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Diffstat (limited to 'chromium/media/cast/transport/cast_transport_sender.h')
-rw-r--r-- | chromium/media/cast/transport/cast_transport_sender.h | 113 |
1 files changed, 113 insertions, 0 deletions
diff --git a/chromium/media/cast/transport/cast_transport_sender.h b/chromium/media/cast/transport/cast_transport_sender.h new file mode 100644 index 00000000000..e88f2f4f098 --- /dev/null +++ b/chromium/media/cast/transport/cast_transport_sender.h @@ -0,0 +1,113 @@ +// 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 CastTransportAudioConfig& config) = 0; + + virtual void InitializeVideo(const CastTransportVideoConfig& 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_ |