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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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50 void FakeAudioSendStream::SetMuted(bool muted) { | 50 void FakeAudioSendStream::SetMuted(bool muted) { |
51 muted_ = muted; | 51 muted_ = muted; |
52 } | 52 } |
53 | 53 |
54 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { | 54 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { |
55 return stats_; | 55 return stats_; |
56 } | 56 } |
57 | 57 |
58 FakeAudioReceiveStream::FakeAudioReceiveStream( | 58 FakeAudioReceiveStream::FakeAudioReceiveStream( |
59 const webrtc::AudioReceiveStream::Config& config) | 59 const webrtc::AudioReceiveStream::Config& config) |
60 : config_(config), received_packets_(0) { | 60 : config_(config) { |
61 RTC_DCHECK(config.voe_channel_id != -1); | 61 RTC_DCHECK(config.voe_channel_id != -1); |
62 } | 62 } |
63 | 63 |
64 const webrtc::AudioReceiveStream::Config& | 64 const webrtc::AudioReceiveStream::Config& |
65 FakeAudioReceiveStream::GetConfig() const { | 65 FakeAudioReceiveStream::GetConfig() const { |
66 return config_; | 66 return config_; |
67 } | 67 } |
68 | 68 |
69 void FakeAudioReceiveStream::SetStats( | 69 void FakeAudioReceiveStream::SetStats( |
70 const webrtc::AudioReceiveStream::Stats& stats) { | 70 const webrtc::AudioReceiveStream::Stats& stats) { |
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86 | 86 |
87 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { | 87 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { |
88 return stats_; | 88 return stats_; |
89 } | 89 } |
90 | 90 |
91 void FakeAudioReceiveStream::SetSink( | 91 void FakeAudioReceiveStream::SetSink( |
92 std::unique_ptr<webrtc::AudioSinkInterface> sink) { | 92 std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
93 sink_ = std::move(sink); | 93 sink_ = std::move(sink); |
94 } | 94 } |
95 | 95 |
| 96 void FakeAudioReceiveStream::SetGain(float gain) { |
| 97 gain_ = gain; |
| 98 } |
| 99 |
96 FakeVideoSendStream::FakeVideoSendStream( | 100 FakeVideoSendStream::FakeVideoSendStream( |
97 const webrtc::VideoSendStream::Config& config, | 101 const webrtc::VideoSendStream::Config& config, |
98 const webrtc::VideoEncoderConfig& encoder_config) | 102 const webrtc::VideoEncoderConfig& encoder_config) |
99 : sending_(false), | 103 : sending_(false), |
100 config_(config), | 104 config_(config), |
101 codec_settings_set_(false), | 105 codec_settings_set_(false), |
102 num_swapped_frames_(0) { | 106 num_swapped_frames_(0) { |
103 RTC_DCHECK(config.encoder_settings.encoder != NULL); | 107 RTC_DCHECK(config.encoder_settings.encoder != NULL); |
104 ReconfigureVideoEncoder(encoder_config); | 108 ReconfigureVideoEncoder(encoder_config); |
105 } | 109 } |
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468 } | 472 } |
469 | 473 |
470 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 474 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
471 last_sent_packet_ = sent_packet; | 475 last_sent_packet_ = sent_packet; |
472 if (sent_packet.packet_id >= 0) { | 476 if (sent_packet.packet_id >= 0) { |
473 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 477 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |
474 } | 478 } |
475 } | 479 } |
476 | 480 |
477 } // namespace cricket | 481 } // namespace cricket |
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