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Side by Side Diff: talk/media/webrtc/fakewebrtccall.cc

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Address comments Created 5 years ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2015 Google Inc. 3 * Copyright 2015 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 10 matching lines...) Expand all
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 26 */
27 27
28 #include "talk/media/webrtc/fakewebrtccall.h" 28 #include "talk/media/webrtc/fakewebrtccall.h"
29 29
30 #include <algorithm> 30 #include <algorithm>
31 #include <utility>
31 32
32 #include "talk/media/base/rtputils.h" 33 #include "talk/media/base/rtputils.h"
33 #include "webrtc/base/checks.h" 34 #include "webrtc/base/checks.h"
34 #include "webrtc/base/gunit.h" 35 #include "webrtc/base/gunit.h"
36 #include "webrtc/audio/audio_sink.h"
35 37
36 namespace cricket { 38 namespace cricket {
37 FakeAudioSendStream::FakeAudioSendStream( 39 FakeAudioSendStream::FakeAudioSendStream(
38 const webrtc::AudioSendStream::Config& config) : config_(config) { 40 const webrtc::AudioSendStream::Config& config) : config_(config) {
39 RTC_DCHECK(config.voe_channel_id != -1); 41 RTC_DCHECK(config.voe_channel_id != -1);
40 } 42 }
41 43
42 const webrtc::AudioSendStream::Config& 44 const webrtc::AudioSendStream::Config&
43 FakeAudioSendStream::GetConfig() const { 45 FakeAudioSendStream::GetConfig() const {
44 return config_; 46 return config_;
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83 } 85 }
84 86
85 void FakeAudioReceiveStream::IncrementReceivedPackets() { 87 void FakeAudioReceiveStream::IncrementReceivedPackets() {
86 received_packets_++; 88 received_packets_++;
87 } 89 }
88 90
89 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { 91 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
90 return stats_; 92 return stats_;
91 } 93 }
92 94
95 void FakeAudioReceiveStream::SetSink(
96 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
97 sink_ = std::move(sink);
98 }
99
93 FakeVideoSendStream::FakeVideoSendStream( 100 FakeVideoSendStream::FakeVideoSendStream(
94 const webrtc::VideoSendStream::Config& config, 101 const webrtc::VideoSendStream::Config& config,
95 const webrtc::VideoEncoderConfig& encoder_config) 102 const webrtc::VideoEncoderConfig& encoder_config)
96 : sending_(false), 103 : sending_(false),
97 config_(config), 104 config_(config),
98 codec_settings_set_(false), 105 codec_settings_set_(false),
99 num_swapped_frames_(0) { 106 num_swapped_frames_(0) {
100 RTC_DCHECK(config.encoder_settings.encoder != NULL); 107 RTC_DCHECK(config.encoder_settings.encoder != NULL);
101 ReconfigureVideoEncoder(encoder_config); 108 ReconfigureVideoEncoder(encoder_config);
102 } 109 }
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427 } 434 }
428 435
429 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { 436 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
430 network_state_ = state; 437 network_state_ = state;
431 } 438 }
432 439
433 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { 440 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
434 last_sent_packet_ = sent_packet; 441 last_sent_packet_ = sent_packet;
435 } 442 }
436 } // namespace cricket 443 } // namespace cricket
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