Index: media/cast/net/cast_transport_impl.h |
diff --git a/media/cast/net/cast_transport_impl.h b/media/cast/net/cast_transport_impl.h |
index 5dff2b51a8809af664acbf1e02fb4bdef91ca14b..98462fa8fa7ccf02b2960fe20c75491b5877c6bb 100644 |
--- a/media/cast/net/cast_transport_impl.h |
+++ b/media/cast/net/cast_transport_impl.h |
@@ -65,10 +65,8 @@ class CastTransportImpl final : public CastTransport { |
~CastTransportImpl() final; |
// CastTransport implementation for sending. |
- void InitializeAudio(const CastTransportRtpConfig& config, |
- std::unique_ptr<RtcpObserver> rtcp_observer) final; |
- void InitializeVideo(const CastTransportRtpConfig& config, |
- std::unique_ptr<RtcpObserver> rtcp_observer) final; |
+ void InitializeStream(const CastTransportRtpConfig& config, |
+ std::unique_ptr<RtcpObserver> rtcp_observer) final; |
void InsertFrame(uint32_t ssrc, const EncodedFrame& frame) final; |
void SendSenderReport(uint32_t ssrc, |
@@ -116,6 +114,8 @@ class CastTransportImpl final : public CastTransport { |
// Handle received RTCP messages on RTP sender. |
class RtcpClient; |
+ struct RtpStreamSession; |
+ |
FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, NacksCancelRetransmits); |
FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, CancelRetransmits); |
FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, Kickstart); |
@@ -160,25 +160,6 @@ class CastTransportImpl final : public CastTransport { |
// Packet sender that performs pacing. |
PacedSender pacer_; |
- // Packetizer for audio and video frames. |
- std::unique_ptr<RtpSender> audio_sender_; |
- std::unique_ptr<RtpSender> video_sender_; |
- |
- // Maintains RTCP session for audio and video. |
- std::unique_ptr<SenderRtcpSession> audio_rtcp_session_; |
- std::unique_ptr<SenderRtcpSession> video_rtcp_session_; |
- |
- // RTCP observer for SenderRtcpSession. |
- std::unique_ptr<RtcpObserver> audio_rtcp_observer_; |
- std::unique_ptr<RtcpObserver> video_rtcp_observer_; |
- |
- // Encrypts data in EncodedFrames before they are sent. Note that it's |
- // important for the encryption to happen here, in code that would execute in |
- // the main browser process, for security reasons. This helps to mitigate |
- // the damage that could be caused by a compromised renderer process. |
- TransportEncryptionHandler audio_encryptor_; |
- TransportEncryptionHandler video_encryptor_; |
- |
// Right after a frame is sent we record the number of bytes sent to the |
// socket. We record the corresponding bytes sent for the most recent ACKed |
// audio packet. |
@@ -199,6 +180,11 @@ class CastTransportImpl final : public CastTransport { |
std::unique_ptr<RtcpBuilder> rtcp_builder_at_rtp_receiver_; |
+ // Records the initialized stream sessions on RTP sender. The sender SSRC is |
+ // used as key since it is unique for each RTP stream. |
+ using SessionMap = std::map<uint32_t, std::unique_ptr<RtpStreamSession>>; |
+ SessionMap sessions_; |
+ |
base::WeakPtrFactory<CastTransportImpl> weak_factory_; |
DISALLOW_COPY_AND_ASSIGN(CastTransportImpl); |