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Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: No changes to const parameters. Created 5 years, 2 months 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 183 matching lines...) Expand 10 before | Expand all | Expand 10 after
194 } 194 }
195 195
196 void FakeVideoReceiveStream::SetStats( 196 void FakeVideoReceiveStream::SetStats(
197 const webrtc::VideoReceiveStream::Stats& stats) { 197 const webrtc::VideoReceiveStream::Stats& stats) {
198 stats_ = stats; 198 stats_ = stats;
199 } 199 }
200 200
201 FakeCall::FakeCall(const webrtc::Call::Config& config) 201 FakeCall::FakeCall(const webrtc::Call::Config& config)
202 : config_(config), 202 : config_(config),
203 network_state_(webrtc::kNetworkUp), 203 network_state_(webrtc::kNetworkUp),
204 on_packet_sent_called_(false),
204 num_created_send_streams_(0), 205 num_created_send_streams_(0),
205 num_created_receive_streams_(0) { 206 num_created_receive_streams_(0) {}
206 }
207 207
208 FakeCall::~FakeCall() { 208 FakeCall::~FakeCall() {
209 EXPECT_EQ(0u, video_send_streams_.size()); 209 EXPECT_EQ(0u, video_send_streams_.size());
210 EXPECT_EQ(0u, video_receive_streams_.size()); 210 EXPECT_EQ(0u, video_receive_streams_.size());
211 EXPECT_EQ(0u, audio_receive_streams_.size()); 211 EXPECT_EQ(0u, audio_receive_streams_.size());
212 } 212 }
213 213
214 webrtc::Call::Config FakeCall::GetConfig() const { 214 webrtc::Call::Config FakeCall::GetConfig() const {
215 return config_; 215 return config_;
216 } 216 }
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229 229
230 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) { 230 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
231 for (const auto p : GetAudioReceiveStreams()) { 231 for (const auto p : GetAudioReceiveStreams()) {
232 if (p->GetConfig().rtp.remote_ssrc == ssrc) { 232 if (p->GetConfig().rtp.remote_ssrc == ssrc) {
233 return p; 233 return p;
234 } 234 }
235 } 235 }
236 return nullptr; 236 return nullptr;
237 } 237 }
238 238
239 bool FakeCall::on_packet_sent_called() const {
240 return on_packet_sent_called_;
241 }
242
239 webrtc::NetworkState FakeCall::GetNetworkState() const { 243 webrtc::NetworkState FakeCall::GetNetworkState() const {
240 return network_state_; 244 return network_state_;
241 } 245 }
242 246
243 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( 247 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
244 const webrtc::AudioSendStream::Config& config) { 248 const webrtc::AudioSendStream::Config& config) {
245 return nullptr; 249 return nullptr;
246 } 250 }
247 251
248 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { 252 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
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360 } 364 }
361 365
362 void FakeCall::SetBitrateConfig( 366 void FakeCall::SetBitrateConfig(
363 const webrtc::Call::Config::BitrateConfig& bitrate_config) { 367 const webrtc::Call::Config::BitrateConfig& bitrate_config) {
364 config_.bitrate_config = bitrate_config; 368 config_.bitrate_config = bitrate_config;
365 } 369 }
366 370
367 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { 371 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
368 network_state_ = state; 372 network_state_ = state;
369 } 373 }
374
375 void FakeCall::OnPacketSent(const webrtc::SentPacket& sent_packet) {
376 on_packet_sent_called_ = true;
377 }
370 } // namespace cricket 378 } // namespace cricket
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