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| 1 /* | 1 /* |
| 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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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| 154 bool mute, | 154 bool mute, |
| 155 const VideoOptions* options) override; | 155 const VideoOptions* options) override; |
| 156 bool AddSendStream(const StreamParams& sp) override; | 156 bool AddSendStream(const StreamParams& sp) override; |
| 157 bool RemoveSendStream(uint32_t ssrc) override; | 157 bool RemoveSendStream(uint32_t ssrc) override; |
| 158 bool AddRecvStream(const StreamParams& sp) override; | 158 bool AddRecvStream(const StreamParams& sp) override; |
| 159 bool AddRecvStream(const StreamParams& sp, bool default_stream); | 159 bool AddRecvStream(const StreamParams& sp, bool default_stream); |
| 160 bool RemoveRecvStream(uint32_t ssrc) override; | 160 bool RemoveRecvStream(uint32_t ssrc) override; |
| 161 bool SetSink(uint32_t ssrc, | 161 bool SetSink(uint32_t ssrc, |
| 162 rtc::VideoSinkInterface<VideoFrame>* sink) override; | 162 rtc::VideoSinkInterface<VideoFrame>* sink) override; |
| 163 bool GetStats(VideoMediaInfo* info) override; | 163 bool GetStats(VideoMediaInfo* info) override; |
| 164 bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) override; | 164 void SetSource( |
| 165 uint32_t ssrc, |
| 166 rtc::VideoSourceInterface<cricket::VideoFrame>* source) override; |
| 165 | 167 |
| 166 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, | 168 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
| 167 const rtc::PacketTime& packet_time) override; | 169 const rtc::PacketTime& packet_time) override; |
| 168 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, | 170 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
| 169 const rtc::PacketTime& packet_time) override; | 171 const rtc::PacketTime& packet_time) override; |
| 170 void OnReadyToSend(bool ready) override; | 172 void OnReadyToSend(bool ready) override; |
| 171 void SetInterface(NetworkInterface* iface) override; | 173 void SetInterface(NetworkInterface* iface) override; |
| 172 | 174 |
| 173 // Implemented for VideoMediaChannelTest. | 175 // Implemented for VideoMediaChannelTest. |
| 174 bool sending() const { return sending_; } | 176 bool sending() const { return sending_; } |
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| 200 // These optionals are unset if not changed. | 202 // These optionals are unset if not changed. |
| 201 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings; | 203 rtc::Optional<std::vector<VideoCodecSettings>> codec_settings; |
| 202 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; | 204 rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; |
| 203 }; | 205 }; |
| 204 | 206 |
| 205 bool GetChangedSendParameters(const VideoSendParameters& params, | 207 bool GetChangedSendParameters(const VideoSendParameters& params, |
| 206 ChangedSendParameters* changed_params) const; | 208 ChangedSendParameters* changed_params) const; |
| 207 bool GetChangedRecvParameters(const VideoRecvParameters& params, | 209 bool GetChangedRecvParameters(const VideoRecvParameters& params, |
| 208 ChangedRecvParameters* changed_params) const; | 210 ChangedRecvParameters* changed_params) const; |
| 209 | 211 |
| 210 bool MuteStream(uint32_t ssrc, bool mute); | |
| 211 | |
| 212 void SetMaxSendBandwidth(int bps); | 212 void SetMaxSendBandwidth(int bps); |
| 213 void SetOptions(uint32_t ssrc, const VideoOptions& options); | 213 void SetOptions(uint32_t ssrc, const VideoOptions& options); |
| 214 | 214 |
| 215 void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config, | 215 void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config, |
| 216 const StreamParams& sp) const; | 216 const StreamParams& sp) const; |
| 217 bool CodecIsExternallySupported(const std::string& name) const; | 217 bool CodecIsExternallySupported(const std::string& name) const; |
| 218 bool ValidateSendSsrcAvailability(const StreamParams& sp) const | 218 bool ValidateSendSsrcAvailability(const StreamParams& sp) const |
| 219 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); | 219 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
| 220 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const | 220 bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const |
| 221 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); | 221 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
| 222 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream) | 222 void DeleteReceiveStream(WebRtcVideoReceiveStream* stream) |
| 223 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); | 223 EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
| 224 | 224 |
| 225 static std::string CodecSettingsVectorToString( | 225 static std::string CodecSettingsVectorToString( |
| 226 const std::vector<VideoCodecSettings>& codecs); | 226 const std::vector<VideoCodecSettings>& codecs); |
| 227 | 227 |
| 228 // Wrapper for the sender part, this is where the capturer is connected and | 228 // Wrapper for the sender part, this is where the source is connected and |
| 229 // frames are then converted from cricket frames to webrtc frames. | 229 // frames are then converted from cricket frames to webrtc frames. |
| 230 class WebRtcVideoSendStream | 230 class WebRtcVideoSendStream |
| 231 : public rtc::VideoSinkInterface<cricket::VideoFrame>, | 231 : public rtc::VideoSinkInterface<cricket::VideoFrame>, |
| 232 public webrtc::LoadObserver { | 232 public webrtc::LoadObserver { |
| 233 public: | 233 public: |
| 234 WebRtcVideoSendStream( | 234 WebRtcVideoSendStream( |
| 235 webrtc::Call* call, | 235 webrtc::Call* call, |
| 236 const StreamParams& sp, | 236 const StreamParams& sp, |
| 237 const webrtc::VideoSendStream::Config& config, | 237 const webrtc::VideoSendStream::Config& config, |
| 238 const VideoOptions& options, | 238 const VideoOptions& options, |
| 239 WebRtcVideoEncoderFactory* external_encoder_factory, | 239 WebRtcVideoEncoderFactory* external_encoder_factory, |
| 240 bool enable_cpu_overuse_detection, | 240 bool enable_cpu_overuse_detection, |
| 241 int max_bitrate_bps, | 241 int max_bitrate_bps, |
| 242 const rtc::Optional<VideoCodecSettings>& codec_settings, | 242 const rtc::Optional<VideoCodecSettings>& codec_settings, |
| 243 const std::vector<webrtc::RtpExtension>& rtp_extensions, | 243 const std::vector<webrtc::RtpExtension>& rtp_extensions, |
| 244 const VideoSendParameters& send_params); | 244 const VideoSendParameters& send_params); |
| 245 virtual ~WebRtcVideoSendStream(); | 245 virtual ~WebRtcVideoSendStream(); |
| 246 | 246 |
| 247 void SetOptions(const VideoOptions& options); | 247 void SetOptions(const VideoOptions& options); |
| 248 // TODO(pbos): Move logic from SetOptions into this method. | 248 // TODO(pbos): Move logic from SetOptions into this method. |
| 249 void SetSendParameters(const ChangedSendParameters& send_params); | 249 void SetSendParameters(const ChangedSendParameters& send_params); |
| 250 bool SetRtpParameters(const webrtc::RtpParameters& parameters); | 250 bool SetRtpParameters(const webrtc::RtpParameters& parameters); |
| 251 | 251 |
| 252 void OnFrame(const cricket::VideoFrame& frame) override; | 252 void OnFrame(const cricket::VideoFrame& frame) override; |
| 253 bool SetCapturer(VideoCapturer* capturer); | 253 void SetSource(rtc::VideoSourceInterface<cricket::VideoFrame>* source); |
| 254 void MuteStream(bool mute); | 254 void DisconnectSource(); |
| 255 bool DisconnectCapturer(); | |
| 256 | 255 |
| 257 void Start(); | 256 void Start(); |
| 258 void Stop(); | 257 void Stop(); |
| 259 | 258 |
| 260 webrtc::RtpParameters rtp_parameters() const { return rtp_parameters_; } | 259 webrtc::RtpParameters rtp_parameters() const { return rtp_parameters_; } |
| 261 | 260 |
| 262 // Implements webrtc::LoadObserver. | 261 // Implements webrtc::LoadObserver. |
| 263 void OnLoadUpdate(Load load) override; | 262 void OnLoadUpdate(Load load) override; |
| 264 | 263 |
| 265 const std::vector<uint32_t>& GetSsrcs() const; | 264 const std::vector<uint32_t>& GetSsrcs() const; |
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| 346 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); | 345 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); |
| 347 | 346 |
| 348 rtc::ThreadChecker thread_checker_; | 347 rtc::ThreadChecker thread_checker_; |
| 349 rtc::AsyncInvoker invoker_; | 348 rtc::AsyncInvoker invoker_; |
| 350 rtc::Thread* worker_thread_; | 349 rtc::Thread* worker_thread_; |
| 351 const std::vector<uint32_t> ssrcs_; | 350 const std::vector<uint32_t> ssrcs_; |
| 352 const std::vector<SsrcGroup> ssrc_groups_; | 351 const std::vector<SsrcGroup> ssrc_groups_; |
| 353 webrtc::Call* const call_; | 352 webrtc::Call* const call_; |
| 354 rtc::VideoSinkWants sink_wants_; | 353 rtc::VideoSinkWants sink_wants_; |
| 355 // Counter used for deciding if the video resolution is currently | 354 // Counter used for deciding if the video resolution is currently |
| 356 // restricted by CPU usage. It is reset if |capturer_| is changed. | 355 // restricted by CPU usage. It is reset if |source_| is changed. |
| 357 int cpu_restricted_counter_; | 356 int cpu_restricted_counter_; |
| 358 // Total number of times resolution as been requested to be changed due to | 357 // Total number of times resolution as been requested to be changed due to |
| 359 // CPU adaptation. | 358 // CPU adaptation. |
| 360 int number_of_cpu_adapt_changes_; | 359 int number_of_cpu_adapt_changes_; |
| 361 VideoCapturer* capturer_; | 360 rtc::VideoSourceInterface<cricket::VideoFrame>* source_; |
| 362 WebRtcVideoEncoderFactory* const external_encoder_factory_ | 361 WebRtcVideoEncoderFactory* const external_encoder_factory_ |
| 363 GUARDED_BY(lock_); | 362 GUARDED_BY(lock_); |
| 364 | 363 |
| 365 rtc::CriticalSection lock_; | 364 rtc::CriticalSection lock_; |
| 366 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); | 365 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); |
| 367 // Contains settings that are the same for all streams in the MediaChannel, | 366 // Contains settings that are the same for all streams in the MediaChannel, |
| 368 // such as codecs, header extensions, and the global bitrate limit for the | 367 // such as codecs, header extensions, and the global bitrate limit for the |
| 369 // entire channel. | 368 // entire channel. |
| 370 VideoSendStreamParameters parameters_ GUARDED_BY(lock_); | 369 VideoSendStreamParameters parameters_ GUARDED_BY(lock_); |
| 371 // Contains settings that are unique for each stream, such as max_bitrate. | 370 // Contains settings that are unique for each stream, such as max_bitrate. |
| 372 // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_. | 371 // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_. |
| 373 // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only | 372 // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only |
| 374 // one stream per MediaChannel. | 373 // one stream per MediaChannel. |
| 375 webrtc::RtpParameters rtp_parameters_; | 374 webrtc::RtpParameters rtp_parameters_; |
| 376 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); | 375 bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); |
| 377 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); | 376 VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); |
| 378 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); | 377 AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); |
| 378 // Original frame dimension, before adaptation due to cpu or |
| 379 // network bottlenecks. |
| 380 Dimensions input_dimensions_ GUARDED_BY(lock_); |
| 379 Dimensions last_dimensions_ GUARDED_BY(lock_); | 381 Dimensions last_dimensions_ GUARDED_BY(lock_); |
| 380 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = | 382 webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = |
| 381 webrtc::kVideoRotation_0; | 383 webrtc::kVideoRotation_0; |
| 382 | 384 |
| 383 bool sending_ GUARDED_BY(lock_); | 385 bool sending_ GUARDED_BY(lock_); |
| 384 bool muted_ GUARDED_BY(lock_); | |
| 385 | 386 |
| 386 // The timestamp of the first frame received | 387 // The timestamp of the first frame received |
| 387 // Used to generate the timestamps of subsequent frames | 388 // Used to generate the timestamps of subsequent frames |
| 388 int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_); | 389 int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_); |
| 389 | 390 |
| 390 // The timestamp of the last frame received | 391 // The timestamp of the last frame received |
| 391 // Used to generate timestamp for the black frame when capturer is removed | 392 // Used to generate timestamp for the black frame when source is removed |
| 392 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_); | 393 int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_); |
| 393 }; | 394 }; |
| 394 | 395 |
| 395 // Wrapper for the receiver part, contains configs etc. that are needed to | 396 // Wrapper for the receiver part, contains configs etc. that are needed to |
| 396 // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper | 397 // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper |
| 397 // between webrtc::VideoRenderer and cricket::VideoRenderer. | 398 // between webrtc::VideoRenderer and cricket::VideoRenderer. |
| 398 class WebRtcVideoReceiveStream : public webrtc::VideoRenderer { | 399 class WebRtcVideoReceiveStream : public webrtc::VideoRenderer { |
| 399 public: | 400 public: |
| 400 WebRtcVideoReceiveStream( | 401 WebRtcVideoReceiveStream( |
| 401 webrtc::Call* call, | 402 webrtc::Call* call, |
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| 530 // TODO(deadbeef): Don't duplicate information between | 531 // TODO(deadbeef): Don't duplicate information between |
| 531 // send_params/recv_params, rtp_extensions, options, etc. | 532 // send_params/recv_params, rtp_extensions, options, etc. |
| 532 VideoSendParameters send_params_; | 533 VideoSendParameters send_params_; |
| 533 VideoOptions default_send_options_; | 534 VideoOptions default_send_options_; |
| 534 VideoRecvParameters recv_params_; | 535 VideoRecvParameters recv_params_; |
| 535 }; | 536 }; |
| 536 | 537 |
| 537 } // namespace cricket | 538 } // namespace cricket |
| 538 | 539 |
| 539 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ | 540 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ |
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