Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(467)

Side by Side Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 1909333002: Switch voice transport to use Call and Stream instead of VoENetwork. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed coments on ps#6 Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/audio/audio_send_stream.cc ('k') | webrtc/audio_receive_stream.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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
(...skipping 71 matching lines...) Expand 10 before | Expand all | Expand 10 after
82 kTransportSequenceNumberId)) 82 kTransportSequenceNumberId))
83 .Times(1); 83 .Times(1);
84 EXPECT_CALL(*channel_proxy_, 84 EXPECT_CALL(*channel_proxy_,
85 RegisterSenderCongestionControlObjects( 85 RegisterSenderCongestionControlObjects(
86 congestion_controller_.pacer(), 86 congestion_controller_.pacer(),
87 congestion_controller_.GetTransportFeedbackObserver(), 87 congestion_controller_.GetTransportFeedbackObserver(),
88 congestion_controller_.packet_router())) 88 congestion_controller_.packet_router()))
89 .Times(1); 89 .Times(1);
90 EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()) 90 EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
91 .Times(1); 91 .Times(1);
92 EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
93 .Times(1);
94 EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
95 .Times(1);
92 return channel_proxy_; 96 return channel_proxy_;
93 })); 97 }));
94 stream_config_.voe_channel_id = kChannelId; 98 stream_config_.voe_channel_id = kChannelId;
95 stream_config_.rtp.ssrc = kSsrc; 99 stream_config_.rtp.ssrc = kSsrc;
96 stream_config_.rtp.c_name = kCName; 100 stream_config_.rtp.c_name = kCName;
97 stream_config_.rtp.extensions.push_back( 101 stream_config_.rtp.extensions.push_back(
98 RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId)); 102 RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
99 stream_config_.rtp.extensions.push_back( 103 stream_config_.rtp.extensions.push_back(
100 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); 104 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
101 stream_config_.rtp.extensions.push_back(RtpExtension( 105 stream_config_.rtp.extensions.push_back(RtpExtension(
(...skipping 132 matching lines...) Expand 10 before | Expand all | Expand 10 after
234 static_cast<internal::AudioState*>(helper.audio_state().get()); 238 static_cast<internal::AudioState*>(helper.audio_state().get());
235 VoiceEngineObserver* voe_observer = 239 VoiceEngineObserver* voe_observer =
236 static_cast<VoiceEngineObserver*>(internal_audio_state); 240 static_cast<VoiceEngineObserver*>(internal_audio_state);
237 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING); 241 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
238 EXPECT_TRUE(send_stream.GetStats().typing_noise_detected); 242 EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
239 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING); 243 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
240 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); 244 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
241 } 245 }
242 } // namespace test 246 } // namespace test
243 } // namespace webrtc 247 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/audio/audio_send_stream.cc ('k') | webrtc/audio_receive_stream.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698