, |
| 430 config)); |
428 ++num_created_receive_streams_; | 431 ++num_created_receive_streams_; |
429 return audio_receive_streams_.back(); | 432 return audio_receive_streams_.back(); |
430 } | 433 } |
431 | 434 |
432 void FakeCall::DestroyAudioReceiveStream( | 435 void FakeCall::DestroyAudioReceiveStream( |
433 webrtc::AudioReceiveStream* receive_stream) { | 436 webrtc::AudioReceiveStream* receive_stream) { |
434 auto it = std::find(audio_receive_streams_.begin(), | 437 auto it = std::find(audio_receive_streams_.begin(), |
435 audio_receive_streams_.end(), | 438 audio_receive_streams_.end(), |
436 static_cast<FakeAudioReceiveStream*>(receive_stream)); | 439 static_cast<FakeAudioReceiveStream*>(receive_stream)); |
437 if (it == audio_receive_streams_.end()) { | 440 if (it == audio_receive_streams_.end()) { |
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594 } | 597 } |
595 | 598 |
596 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 599 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
597 last_sent_packet_ = sent_packet; | 600 last_sent_packet_ = sent_packet; |
598 if (sent_packet.packet_id >= 0) { | 601 if (sent_packet.packet_id >= 0) { |
599 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 602 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |
600 } | 603 } |
601 } | 604 } |
602 | 605 |
603 } // namespace cricket | 606 } // namespace cricket |
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