Lines Matching refs:webrtc

47       webrtc::TaskQueueFactory* task_queue_factory,
48       webrtc::AudioDeviceModule* adm,
49       const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
50       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
51       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
52       rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
58   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
60       webrtc::Call* call,
63       const webrtc::CryptoOptions& crypto_options) override;
67   std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
80   bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
93   webrtc::TaskQueueFactory* const task_queue_factory_;
96   webrtc::AudioDeviceModule* adm();
97   webrtc::AudioProcessing* apm() const;
98   webrtc::AudioState* audio_state();
101       const std::vector<webrtc::AudioCodecSpec>& specs) const;
107   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
108   rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;
109   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
110   rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer_;
112   rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
114   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
137                                       public webrtc::Transport {
142                           const webrtc::CryptoOptions& crypto_options,
143                           webrtc::Call* call);
150   webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
151   webrtc::RTCError SetRtpSendParameters(
153       const webrtc::RtpParameters& parameters) override;
154   webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
155   webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
174                          rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
180                          rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
204       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
208       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
210   std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
216       rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
220       rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
226                const webrtc::PacketOptions& options) override {  in SendRtp()
274   std::map<int, webrtc::SdpAudioFormat> decoder_map_;
286   webrtc::Call* const call_ = nullptr;
306   std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
314   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
319   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
321   absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
325   const webrtc::AudioCodecPairId codec_pair_id_ =
326       webrtc::AudioCodecPairId::Create();
330   const webrtc::CryptoOptions crypto_options_;
332   rtc::scoped_refptr<webrtc::FrameDecryptorInterface>