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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 |
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46 latest_telephone_event_.duration_ms = duration_ms; | 46 latest_telephone_event_.duration_ms = duration_ms; |
47 return true; | 47 return true; |
48 } | 48 } |
49 | 49 |
50 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { | 50 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { |
51 return stats_; | 51 return stats_; |
52 } | 52 } |
53 | 53 |
54 FakeAudioReceiveStream::FakeAudioReceiveStream( | 54 FakeAudioReceiveStream::FakeAudioReceiveStream( |
55 const webrtc::AudioReceiveStream::Config& config) | 55 const webrtc::AudioReceiveStream::Config& config) |
56 : config_(config), received_packets_(0) { | 56 : config_(config) { |
57 RTC_DCHECK(config.voe_channel_id != -1); | 57 RTC_DCHECK(config.voe_channel_id != -1); |
58 } | 58 } |
59 | 59 |
60 const webrtc::AudioReceiveStream::Config& | 60 const webrtc::AudioReceiveStream::Config& |
61 FakeAudioReceiveStream::GetConfig() const { | 61 FakeAudioReceiveStream::GetConfig() const { |
62 return config_; | 62 return config_; |
63 } | 63 } |
64 | 64 |
65 void FakeAudioReceiveStream::SetStats( | 65 void FakeAudioReceiveStream::SetStats( |
66 const webrtc::AudioReceiveStream::Stats& stats) { | 66 const webrtc::AudioReceiveStream::Stats& stats) { |
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82 | 82 |
83 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { | 83 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { |
84 return stats_; | 84 return stats_; |
85 } | 85 } |
86 | 86 |
87 void FakeAudioReceiveStream::SetSink( | 87 void FakeAudioReceiveStream::SetSink( |
88 std::unique_ptr<webrtc::AudioSinkInterface> sink) { | 88 std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
89 sink_ = std::move(sink); | 89 sink_ = std::move(sink); |
90 } | 90 } |
91 | 91 |
| 92 void FakeAudioReceiveStream::SetGain(float gain) { |
| 93 gain_ = gain; |
| 94 } |
| 95 |
92 FakeVideoSendStream::FakeVideoSendStream( | 96 FakeVideoSendStream::FakeVideoSendStream( |
93 const webrtc::VideoSendStream::Config& config, | 97 const webrtc::VideoSendStream::Config& config, |
94 const webrtc::VideoEncoderConfig& encoder_config) | 98 const webrtc::VideoEncoderConfig& encoder_config) |
95 : sending_(false), | 99 : sending_(false), |
96 config_(config), | 100 config_(config), |
97 codec_settings_set_(false), | 101 codec_settings_set_(false), |
98 num_swapped_frames_(0) { | 102 num_swapped_frames_(0) { |
99 RTC_DCHECK(config.encoder_settings.encoder != NULL); | 103 RTC_DCHECK(config.encoder_settings.encoder != NULL); |
100 ReconfigureVideoEncoder(encoder_config); | 104 ReconfigureVideoEncoder(encoder_config); |
101 } | 105 } |
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460 case webrtc::MediaType::ANY: | 464 case webrtc::MediaType::ANY: |
461 ADD_FAILURE() | 465 ADD_FAILURE() |
462 << "SignalChannelNetworkState called with unknown parameter."; | 466 << "SignalChannelNetworkState called with unknown parameter."; |
463 } | 467 } |
464 } | 468 } |
465 | 469 |
466 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 470 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
467 last_sent_packet_ = sent_packet; | 471 last_sent_packet_ = sent_packet; |
468 } | 472 } |
469 } // namespace cricket | 473 } // namespace cricket |
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