diff options
Diffstat (limited to 'talk/session/media/channel.h')
| -rw-r--r-- | talk/session/media/channel.h | 43 |
1 files changed, 25 insertions, 18 deletions
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h index 603115cee7..d8fde670a0 100644 --- a/talk/session/media/channel.h +++ b/talk/session/media/channel.h @@ -38,19 +38,24 @@ #include "talk/media/base/mediaengine.h" #include "talk/media/base/streamparams.h" #include "talk/media/base/videocapturer.h" -#include "webrtc/p2p/base/transportcontroller.h" -#include "webrtc/p2p/client/socketmonitor.h" #include "talk/session/media/audiomonitor.h" #include "talk/session/media/bundlefilter.h" #include "talk/session/media/mediamonitor.h" #include "talk/session/media/mediasession.h" #include "talk/session/media/rtcpmuxfilter.h" #include "talk/session/media/srtpfilter.h" +#include "webrtc/audio/audio_sink.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/network.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/window.h" +#include "webrtc/p2p/base/transportcontroller.h" +#include "webrtc/p2p/client/socketmonitor.h" + +namespace webrtc { +class AudioSinkInterface; +} // namespace webrtc namespace cricket { @@ -174,8 +179,11 @@ class BaseChannel // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is // true). Gets the transport channels from |transport_controller_|. bool SetTransport_w(const std::string& transport_name); + void set_transport_channel(TransportChannel* transport); - void set_rtcp_transport_channel(TransportChannel* transport); + void set_rtcp_transport_channel(TransportChannel* transport, + bool update_writablity); + bool was_ever_writable() const { return was_ever_writable_; } void set_local_content_direction(MediaContentDirection direction) { local_content_direction_ = direction; @@ -213,6 +221,8 @@ class BaseChannel int flags); void OnReadyToSend(TransportChannel* channel); + void OnDtlsState(TransportChannel* channel, DtlsTransportState state); + bool PacketIsRtcp(const TransportChannel* channel, const char* data, size_t len); bool SendPacket(bool rtcp, @@ -235,8 +245,9 @@ class BaseChannel // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters. // |rtcp_channel| indicates whether to set up the RTP or RTCP filter. bool SetupDtlsSrtp(bool rtcp_channel); + void MaybeSetupDtlsSrtp_w(); // Set the DTLS-SRTP cipher policy on this channel as appropriate. - bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp); + bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp); virtual void ChangeState() = 0; @@ -282,9 +293,8 @@ class BaseChannel void OnMessage(rtc::Message* pmsg) override; // Handled in derived classes - // Get the SRTP ciphers to use for RTP media - virtual void GetSrtpCryptoSuiteNames( - std::vector<std::string>* ciphers) const = 0; + // Get the SRTP crypto suites to use for RTP media + virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0; virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) = 0; @@ -356,8 +366,6 @@ class VoiceChannel : public BaseChannel { // own ringing sound sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout; - // TODO(ronghuawu): Replace PressDTMF with InsertDtmf. - bool PressDTMF(int digit, bool playout); // Returns if the telephone-event has been negotiated. bool CanInsertDtmf(); // Send and/or play a DTMF |event| according to the |flags|. @@ -365,8 +373,11 @@ class VoiceChannel : public BaseChannel { // The |ssrc| should be either 0 or a valid send stream ssrc. // The valid value for the |event| are 0 which corresponding to DTMF // event 0-9, *, #, A-D. - bool InsertDtmf(uint32_t ssrc, int event_code, int duration, int flags); + bool InsertDtmf(uint32_t ssrc, int event_code, int duration); bool SetOutputVolume(uint32_t ssrc, double volume); + void SetRawAudioSink(uint32_t ssrc, + rtc::scoped_ptr<webrtc::AudioSinkInterface> sink); + // Get statistics about the current media session. bool GetStats(VoiceMediaInfo* stats); @@ -402,12 +413,12 @@ class VoiceChannel : public BaseChannel { ContentAction action, std::string* error_desc); void HandleEarlyMediaTimeout(); - bool InsertDtmf_w(uint32_t ssrc, int event, int duration, int flags); + bool InsertDtmf_w(uint32_t ssrc, int event, int duration); bool SetOutputVolume_w(uint32_t ssrc, double volume); bool GetStats_w(VoiceMediaInfo* stats); virtual void OnMessage(rtc::Message* pmsg); - virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const; + virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const; virtual void OnConnectionMonitorUpdate( ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos); virtual void OnMediaMonitorUpdate( @@ -456,8 +467,6 @@ class VideoChannel : public BaseChannel { // True if we've added a screencast. Doesn't matter if the capturer // has been started or not. bool IsScreencasting(); - int GetScreencastFps(uint32_t ssrc); - int GetScreencastMaxPixels(uint32_t ssrc); // Get statistics about the current media session. bool GetStats(VideoMediaInfo* stats); @@ -476,7 +485,6 @@ class VideoChannel : public BaseChannel { private: typedef std::map<uint32_t, VideoCapturer*> ScreencastMap; - struct ScreencastDetailsData; // overrides from BaseChannel virtual void ChangeState(); @@ -493,11 +501,10 @@ class VideoChannel : public BaseChannel { bool RemoveScreencast_w(uint32_t ssrc); void OnScreencastWindowEvent_s(uint32_t ssrc, rtc::WindowEvent we); bool IsScreencasting_w() const; - void GetScreencastDetails_w(ScreencastDetailsData* d) const; bool GetStats_w(VideoMediaInfo* stats); virtual void OnMessage(rtc::Message* pmsg); - virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const; + virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const; virtual void OnConnectionMonitorUpdate( ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos); virtual void OnMediaMonitorUpdate( @@ -614,7 +621,7 @@ class DataChannel : public BaseChannel { virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet); virtual void OnMessage(rtc::Message* pmsg); - virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const; + virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const; virtual void OnConnectionMonitorUpdate( ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos); virtual void OnMediaMonitorUpdate( |