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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 int id, const webrtc::AudioSendStream::Config& config) | 24 int id, const webrtc::AudioSendStream::Config& config) |
| 25 : id_(id), config_(config) { | 25 : id_(id), config_(config) { |
| 26 RTC_DCHECK(config.voe_channel_id != -1); | 26 RTC_DCHECK(config.voe_channel_id != -1); |
| 27 } | 27 } |
| 28 | 28 |
| 29 void FakeAudioSendStream::Reconfigure( |
| 30 const webrtc::AudioSendStream::Config& config) { |
| 31 config_ = config; |
| 32 } |
| 33 |
| 29 const webrtc::AudioSendStream::Config& | 34 const webrtc::AudioSendStream::Config& |
| 30 FakeAudioSendStream::GetConfig() const { | 35 FakeAudioSendStream::GetConfig() const { |
| 31 return config_; | 36 return config_; |
| 32 } | 37 } |
| 33 | 38 |
| 34 void FakeAudioSendStream::SetStats( | 39 void FakeAudioSendStream::SetStats( |
| 35 const webrtc::AudioSendStream::Stats& stats) { | 40 const webrtc::AudioSendStream::Stats& stats) { |
| 36 stats_ = stats; | 41 stats_ = stats; |
| 37 } | 42 } |
| 38 | 43 |
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| 616 } | 621 } |
| 617 | 622 |
| 618 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { | 623 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { |
| 619 last_sent_packet_ = sent_packet; | 624 last_sent_packet_ = sent_packet; |
| 620 if (sent_packet.packet_id >= 0) { | 625 if (sent_packet.packet_id >= 0) { |
| 621 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; | 626 last_sent_nonnegative_packet_id_ = sent_packet.packet_id; |
| 622 } | 627 } |
| 623 } | 628 } |
| 624 | 629 |
| 625 } // namespace cricket | 630 } // namespace cricket |
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