diff options
| author | Allan Sandfeld Jensen <allan.jensen@qt.io> | 2020-10-12 14:27:29 +0200 |
|---|---|---|
| committer | Allan Sandfeld Jensen <allan.jensen@qt.io> | 2020-10-13 09:35:20 +0000 |
| commit | c30a6232df03e1efbd9f3b226777b07e087a1122 (patch) | |
| tree | e992f45784689f373bcc38d1b79a239ebe17ee23 /chromium/third_party/webrtc/pc/channel.h | |
| parent | 7b5b123ac58f58ffde0f4f6e488bcd09aa4decd3 (diff) | |
| download | qtwebengine-chromium-85-based.tar.gz | |
BASELINE: Update Chromium to 85.0.4183.14085-based
Change-Id: Iaa42f4680837c57725b1344f108c0196741f6057
Reviewed-by: Allan Sandfeld Jensen <allan.jensen@qt.io>
Diffstat (limited to 'chromium/third_party/webrtc/pc/channel.h')
| -rw-r--r-- | chromium/third_party/webrtc/pc/channel.h | 22 |
1 files changed, 7 insertions, 15 deletions
diff --git a/chromium/third_party/webrtc/pc/channel.h b/chromium/third_party/webrtc/pc/channel.h index 238a8e20fee..406058ed4f2 100644 --- a/chromium/third_party/webrtc/pc/channel.h +++ b/chromium/third_party/webrtc/pc/channel.h @@ -22,7 +22,6 @@ #include "api/function_view.h" #include "api/jsep.h" #include "api/rtp_receiver_interface.h" -#include "api/transport/media/media_transport_config.h" #include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "call/rtp_packet_sink_interface.h" @@ -46,7 +45,6 @@ namespace webrtc { class AudioSinkInterface; -class MediaTransportInterface; } // namespace webrtc namespace cricket { @@ -72,7 +70,7 @@ struct CryptoParams; // NetworkInterface. class BaseChannel : public ChannelInterface, - public rtc::MessageHandler, + public rtc::MessageHandlerAutoCleanup, public sigslot::has_slots<>, public MediaChannel::NetworkInterface, public webrtc::RtpPacketSinkInterface { @@ -92,9 +90,7 @@ class BaseChannel : public ChannelInterface, webrtc::CryptoOptions crypto_options, rtc::UniqueRandomIdGenerator* ssrc_generator); virtual ~BaseChannel(); - virtual void Init_w( - webrtc::RtpTransportInternal* rtp_transport, - const webrtc::MediaTransportConfig& media_transport_config); + virtual void Init_w(webrtc::RtpTransportInternal* rtp_transport); // Deinit may be called multiple times and is simply ignored if it's already // done. @@ -275,6 +271,9 @@ class BaseChannel : public ChannelInterface, bool RegisterRtpDemuxerSink(); + // Return description of media channel to facilitate logging + std::string ToString() const; + bool has_received_packet_ = false; private: @@ -296,9 +295,6 @@ class BaseChannel : public ChannelInterface, webrtc::RtpTransportInternal* rtp_transport_ = nullptr; - // Optional media transport configuration (experimental). - webrtc::MediaTransportConfig media_transport_config_; - std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; bool writable_ = false; @@ -350,9 +346,7 @@ class VoiceChannel : public BaseChannel { cricket::MediaType media_type() const override { return cricket::MEDIA_TYPE_AUDIO; } - void Init_w( - webrtc::RtpTransportInternal* rtp_transport, - const webrtc::MediaTransportConfig& media_transport_config) override; + void Init_w(webrtc::RtpTransportInternal* rtp_transport) override; private: // overrides from BaseChannel @@ -432,9 +426,7 @@ class RtpDataChannel : public BaseChannel { DtlsTransportInternal* rtcp_dtls_transport, rtc::PacketTransportInternal* rtp_packet_transport, rtc::PacketTransportInternal* rtcp_packet_transport); - void Init_w( - webrtc::RtpTransportInternal* rtp_transport, - const webrtc::MediaTransportConfig& media_transport_config) override; + void Init_w(webrtc::RtpTransportInternal* rtp_transport) override; virtual bool SendData(const SendDataParams& params, const rtc::CopyOnWriteBuffer& payload, |