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      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.
    194   bool sending() const { return sending_; }
    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.
    315       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