Cast:Transport: Dividing A/V Initialization pipeline

media/cast/transport/cast_transport_sender.h

 // 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. The cast sender
 // handles the cast pipeline from encoded frames (both audio and video), to
 // encryption, packetization and transport.
-// All configurations are done at creation.
 
 // 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.
+// 4. Initialize audio and/or video.
+// Steps 3 & 4 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_
 
@@ skipping 12 matching lines @@
 typedef base::Callback<void(CastTransportStatus status)>
     CastTransportStatusCallback;
 
 typedef base::Callback<void(const RtcpSenderInfo& sender_info,
                             base::TimeTicks time_sent,
                             uint32 rtp_timestamp)> CastTransportRtpStatistics;
 
 // The application should only trigger this class from the transport thread.
 class CastTransportSender : public base::NonThreadSafe {
  public:
-  static CastTransportSender* CreateCastTransportSender(
+  static scoped_ptr<CastTransportSender> Create(
       base::TickClock* clock,
-      const CastTransportConfig& config,
+      const net::IPEndPoint& local_end_point,
+      const net::IPEndPoint& remote_end_point,
       const CastTransportStatusCallback& status_callback,
       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner);
 
   virtual ~CastTransportSender() {}
 
+  // Initialize audio pipeline. Encoded audio frames cannot be transmitted until
+  // this method is called.
+  virtual void InitializeAudio(const CastTransportAudioConfig& config) = 0;

[hubbe, 2014/02/21 23:41:23]

Can we have these two functions take a CastTransportRtpStatisticsCallback and
get rid of the two Subscribe* functions below?

[mikhal1, 2014/02/24 17:02:50]

Sorry, but no.
These callbacks are called from the cast audio and video senders, which get the
initialized transport as an input. Perhaps this can be revisited, but not part
of this cl.
On 2014/02/21 23:41:23, hubbe wrote:

[hubbe, 2014/02/25 00:37:04]

Ugh, this is soooo messy, but I don't have a really good answer for how to make
it better either. :(

+
+  // Initialize video pipeline. Encoded video frames cannot be transmitted until
+  // this method is called.
+  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 EncodedAudioFrame* audio_frame,
                                      const base::TimeTicks& recorded_time) = 0;
@@ skipping 10 matching lines @@
                                      const std::string& c_name) = 0;
 
   // Retransmission request.
   virtual void ResendPackets(
       bool is_audio,
       const MissingFramesAndPacketsMap& missing_packets) = 0;
 
   // Audio/Video RTP statistics.
   // RTP statistics will be returned on a regular interval on the designated
   // callback.
+  // Must be called following initialization (InitializeAudio/InitializeVideo).
   virtual void SubscribeAudioRtpStatsCallback(
       const CastTransportRtpStatistics& callback) = 0;
 
   virtual void SubscribeVideoRtpStatsCallback(
       const CastTransportRtpStatistics& callback) = 0;
 };
 
 }  // namespace transport
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_SENDER_H_