Index: media/cast/transport/cast_transport_sender.h |
diff --git a/media/cast/transport/cast_transport_sender.h b/media/cast/transport/cast_transport_sender.h |
deleted file mode 100644 |
index c9ae2fe01b0f7edb9eb69e0d0187ad5364005d1c..0000000000000000000000000000000000000000 |
--- a/media/cast/transport/cast_transport_sender.h |
+++ /dev/null |
@@ -1,112 +0,0 @@ |
-// Copyright 2013 The Chromium Authors. All rights reserved. |
-// Use of this source code is governed by a BSD-style license that can be |
-// found in the LICENSE file. |
- |
-// This is the main interface for the cast transport sender. It accepts encoded |
-// frames (both audio and video), encrypts their encoded data, packetizes them |
-// and feeds them into a transport (e.g., UDP). |
- |
-// Construction of the Cast Sender and the Cast Transport Sender should be done |
-// in the following order: |
-// 1. Create CastTransportSender. |
-// 2. Create CastSender (accepts CastTransportSender as an input). |
-// 3. Call CastTransportSender::SetPacketReceiver to ensure that the packets |
-// received by the CastTransportSender will be sent to the CastSender. |
-// Steps 3 can be done interchangeably. |
- |
-// Destruction: The CastTransportSender is assumed to be valid as long as the |
-// CastSender is alive. Therefore the CastSender should be destructed before the |
-// CastTransportSender. |
-// This also works when the CastSender acts as a receiver for the RTCP packets |
-// due to the weak pointers in the ReceivedPacket method in cast_sender_impl.cc. |
- |
-#ifndef MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_ |
-#define MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_ |
- |
-#include "base/basictypes.h" |
-#include "base/callback.h" |
-#include "base/single_thread_task_runner.h" |
-#include "base/threading/non_thread_safe.h" |
-#include "base/time/tick_clock.h" |
-#include "media/cast/logging/logging_defines.h" |
-#include "media/cast/transport/cast_transport_config.h" |
-#include "media/cast/transport/cast_transport_defines.h" |
- |
-namespace net { |
-class NetLog; |
-} // namespace net |
- |
-namespace media { |
-namespace cast { |
-namespace transport { |
- |
-// Following the initialization of either audio or video an initialization |
-// status will be sent via this callback. |
-typedef base::Callback<void(CastTransportStatus status)> |
- CastTransportStatusCallback; |
- |
-typedef base::Callback<void(const std::vector<PacketEvent>&)> |
- BulkRawEventsCallback; |
- |
-// The application should only trigger this class from the transport thread. |
-class CastTransportSender : public base::NonThreadSafe { |
- public: |
- static scoped_ptr<CastTransportSender> Create( |
- net::NetLog* net_log, |
- base::TickClock* clock, |
- const net::IPEndPoint& remote_end_point, |
- const CastTransportStatusCallback& status_callback, |
- const BulkRawEventsCallback& raw_events_callback, |
- base::TimeDelta raw_events_callback_interval, |
- const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner); |
- |
- virtual ~CastTransportSender() {} |
- |
- // Audio/Video initialization. |
- // Encoded frames cannot be transmitted until the relevant initialize method |
- // is called. Usually called by CastSender. |
- virtual void InitializeAudio(const CastTransportRtpConfig& config) = 0; |
- virtual void InitializeVideo(const CastTransportRtpConfig& config) = 0; |
- |
- // Sets the Cast packet receiver. Should be called after creation on the |
- // Cast sender. Packets won't be received until this function is called. |
- virtual void SetPacketReceiver( |
- const PacketReceiverCallback& packet_receiver) = 0; |
- |
- // The following two functions handle the encoded media frames (audio and |
- // video) to be processed. |
- // Frames will be encrypted, packetized and transmitted to the network. |
- virtual void InsertCodedAudioFrame(const EncodedFrame& audio_frame) = 0; |
- virtual void InsertCodedVideoFrame(const EncodedFrame& video_frame) = 0; |
- |
- // Builds an RTCP packet and sends it to the network. |
- // |ntp_seconds|, |ntp_fraction| and |rtp_timestamp| are used in the |
- // RTCP Sender Report. |
- virtual void SendRtcpFromRtpSender(uint32 packet_type_flags, |
- uint32 ntp_seconds, |
- uint32 ntp_fraction, |
- uint32 rtp_timestamp, |
- const RtcpDlrrReportBlock& dlrr, |
- uint32 sending_ssrc, |
- const std::string& c_name) = 0; |
- |
- // Retransmission request. |
- // |missing_packets| includes the list of frames and packets in each |
- // frame to be re-transmitted. |
- // If |cancel_rtx_if_not_in_list| is used as an optimization to cancel |
- // pending re-transmission requests of packets not listed in |
- // |missing_packets|. If the requested packet(s) were sent recently |
- // (how long is specified by |dedupe_window|) then this re-transmit |
- // will be ignored. |
- virtual void ResendPackets( |
- bool is_audio, |
- const MissingFramesAndPacketsMap& missing_packets, |
- bool cancel_rtx_if_not_in_list, |
- base::TimeDelta dedupe_window) = 0; |
-}; |
- |
-} // namespace transport |
-} // namespace cast |
-} // namespace media |
- |
-#endif // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_ |