Update talk to 50918584.

Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/interface/rtp_receiver.h

@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class RTPPayloadRegistry;
+
+class TelephoneEventHandler {
+ public:
+  virtual ~TelephoneEventHandler() {}
+
+  // The following three methods implement the TelephoneEventHandler interface.
+  // Forward DTMFs to decoder for playout.
+  virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
+
+  // Is forwarding of outband telephone events turned on/off?
+  virtual bool TelephoneEventForwardToDecoder() const = 0;
+
+  // Is TelephoneEvent configured with payload type payload_type
+  virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
+};
+
+class RtpReceiver {
+ public:
+  // Creates a video-enabled RTP receiver.
+  static RtpReceiver* CreateVideoReceiver(
+      int id, Clock* clock,
+      RtpData* incoming_payload_callback,
+      RtpFeedback* incoming_messages_callback,
+      RTPPayloadRegistry* rtp_payload_registry);
+
+  // Creates an audio-enabled RTP receiver.
+  static RtpReceiver* CreateAudioReceiver(
+      int id, Clock* clock,
+      RtpAudioFeedback* incoming_audio_feedback,
+      RtpData* incoming_payload_callback,
+      RtpFeedback* incoming_messages_callback,
+      RTPPayloadRegistry* rtp_payload_registry);
+
+  virtual ~RtpReceiver() {}
+
+  // Returns a TelephoneEventHandler if available.
+  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
+
+  // Registers a receive payload in the payload registry and notifies the media
+  // receiver strategy.
+  virtual int32_t RegisterReceivePayload(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const int8_t payload_type,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate) = 0;
+
+  // De-registers |payload_type| from the payload registry.
+  virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
+
+  // Parses the media specific parts of an RTP packet and updates the receiver
+  // state. This for instance means that any changes in SSRC and payload type is
+  // detected and acted upon.
+  virtual bool IncomingRtpPacket(RTPHeader* rtp_header,
+                                 const uint8_t* incoming_rtp_packet,
+                                 int incoming_rtp_packet_length,
+                                 PayloadUnion payload_specific,
+                                 bool in_order) = 0;
+
+  // Returns the currently configured NACK method.
+  virtual NACKMethod NACK() const = 0;
+
+  // Turn negative acknowledgement (NACK) requests on/off.
+  virtual int32_t SetNACKStatus(const NACKMethod method,
+                                int max_reordering_threshold) = 0;
+
+  // Returns the last received timestamp.
+  virtual uint32_t Timestamp() const = 0;
+  // Returns the time in milliseconds when the last timestamp was received.
+  virtual int32_t LastReceivedTimeMs() const = 0;
+
+  // Returns the remote SSRC of the currently received RTP stream.
+  virtual uint32_t SSRC() const = 0;
+
+  // Returns the current remote CSRCs.
+  virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
+
+  // Returns the current energy of the RTP stream received.
+  virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
+
+  // Enable/disable RTX and set the SSRC to be used.
+  virtual void SetRTXStatus(bool enable, uint32_t ssrc) = 0;
+
+  // Returns the current RTX status and the SSRC and payload type used.
+  virtual void RTXStatus(bool* enable, uint32_t* ssrc,
+                         int* payload_type) const = 0;
+
+  // Sets the RTX payload type.
+  virtual void SetRtxPayloadType(int payload_type) = 0;
+
+  // Returns true if the packet with RTP header |header| is likely to be a
+  // retransmitted packet, false otherwise.
+  virtual bool RetransmitOfOldPacket(const RTPHeader& header, int jitter,
+                                     int min_rtt) const = 0;
+
+  // Returns true if |sequence_number| is received in order, false otherwise.
+  virtual bool InOrderPacket(const uint16_t sequence_number) const = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_