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
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
new file mode 100644
index 0000000..40145e4
--- /dev/null
+++ b/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_