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Side by Side Diff: webrtc/media/engine/fakewebrtccall.h

Issue 2974453002: Protected streams report RTP messages directly to the FlexFec streams (Closed)
Patch Set: Appease lint. Created 3 years, 4 months ago
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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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22 22
23 #include <memory> 23 #include <memory>
24 #include <string> 24 #include <string>
25 #include <vector> 25 #include <vector>
26 26
27 #include "webrtc/api/video/video_frame.h" 27 #include "webrtc/api/video/video_frame.h"
28 #include "webrtc/call/audio_receive_stream.h" 28 #include "webrtc/call/audio_receive_stream.h"
29 #include "webrtc/call/audio_send_stream.h" 29 #include "webrtc/call/audio_send_stream.h"
30 #include "webrtc/call/call.h" 30 #include "webrtc/call/call.h"
31 #include "webrtc/call/flexfec_receive_stream.h" 31 #include "webrtc/call/flexfec_receive_stream.h"
32 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
32 #include "webrtc/rtc_base/buffer.h" 33 #include "webrtc/rtc_base/buffer.h"
33 #include "webrtc/video_receive_stream.h" 34 #include "webrtc/video_receive_stream.h"
34 #include "webrtc/video_send_stream.h" 35 #include "webrtc/video_send_stream.h"
35 36
36 namespace cricket { 37 namespace cricket {
37 class FakeAudioSendStream final : public webrtc::AudioSendStream { 38 class FakeAudioSendStream final : public webrtc::AudioSendStream {
38 public: 39 public:
39 struct TelephoneEvent { 40 struct TelephoneEvent {
40 int payload_type = -1; 41 int payload_type = -1;
41 int payload_frequency = -1; 42 int payload_frequency = -1;
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191 192
192 bool IsReceiving() const; 193 bool IsReceiving() const;
193 194
194 void InjectFrame(const webrtc::VideoFrame& frame); 195 void InjectFrame(const webrtc::VideoFrame& frame);
195 196
196 void SetStats(const webrtc::VideoReceiveStream::Stats& stats); 197 void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
197 198
198 void EnableEncodedFrameRecording(rtc::PlatformFile file, 199 void EnableEncodedFrameRecording(rtc::PlatformFile file,
199 size_t byte_limit) override; 200 size_t byte_limit) override;
200 201
202 void AddSecondarySink(webrtc::RtpPacketSinkInterface* sink) override;
203 void RemoveSecondarySink(const webrtc::RtpPacketSinkInterface* sink) override;
204
201 private: 205 private:
202 // webrtc::VideoReceiveStream implementation. 206 // webrtc::VideoReceiveStream implementation.
203 void Start() override; 207 void Start() override;
204 void Stop() override; 208 void Stop() override;
205 209
206 webrtc::VideoReceiveStream::Stats GetStats() const override; 210 webrtc::VideoReceiveStream::Stats GetStats() const override;
207 211
208 rtc::Optional<webrtc::TimingFrameInfo> GetAndResetTimingFrameInfo() override; 212 rtc::Optional<webrtc::TimingFrameInfo> GetAndResetTimingFrameInfo() override;
209 213
210 webrtc::VideoReceiveStream::Config config_; 214 webrtc::VideoReceiveStream::Config config_;
211 bool receiving_; 215 bool receiving_;
212 webrtc::VideoReceiveStream::Stats stats_; 216 webrtc::VideoReceiveStream::Stats stats_;
213 }; 217 };
214 218
215 class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream { 219 class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
216 public: 220 public:
217 explicit FakeFlexfecReceiveStream( 221 explicit FakeFlexfecReceiveStream(
218 const webrtc::FlexfecReceiveStream::Config& config); 222 const webrtc::FlexfecReceiveStream::Config& config);
219 223
220 const webrtc::FlexfecReceiveStream::Config& GetConfig() const override; 224 const webrtc::FlexfecReceiveStream::Config& GetConfig() const override;
221 225
222 private: 226 private:
223 // webrtc::FlexfecReceiveStream implementation. 227 // webrtc::FlexfecReceiveStream implementation.
224 void Start() override; 228 void Start() override;
225 void Stop() override; 229 void Stop() override;
226 230
227 webrtc::FlexfecReceiveStream::Stats GetStats() const override; 231 webrtc::FlexfecReceiveStream::Stats GetStats() const override;
228 232
233 void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
234
229 webrtc::FlexfecReceiveStream::Config config_; 235 webrtc::FlexfecReceiveStream::Config config_;
230 bool receiving_; 236 bool receiving_;
231 }; 237 };
232 238
233 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 239 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
234 public: 240 public:
235 explicit FakeCall(const webrtc::Call::Config& config); 241 explicit FakeCall(const webrtc::Call::Config& config);
236 ~FakeCall() override; 242 ~FakeCall() override;
237 243
238 webrtc::Call::Config GetConfig() const; 244 webrtc::Call::Config GetConfig() const;
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320 326
321 int num_created_send_streams_; 327 int num_created_send_streams_;
322 int num_created_receive_streams_; 328 int num_created_receive_streams_;
323 329
324 int audio_transport_overhead_; 330 int audio_transport_overhead_;
325 int video_transport_overhead_; 331 int video_transport_overhead_;
326 }; 332 };
327 333
328 } // namespace cricket 334 } // namespace cricket
329 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ 335 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
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