// Copyright 2014 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).

// Destruction: The CastTransportSender is assumed to be valid as long as the
// CastSender is alive. Therefore the CastSender should be destructed before the
// CastTransportSender.

#ifndef MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_
#define MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_

#include <stdint.h>

#include "base/callback.h"
#include "base/memory/scoped_ptr.h"
#include "base/single_thread_task_runner.h"
#include "base/threading/non_thread_safe.h"
#include "base/time/tick_clock.h"
#include "base/values.h"
#include "media/cast/logging/logging_defines.h"
#include "media/cast/net/cast_transport_config.h"
#include "media/cast/net/cast_transport_defines.h"
#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
#include "media/cast/net/rtcp/rtcp_defines.h"
#include "net/base/ip_endpoint.h"

namespace base {
class DictionaryValue;
}  // namespace base

namespace net {
class NetLog;
}  // namespace net

namespace media {
namespace cast {

struct RtpReceiverStatistics;
struct RtcpTimeData;

// 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(scoped_ptr<std::vector<FrameEvent>>,
                            scoped_ptr<std::vector<PacketEvent>>)>
    BulkRawEventsCallback;

// TODO(xjz): Rename CastTransportSender as it also deals with receiving
// packets. http://crbug.com/589157.
// The application should only trigger this class from the transport thread.
class CastTransportSender : public base::NonThreadSafe {
 public:
  // Interface used for receiving status updates, raw events, and RTP packets
  // from CastTransportSender.
  class Client {
   public:
    virtual ~Client(){};

    // Audio and Video transport status change is reported on this callback.
    virtual void OnStatusChanged(CastTransportStatus status) = 0;

    // Raw events will be invoked on this callback periodically, according to
    // the configured logging flush interval passed to
    // CastTransportSender::Create().
    virtual void OnLoggingEventsReceived(
        scoped_ptr<std::vector<FrameEvent>> frame_events,
        scoped_ptr<std::vector<PacketEvent>> packet_events) = 0;

    // Called to pass RTP packets to the Client.
    virtual void ProcessRtpPacket(scoped_ptr<Packet> packet) = 0;
  };

  static scoped_ptr<CastTransportSender> Create(
      base::TickClock* clock,  // Owned by the caller.
      base::TimeDelta logging_flush_interval,
      scoped_ptr<Client> client,
      scoped_ptr<PacketSender> transport,
      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.
  virtual void InitializeAudio(const CastTransportRtpConfig& config,
                               const RtcpCastMessageCallback& cast_message_cb,
                               const RtcpRttCallback& rtt_cb) = 0;
  virtual void InitializeVideo(const CastTransportRtpConfig& config,
                               const RtcpCastMessageCallback& cast_message_cb,
                               const RtcpRttCallback& rtt_cb) = 0;

  // Encrypt, packetize and transmit |frame|. |ssrc| must refer to a
  // a channel already established with InitializeAudio / InitializeVideo.
  virtual void InsertFrame(uint32_t ssrc, const EncodedFrame& frame) = 0;

  // Sends a RTCP sender report to the receiver.
  // |ssrc| is the SSRC for this report.
  // |current_time| is the current time reported by a tick clock.
  // |current_time_as_rtp_timestamp| is the corresponding RTP timestamp.
  virtual void SendSenderReport(uint32_t ssrc,
                                base::TimeTicks current_time,
                                RtpTimeTicks current_time_as_rtp_timestamp) = 0;

  // Cancels sending packets for the frames in the set.
  // |ssrc| is the SSRC for the stream.
  // |frame_ids| contains the IDs of the frames that will be cancelled.
  virtual void CancelSendingFrames(uint32_t ssrc,
                                   const std::vector<uint32_t>& frame_ids) = 0;

  // Resends a frame or part of a frame to kickstart. This is used when the
  // stream appears to be stalled.
  virtual void ResendFrameForKickstart(uint32_t ssrc, uint32_t frame_id) = 0;

  // Returns a callback for receiving packets for testing purposes.
  virtual PacketReceiverCallback PacketReceiverForTesting();

  // The following functions are needed for receving.

  // Add a valid SSRC. This is used to verify that incoming packets
  // come from the right sender. Without valid SSRCs, the return address cannot
  // be automatically established.
  virtual void AddValidSsrc(uint32_t ssrc) = 0;

  // Send an RTCP message from receiver to sender.
  virtual void SendRtcpFromRtpReceiver(
      uint32_t ssrc,
      uint32_t sender_ssrc,
      const RtcpTimeData& time_data,
      const RtcpCastMessage* cast_message,
      base::TimeDelta target_delay,
      const ReceiverRtcpEventSubscriber::RtcpEvents* rtcp_events,
      const RtpReceiverStatistics* rtp_receiver_statistics) = 0;

  // Set options for the PacedSender and Wifi.
  virtual void SetOptions(const base::DictionaryValue& options) = 0;
};

}  // namespace cast
}  // namespace media

#endif  // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_