• Home
  • Line#
  • Scopes#
  • Navigate#
  • Raw
  • Download
1 /*
2  * libjingle
3  * Copyright 2014 Google Inc.
4  *
5  * Redistribution and use in source and binary forms, with or without
6  * modification, are permitted provided that the following conditions are met:
7  *
8  *  1. Redistributions of source code must retain the above copyright notice,
9  *     this list of conditions and the following disclaimer.
10  *  2. Redistributions in binary form must reproduce the above copyright notice,
11  *     this list of conditions and the following disclaimer in the documentation
12  *     and/or other materials provided with the distribution.
13  *  3. The name of the author may not be used to endorse or promote products
14  *     derived from this software without specific prior written permission.
15  *
16  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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,
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
25  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26  */
27 
28 #ifndef TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
29 #define TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
30 
31 #include <map>
32 #include <string>
33 #include <vector>
34 
35 #include "talk/media/base/mediaengine.h"
36 #include "talk/media/webrtc/webrtcvideochannelfactory.h"
37 #include "talk/media/webrtc/webrtcvideodecoderfactory.h"
38 #include "talk/media/webrtc/webrtcvideoencoderfactory.h"
39 #include "webrtc/base/criticalsection.h"
40 #include "webrtc/base/scoped_ptr.h"
41 #include "webrtc/base/thread_annotations.h"
42 #include "webrtc/base/thread_checker.h"
43 #include "webrtc/call.h"
44 #include "webrtc/transport.h"
45 #include "webrtc/video_frame.h"
46 #include "webrtc/video_receive_stream.h"
47 #include "webrtc/video_renderer.h"
48 #include "webrtc/video_send_stream.h"
49 
50 namespace webrtc {
51 class VideoDecoder;
52 class VideoEncoder;
53 }
54 
55 namespace rtc {
56 class Thread;
57 }  // namespace rtc
58 
59 namespace cricket {
60 
61 class VideoCapturer;
62 class VideoFrame;
63 class VideoProcessor;
64 class VideoRenderer;
65 class VoiceMediaChannel;
66 class WebRtcDecoderObserver;
67 class WebRtcEncoderObserver;
68 class WebRtcLocalStreamInfo;
69 class WebRtcRenderAdapter;
70 class WebRtcVideoChannelRecvInfo;
71 class WebRtcVideoChannelSendInfo;
72 class WebRtcVoiceEngine;
73 class WebRtcVoiceMediaChannel;
74 
75 struct CapturedFrame;
76 struct Device;
77 
78 // Exposed here for unittests.
79 std::vector<VideoCodec> DefaultVideoCodecList();
80 
81 class UnsignalledSsrcHandler {
82  public:
83   enum Action {
84     kDropPacket,
85     kDeliverPacket,
86   };
87   virtual Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
88                                    uint32_t ssrc) = 0;
89 };
90 
91 // TODO(pbos): Remove, use external handlers only.
92 class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
93  public:
94   DefaultUnsignalledSsrcHandler();
95   Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
96                            uint32_t ssrc) override;
97 
98   VideoRenderer* GetDefaultRenderer() const;
99   void SetDefaultRenderer(VideoMediaChannel* channel, VideoRenderer* renderer);
100 
101  private:
102   uint32_t default_recv_ssrc_;
103   VideoRenderer* default_renderer_;
104 };
105 
106 // WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
107 class WebRtcVideoEngine2 {
108  public:
109   WebRtcVideoEngine2();
110   ~WebRtcVideoEngine2();
111 
112   // Basic video engine implementation.
113   void Init();
114 
115   WebRtcVideoChannel2* CreateChannel(webrtc::Call* call,
116                                      const VideoOptions& options);
117 
118   const std::vector<VideoCodec>& codecs() const;
119   RtpCapabilities GetCapabilities() const;
120 
121   // Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
122   // not take the ownership of |decoder_factory|. The caller needs to make sure
123   // that |decoder_factory| outlives the video engine.
124   void SetExternalDecoderFactory(WebRtcVideoDecoderFactory* decoder_factory);
125   // Set a WebRtcVideoEncoderFactory for external encoding. Video engine does
126   // not take the ownership of |encoder_factory|. The caller needs to make sure
127   // that |encoder_factory| outlives the video engine.
128   virtual void SetExternalEncoderFactory(
129       WebRtcVideoEncoderFactory* encoder_factory);
130 
131   bool EnableTimedRender();
132 
133   bool FindCodec(const VideoCodec& in);
134   // Check whether the supplied trace should be ignored.
135   bool ShouldIgnoreTrace(const std::string& trace);
136 
137  private:
138   std::vector<VideoCodec> GetSupportedCodecs() const;
139 
140   std::vector<VideoCodec> video_codecs_;
141 
142   bool initialized_;
143 
144   WebRtcVideoDecoderFactory* external_decoder_factory_;
145   WebRtcVideoEncoderFactory* external_encoder_factory_;
146   rtc::scoped_ptr<WebRtcVideoEncoderFactory> simulcast_encoder_factory_;
147 };
148 
149 class WebRtcVideoChannel2 : public rtc::MessageHandler,
150                             public VideoMediaChannel,
151                             public webrtc::Transport,
152                             public webrtc::LoadObserver {
153  public:
154   WebRtcVideoChannel2(webrtc::Call* call,
155                       const VideoOptions& options,
156                       const std::vector<VideoCodec>& recv_codecs,
157                       WebRtcVideoEncoderFactory* external_encoder_factory,
158                       WebRtcVideoDecoderFactory* external_decoder_factory);
159   ~WebRtcVideoChannel2() override;
160 
161   // VideoMediaChannel implementation
162   bool SetSendParameters(const VideoSendParameters& params) override;
163   bool SetRecvParameters(const VideoRecvParameters& params) override;
164   bool GetSendCodec(VideoCodec* send_codec) override;
165   bool SetSendStreamFormat(uint32_t ssrc, const VideoFormat& format) override;
166   bool SetSend(bool send) override;
167   bool SetVideoSend(uint32_t ssrc,
168                     bool mute,
169                     const VideoOptions* options) override;
170   bool AddSendStream(const StreamParams& sp) override;
171   bool RemoveSendStream(uint32_t ssrc) override;
172   bool AddRecvStream(const StreamParams& sp) override;
173   bool AddRecvStream(const StreamParams& sp, bool default_stream);
174   bool RemoveRecvStream(uint32_t ssrc) override;
175   bool SetRenderer(uint32_t ssrc, VideoRenderer* renderer) override;
176   bool GetStats(VideoMediaInfo* info) override;
177   bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) override;
178   bool SendIntraFrame() override;
179   bool RequestIntraFrame() override;
180 
181   void OnPacketReceived(rtc::Buffer* packet,
182                         const rtc::PacketTime& packet_time) override;
183   void OnRtcpReceived(rtc::Buffer* packet,
184                       const rtc::PacketTime& packet_time) override;
185   void OnReadyToSend(bool ready) override;
186   void SetInterface(NetworkInterface* iface) override;
187   void UpdateAspectRatio(int ratio_w, int ratio_h) override;
188 
189   void OnMessage(rtc::Message* msg) override;
190 
191   void OnLoadUpdate(Load load) override;
192 
193   // Implemented for VideoMediaChannelTest.
sending()194   bool sending() const { return sending_; }
GetDefaultSendChannelSsrc()195   uint32_t GetDefaultSendChannelSsrc() { return default_send_ssrc_; }
196   bool GetRenderer(uint32_t ssrc, VideoRenderer** renderer);
197 
198  private:
199   bool MuteStream(uint32_t ssrc, bool mute);
200   class WebRtcVideoReceiveStream;
201 
202   bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
203   bool SetSendRtpHeaderExtensions(
204       const std::vector<RtpHeaderExtension>& extensions);
205   bool SetMaxSendBandwidth(int bps);
206   bool SetOptions(const VideoOptions& options);
207   bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
208   bool SetRecvRtpHeaderExtensions(
209       const std::vector<RtpHeaderExtension>& extensions);
210 
211   void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
212                             const StreamParams& sp) const;
213   bool CodecIsExternallySupported(const std::string& name) const;
214   bool ValidateSendSsrcAvailability(const StreamParams& sp) const
215       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
216   bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
217       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
218   void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
219       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
220 
221   struct VideoCodecSettings {
222     VideoCodecSettings();
223 
224     bool operator==(const VideoCodecSettings& other) const;
225     bool operator!=(const VideoCodecSettings& other) const;
226 
227     VideoCodec codec;
228     webrtc::FecConfig fec;
229     int rtx_payload_type;
230   };
231 
232   static std::string CodecSettingsVectorToString(
233       const std::vector<VideoCodecSettings>& codecs);
234 
235   // Wrapper for the sender part, this is where the capturer is connected and
236   // frames are then converted from cricket frames to webrtc frames.
237   class WebRtcVideoSendStream : public sigslot::has_slots<> {
238    public:
239     WebRtcVideoSendStream(
240         webrtc::Call* call,
241         const StreamParams& sp,
242         const webrtc::VideoSendStream::Config& config,
243         WebRtcVideoEncoderFactory* external_encoder_factory,
244         const VideoOptions& options,
245         int max_bitrate_bps,
246         const rtc::Optional<VideoCodecSettings>& codec_settings,
247         const std::vector<webrtc::RtpExtension>& rtp_extensions,
248         const VideoSendParameters& send_params);
249     ~WebRtcVideoSendStream();
250 
251     void SetOptions(const VideoOptions& options);
252     void SetCodec(const VideoCodecSettings& codec);
253     void SetRtpExtensions(
254         const std::vector<webrtc::RtpExtension>& rtp_extensions);
255     // TODO(deadbeef): Move logic from SetCodec/SetRtpExtensions/etc.
256     // into this method. Currently this method only sets the RTCP mode.
257     void SetSendParameters(const VideoSendParameters& send_params);
258 
259     void InputFrame(VideoCapturer* capturer, const VideoFrame* frame);
260     bool SetCapturer(VideoCapturer* capturer);
261     bool SetVideoFormat(const VideoFormat& format);
262     void MuteStream(bool mute);
263     bool DisconnectCapturer();
264 
265     void SetApplyRotation(bool apply_rotation);
266 
267     void Start();
268     void Stop();
269 
270     const std::vector<uint32_t>& GetSsrcs() const;
271     VideoSenderInfo GetVideoSenderInfo();
272     void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
273 
274     void SetMaxBitrateBps(int max_bitrate_bps);
275 
276    private:
277     // Parameters needed to reconstruct the underlying stream.
278     // webrtc::VideoSendStream doesn't support setting a lot of options on the
279     // fly, so when those need to be changed we tear down and reconstruct with
280     // similar parameters depending on which options changed etc.
281     struct VideoSendStreamParameters {
282       VideoSendStreamParameters(
283           const webrtc::VideoSendStream::Config& config,
284           const VideoOptions& options,
285           int max_bitrate_bps,
286           const rtc::Optional<VideoCodecSettings>& codec_settings);
287       webrtc::VideoSendStream::Config config;
288       VideoOptions options;
289       int max_bitrate_bps;
290       rtc::Optional<VideoCodecSettings> codec_settings;
291       // Sent resolutions + bitrates etc. by the underlying VideoSendStream,
292       // typically changes when setting a new resolution or reconfiguring
293       // bitrates.
294       webrtc::VideoEncoderConfig encoder_config;
295     };
296 
297     struct AllocatedEncoder {
298       AllocatedEncoder(webrtc::VideoEncoder* encoder,
299                        webrtc::VideoCodecType type,
300                        bool external);
301       webrtc::VideoEncoder* encoder;
302       webrtc::VideoEncoder* external_encoder;
303       webrtc::VideoCodecType type;
304       bool external;
305     };
306 
307     struct Dimensions {
308       // Initial encoder configuration (QCIF, 176x144) frame (to ensure that
309       // hardware encoders can be initialized). This gives us low memory usage
310       // but also makes it so configuration errors are discovered at the time we
311       // apply the settings rather than when we get the first frame (waiting for
312       // the first frame to know that you gave a bad codec parameter could make
313       // debugging hard).
314       // TODO(pbos): Consider setting up encoders lazily.
DimensionsDimensions315       Dimensions() : width(176), height(144), is_screencast(false) {}
316       int width;
317       int height;
318       bool is_screencast;
319     };
320 
321     union VideoEncoderSettings {
322       webrtc::VideoCodecVP8 vp8;
323       webrtc::VideoCodecVP9 vp9;
324     };
325 
326     static std::vector<webrtc::VideoStream> CreateVideoStreams(
327         const VideoCodec& codec,
328         const VideoOptions& options,
329         int max_bitrate_bps,
330         size_t num_streams);
331     static std::vector<webrtc::VideoStream> CreateSimulcastVideoStreams(
332         const VideoCodec& codec,
333         const VideoOptions& options,
334         int max_bitrate_bps,
335         size_t num_streams);
336 
337     void* ConfigureVideoEncoderSettings(const VideoCodec& codec,
338                                         const VideoOptions& options,
339                                         bool is_screencast)
340         EXCLUSIVE_LOCKS_REQUIRED(lock_);
341 
342     AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec)
343         EXCLUSIVE_LOCKS_REQUIRED(lock_);
344     void DestroyVideoEncoder(AllocatedEncoder* encoder)
345         EXCLUSIVE_LOCKS_REQUIRED(lock_);
346     void SetCodecAndOptions(const VideoCodecSettings& codec,
347                             const VideoOptions& options)
348         EXCLUSIVE_LOCKS_REQUIRED(lock_);
349     void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_);
350     webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
351         const Dimensions& dimensions,
352         const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_);
353     void SetDimensions(int width, int height, bool is_screencast)
354         EXCLUSIVE_LOCKS_REQUIRED(lock_);
355 
356     const std::vector<uint32_t> ssrcs_;
357     const std::vector<SsrcGroup> ssrc_groups_;
358     webrtc::Call* const call_;
359     WebRtcVideoEncoderFactory* const external_encoder_factory_
360         GUARDED_BY(lock_);
361 
362     rtc::CriticalSection lock_;
363     webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
364     VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
365     VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
366     AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
367     Dimensions last_dimensions_ GUARDED_BY(lock_);
368 
369     VideoCapturer* capturer_ GUARDED_BY(lock_);
370     bool sending_ GUARDED_BY(lock_);
371     bool muted_ GUARDED_BY(lock_);
372     VideoFormat format_ GUARDED_BY(lock_);
373     int old_adapt_changes_ GUARDED_BY(lock_);
374 
375     // The timestamp of the first frame received
376     // Used to generate the timestamps of subsequent frames
377     int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_);
378 
379     // The timestamp of the last frame received
380     // Used to generate timestamp for the black frame when capturer is removed
381     int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
382   };
383 
384   // Wrapper for the receiver part, contains configs etc. that are needed to
385   // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper
386   // between webrtc::VideoRenderer and cricket::VideoRenderer.
387   class WebRtcVideoReceiveStream : public webrtc::VideoRenderer {
388    public:
389     WebRtcVideoReceiveStream(
390         webrtc::Call* call,
391         const StreamParams& sp,
392         const webrtc::VideoReceiveStream::Config& config,
393         WebRtcVideoDecoderFactory* external_decoder_factory,
394         bool default_stream,
395         const std::vector<VideoCodecSettings>& recv_codecs,
396         bool disable_prerenderer_smoothing);
397     ~WebRtcVideoReceiveStream();
398 
399     const std::vector<uint32_t>& GetSsrcs() const;
400 
401     void SetLocalSsrc(uint32_t local_ssrc);
402     void SetFeedbackParameters(bool nack_enabled,
403                                bool remb_enabled,
404                                bool transport_cc_enabled);
405     void SetRecvCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
406     void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
407     // TODO(deadbeef): Move logic from SetRecvCodecs/SetRtpExtensions/etc.
408     // into this method. Currently this method only sets the RTCP mode.
409     void SetRecvParameters(const VideoRecvParameters& recv_params);
410 
411     void RenderFrame(const webrtc::VideoFrame& frame,
412                      int time_to_render_ms) override;
413     bool IsTextureSupported() const override;
414     bool SmoothsRenderedFrames() const override;
415     bool IsDefaultStream() const;
416 
417     void SetRenderer(cricket::VideoRenderer* renderer);
418     cricket::VideoRenderer* GetRenderer();
419 
420     VideoReceiverInfo GetVideoReceiverInfo();
421 
422    private:
423     struct AllocatedDecoder {
424       AllocatedDecoder(webrtc::VideoDecoder* decoder,
425                        webrtc::VideoCodecType type,
426                        bool external);
427       webrtc::VideoDecoder* decoder;
428       // Decoder wrapped into a fallback decoder to permit software fallback.
429       webrtc::VideoDecoder* external_decoder;
430       webrtc::VideoCodecType type;
431       bool external;
432     };
433 
434     void SetSize(int width, int height);
435     void RecreateWebRtcStream();
436 
437     AllocatedDecoder CreateOrReuseVideoDecoder(
438         std::vector<AllocatedDecoder>* old_decoder,
439         const VideoCodec& codec);
440     void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders);
441 
442     std::string GetCodecNameFromPayloadType(int payload_type);
443 
444     webrtc::Call* const call_;
445     const std::vector<uint32_t> ssrcs_;
446     const std::vector<SsrcGroup> ssrc_groups_;
447 
448     webrtc::VideoReceiveStream* stream_;
449     const bool default_stream_;
450     webrtc::VideoReceiveStream::Config config_;
451 
452     WebRtcVideoDecoderFactory* const external_decoder_factory_;
453     std::vector<AllocatedDecoder> allocated_decoders_;
454 
455     const bool disable_prerenderer_smoothing_;
456 
457     rtc::CriticalSection renderer_lock_;
458     cricket::VideoRenderer* renderer_ GUARDED_BY(renderer_lock_);
459     int last_width_ GUARDED_BY(renderer_lock_);
460     int last_height_ GUARDED_BY(renderer_lock_);
461     // Expands remote RTP timestamps to int64_t to be able to estimate how long
462     // the stream has been running.
463     rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_
464         GUARDED_BY(renderer_lock_);
465     int64_t first_frame_timestamp_ GUARDED_BY(renderer_lock_);
466     // Start NTP time is estimated as current remote NTP time (estimated from
467     // RTCP) minus the elapsed time, as soon as remote NTP time is available.
468     int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(renderer_lock_);
469   };
470 
471   void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
472   void SetDefaultOptions();
473 
474   bool SendRtp(const uint8_t* data,
475                size_t len,
476                const webrtc::PacketOptions& options) override;
477   bool SendRtcp(const uint8_t* data, size_t len) override;
478 
479   void StartAllSendStreams();
480   void StopAllSendStreams();
481 
482   static std::vector<VideoCodecSettings> MapCodecs(
483       const std::vector<VideoCodec>& codecs);
484   std::vector<VideoCodecSettings> FilterSupportedCodecs(
485       const std::vector<VideoCodecSettings>& mapped_codecs) const;
486   static bool ReceiveCodecsHaveChanged(std::vector<VideoCodecSettings> before,
487                                        std::vector<VideoCodecSettings> after);
488 
489   void FillSenderStats(VideoMediaInfo* info);
490   void FillReceiverStats(VideoMediaInfo* info);
491   void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
492                                     VideoMediaInfo* info);
493 
494   rtc::ThreadChecker thread_checker_;
495 
496   uint32_t rtcp_receiver_report_ssrc_;
497   bool sending_;
498   webrtc::Call* const call_;
499 
500   uint32_t default_send_ssrc_;
501 
502   DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
503   UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;
504 
505   // Separate list of set capturers used to signal CPU adaptation. These should
506   // not be locked while calling methods that take other locks to prevent
507   // lock-order inversions.
508   rtc::CriticalSection capturer_crit_;
509   bool signal_cpu_adaptation_ GUARDED_BY(capturer_crit_);
510   std::map<uint32_t, VideoCapturer*> capturers_ GUARDED_BY(capturer_crit_);
511 
512   rtc::CriticalSection stream_crit_;
513   // Using primary-ssrc (first ssrc) as key.
514   std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
515       GUARDED_BY(stream_crit_);
516   std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
517       GUARDED_BY(stream_crit_);
518   std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_);
519   std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_);
520 
521   rtc::Optional<VideoCodecSettings> send_codec_;
522   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
523 
524   WebRtcVideoEncoderFactory* const external_encoder_factory_;
525   WebRtcVideoDecoderFactory* const external_decoder_factory_;
526   std::vector<VideoCodecSettings> recv_codecs_;
527   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
528   webrtc::Call::Config::BitrateConfig bitrate_config_;
529   VideoOptions options_;
530   // TODO(deadbeef): Don't duplicate information between
531   // send_params/recv_params, rtp_extensions, options, etc.
532   VideoSendParameters send_params_;
533   VideoRecvParameters recv_params_;
534 };
535 
536 }  // namespace cricket
537 
538 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
539