Remove interface directories kept to avoid breaking downstream.

webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h

Index: webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
deleted file mode 100644
index d004f674987231b34f9334e0bd666f7e4cfbd475..0000000000000000000000000000000000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ /dev/null
@@ -1,643 +0,0 @@
-/*
- *  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_INCLUDE_RTP_RTCP_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include <set>
-#include <vector>
-
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-
-namespace webrtc {
-// Forward declarations.
-class ReceiveStatistics;
-class RemoteBitrateEstimator;
-class RtpReceiver;
-class Transport;
-namespace rtcp {
-class TransportFeedback;
-}
-
-class RtpRtcp : public Module {
- public:
-  struct Configuration {
-    Configuration();
-
-   /*  id                   - Unique identifier of this RTP/RTCP module object
-    *  audio                - True for a audio version of the RTP/RTCP module
-    *                         object false will create a video version
-    *  clock                - The clock to use to read time. If NULL object
-    *                         will be using the system clock.
-    *  incoming_data        - Callback object that will receive the incoming
-    *                         data. May not be NULL; default callback will do
-    *                         nothing.
-    *  incoming_messages    - Callback object that will receive the incoming
-    *                         RTP messages. May not be NULL; default callback
-    *                         will do nothing.
-    *  outgoing_transport   - Transport object that will be called when packets
-    *                         are ready to be sent out on the network
-    *  intra_frame_callback - Called when the receiver request a intra frame.
-    *  bandwidth_callback   - Called when we receive a changed estimate from
-    *                         the receiver of out stream.
-    *  audio_messages       - Telephone events. May not be NULL; default
-    *                         callback will do nothing.
-    *  remote_bitrate_estimator - Estimates the bandwidth available for a set of
-    *                             streams from the same client.
-    *  paced_sender             - Spread any bursts of packets into smaller
-    *                             bursts to minimize packet loss.
-    */
-    bool audio;
-    bool receiver_only;
-    Clock* clock;
-    ReceiveStatistics* receive_statistics;
-    Transport* outgoing_transport;
-    RtcpIntraFrameObserver* intra_frame_callback;
-    RtcpBandwidthObserver* bandwidth_callback;
-    TransportFeedbackObserver* transport_feedback_callback;
-    RtcpRttStats* rtt_stats;
-    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
-    RtpAudioFeedback* audio_messages;
-    RemoteBitrateEstimator* remote_bitrate_estimator;
-    RtpPacketSender* paced_sender;
-    TransportSequenceNumberAllocator* transport_sequence_number_allocator;
-    BitrateStatisticsObserver* send_bitrate_observer;
-    FrameCountObserver* send_frame_count_observer;
-    SendSideDelayObserver* send_side_delay_observer;
-  };
-
-  /*
-   *   Create a RTP/RTCP module object using the system clock.
-   *
-   *   configuration  - Configuration of the RTP/RTCP module.
-   */
-  static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
-
-  /**************************************************************************
-   *
-   *   Receiver functions
-   *
-   ***************************************************************************/
-
-    virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
-                                       size_t incoming_packet_length) = 0;
-
-    virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
-
-    /**************************************************************************
-    *
-    *   Sender
-    *
-    ***************************************************************************/
-
-    /*
-    *   set MTU
-    *
-    *   size    -  Max transfer unit in bytes, default is 1500
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetMaxTransferUnit(uint16_t size) = 0;
-
-    /*
-    *   set transtport overhead
-    *   default is IPv4 and UDP with no encryption
-    *
-    *   TCP                     - true for TCP false UDP
-    *   IPv6                    - true for IP version 6 false for version 4
-    *   authenticationOverhead  - number of bytes to leave for an
-    *                             authentication header
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetTransportOverhead(
-        bool TCP,
-        bool IPV6,
-        uint8_t authenticationOverhead = 0) = 0;
-
-    /*
-    *   Get max payload length
-    *
-    *   A combination of the configuration MaxTransferUnit and
-    *   TransportOverhead.
-    *   Does not account FEC/ULP/RED overhead if FEC is enabled.
-    *   Does not account for RTP headers
-    */
-    virtual uint16_t MaxPayloadLength() const = 0;
-
-    /*
-    *   Get max data payload length
-    *
-    *   A combination of the configuration MaxTransferUnit, headers and
-    *   TransportOverhead.
-    *   Takes into account FEC/ULP/RED overhead if FEC is enabled.
-    *   Takes into account RTP headers
-    */
-    virtual uint16_t MaxDataPayloadLength() const = 0;
-
-    /*
-    *   set codec name and payload type
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RegisterSendPayload(
-        const CodecInst& voiceCodec) = 0;
-
-    /*
-    *   set codec name and payload type
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RegisterSendPayload(
-        const VideoCodec& videoCodec) = 0;
-
-    /*
-    *   Unregister a send payload
-    *
-    *   payloadType - payload type of codec
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
-
-   /*
-    *   (De)register RTP header extension type and id.
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
-                                                   uint8_t id) = 0;
-
-    virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
-
-    /*
-    *   get start timestamp
-    */
-    virtual uint32_t StartTimestamp() const = 0;
-
-    /*
-    *   configure start timestamp, default is a random number
-    *
-    *   timestamp   - start timestamp
-    */
-    virtual void SetStartTimestamp(uint32_t timestamp) = 0;
-
-    /*
-    *   Get SequenceNumber
-    */
-    virtual uint16_t SequenceNumber() const = 0;
-
-    /*
-    *   Set SequenceNumber, default is a random number
-    */
-    virtual void SetSequenceNumber(uint16_t seq) = 0;
-
-    // Returns true if the ssrc matched this module, false otherwise.
-    virtual bool SetRtpStateForSsrc(uint32_t ssrc,
-                                    const RtpState& rtp_state) = 0;
-    virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
-
-    /*
-    *   Get SSRC
-    */
-    virtual uint32_t SSRC() const = 0;
-
-    /*
-    *   configure SSRC, default is a random number
-    */
-    virtual void SetSSRC(uint32_t ssrc) = 0;
-
-    /*
-    *   Set CSRC
-    *
-    *   csrcs   - vector of CSRCs
-    */
-    virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
-
-    /*
-    * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination
-    * of values of the enumerator RtxMode.
-    */
-    virtual void SetRtxSendStatus(int modes) = 0;
-
-    /*
-    * Get status of sending RTX (RFC 4588). The returned value can be
-    * a combination of values of the enumerator RtxMode.
-    */
-    virtual int RtxSendStatus() const = 0;
-
-    // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
-    // only the SSRC is set.
-    virtual void SetRtxSsrc(uint32_t ssrc) = 0;
-
-    // Sets the payload type to use when sending RTX packets. Note that this
-    // doesn't enable RTX, only the payload type is set.
-    virtual void SetRtxSendPayloadType(int payload_type,
-                                       int associated_payload_type) = 0;
-
-    // Gets the payload type pair of (RTX, associated) to use when sending RTX
-    // packets.
-    virtual std::pair<int, int> RtxSendPayloadType() const = 0;
-
-    /*
-    *   sends kRtcpByeCode when going from true to false
-    *
-    *   sending - on/off
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetSendingStatus(bool sending) = 0;
-
-    /*
-    *   get send status
-    */
-    virtual bool Sending() const = 0;
-
-    /*
-    *   Starts/Stops media packets, on by default
-    *
-    *   sending - on/off
-    */
-    virtual void SetSendingMediaStatus(bool sending) = 0;
-
-    /*
-    *   get send status
-    */
-    virtual bool SendingMedia() const = 0;
-
-    /*
-    *   get sent bitrate in Kbit/s
-    */
-    virtual void BitrateSent(uint32_t* totalRate,
-                             uint32_t* videoRate,
-                             uint32_t* fecRate,
-                             uint32_t* nackRate) const = 0;
-
-    /*
-    *   Used by the codec module to deliver a video or audio frame for
-    *   packetization.
-    *
-    *   frameType       - type of frame to send
-    *   payloadType     - payload type of frame to send
-    *   timestamp       - timestamp of frame to send
-    *   payloadData     - payload buffer of frame to send
-    *   payloadSize     - size of payload buffer to send
-    *   fragmentation   - fragmentation offset data for fragmented frames such
-    *                     as layers or RED
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendOutgoingData(
-        FrameType frameType,
-        int8_t payloadType,
-        uint32_t timeStamp,
-        int64_t capture_time_ms,
-        const uint8_t* payloadData,
-        size_t payloadSize,
-        const RTPFragmentationHeader* fragmentation = NULL,
-        const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
-
-    virtual bool TimeToSendPacket(uint32_t ssrc,
-                                  uint16_t sequence_number,
-                                  int64_t capture_time_ms,
-                                  bool retransmission) = 0;
-
-    virtual size_t TimeToSendPadding(size_t bytes) = 0;
-
-    // Called on generation of new statistics after an RTP send.
-    virtual void RegisterSendChannelRtpStatisticsCallback(
-        StreamDataCountersCallback* callback) = 0;
-    virtual StreamDataCountersCallback*
-        GetSendChannelRtpStatisticsCallback() const = 0;
-
-    /**************************************************************************
-    *
-    *   RTCP
-    *
-    ***************************************************************************/
-
-    /*
-    *    Get RTCP status
-    */
-    virtual RtcpMode RTCP() const = 0;
-
-    /*
-    *   configure RTCP status i.e on(compound or non- compound)/off
-    *
-    *   method  - RTCP method to use
-    */
-    virtual void SetRTCPStatus(RtcpMode method) = 0;
-
-    /*
-    *   Set RTCP CName (i.e unique identifier)
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetCNAME(const char* c_name) = 0;
-
-    /*
-    *   Get remote CName
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteCNAME(uint32_t remoteSSRC,
-                                char cName[RTCP_CNAME_SIZE]) const = 0;
-
-    /*
-    *   Get remote NTP
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteNTP(
-        uint32_t *ReceivedNTPsecs,
-        uint32_t *ReceivedNTPfrac,
-        uint32_t *RTCPArrivalTimeSecs,
-        uint32_t *RTCPArrivalTimeFrac,
-        uint32_t *rtcp_timestamp) const  = 0;
-
-    /*
-    *   AddMixedCNAME
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
-
-    /*
-    *   RemoveMixedCNAME
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0;
-
-    /*
-    *   Get RoundTripTime
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RTT(uint32_t remoteSSRC,
-                        int64_t* RTT,
-                        int64_t* avgRTT,
-                        int64_t* minRTT,
-                        int64_t* maxRTT) const = 0;
-
-    /*
-    *   Force a send of a RTCP packet
-    *   periodic SR and RR are triggered via the process function
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0;
-
-    /*
-    *   Force a send of a RTCP packet with more than one packet type.
-    *   periodic SR and RR are triggered via the process function
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendCompoundRTCP(
-        const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
-
-    /*
-    *    Good state of RTP receiver inform sender
-    */
-    virtual int32_t SendRTCPReferencePictureSelection(
-        const uint64_t pictureID) = 0;
-
-    /*
-    *    Send a RTCP Slice Loss Indication (SLI)
-    *    6 least significant bits of pictureID
-    */
-    virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
-
-    /*
-    *   Statistics of the amount of data sent
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t DataCountersRTP(
-        size_t* bytesSent,
-        uint32_t* packetsSent) const = 0;
-
-    /*
-    *   Get send statistics for the RTP and RTX stream.
-    */
-    virtual void GetSendStreamDataCounters(
-        StreamDataCounters* rtp_counters,
-        StreamDataCounters* rtx_counters) const = 0;
-
-    /*
-     *  Get packet loss statistics for the RTP stream.
-     */
-    virtual void GetRtpPacketLossStats(
-        bool outgoing,
-        uint32_t ssrc,
-        struct RtpPacketLossStats* loss_stats) const = 0;
-
-    /*
-    *   Get received RTCP sender info
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
-
-    /*
-    *   Get received RTCP report block
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteRTCPStat(
-        std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
-
-    /*
-    *   (APP) Application specific data
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType,
-                                                   uint32_t name,
-                                                   const uint8_t* data,
-                                                   uint16_t length) = 0;
-    /*
-    *   (XR) VOIP metric
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetRTCPVoIPMetrics(
-        const RTCPVoIPMetric* VoIPMetric) = 0;
-
-    /*
-    *   (XR) Receiver Reference Time Report
-    */
-    virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
-
-    virtual bool RtcpXrRrtrStatus() const = 0;
-
-    /*
-    *  (REMB) Receiver Estimated Max Bitrate
-    */
-    virtual bool REMB() const = 0;
-
-    virtual void SetREMBStatus(bool enable) = 0;
-
-    virtual void SetREMBData(uint32_t bitrate,
-                             const std::vector<uint32_t>& ssrcs) = 0;
-
-    /*
-    *   (TMMBR) Temporary Max Media Bit Rate
-    */
-    virtual bool TMMBR() const = 0;
-
-    virtual void SetTMMBRStatus(bool enable) = 0;
-
-    /*
-    *   (NACK)
-    */
-
-    /*
-     *  TODO(holmer): Propagate this API to VideoEngine.
-     *  Returns the currently configured selective retransmission settings.
-     */
-    virtual int SelectiveRetransmissions() const = 0;
-
-    /*
-     *  TODO(holmer): Propagate this API to VideoEngine.
-     *  Sets the selective retransmission settings, which will decide which
-     *  packets will be retransmitted if NACKed. Settings are constructed by
-     *  combining the constants in enum RetransmissionMode with bitwise OR.
-     *  All packets are retransmitted if kRetransmitAllPackets is set, while no
-     *  packets are retransmitted if kRetransmitOff is set.
-     *  By default all packets except FEC packets are retransmitted. For VP8
-     *  with temporal scalability only base layer packets are retransmitted.
-     *
-     *  Returns -1 on failure, otherwise 0.
-     */
-    virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
-
-    /*
-    *   Send a Negative acknowledgement packet
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0;
-
-    /*
-    *   Store the sent packets, needed to answer to a Negative acknowledgement
-    *   requests
-    */
-    virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
-
-    // Returns true if the module is configured to store packets.
-    virtual bool StorePackets() const = 0;
-
-    // Called on receipt of RTCP report block from remote side.
-    virtual void RegisterRtcpStatisticsCallback(
-        RtcpStatisticsCallback* callback) = 0;
-    virtual RtcpStatisticsCallback*
-        GetRtcpStatisticsCallback() = 0;
-    // BWE feedback packets.
-    virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
-
-    /**************************************************************************
-    *
-    *   Audio
-    *
-    ***************************************************************************/
-
-    /*
-    *   set audio packet size, used to determine when it's time to send a DTMF
-    *   packet in silence (CNG)
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0;
-
-    /*
-    *   Send a TelephoneEvent tone using RFC 2833 (4733)
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendTelephoneEventOutband(uint8_t key,
-                                              uint16_t time_ms,
-                                              uint8_t level) = 0;
-
-    /*
-    *   Set payload type for Redundant Audio Data RFC 2198
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0;
-
-    /*
-    *   Get payload type for Redundant Audio Data RFC 2198
-    *
-    *   return -1 on failure else 0
-    */
-     virtual int32_t SendREDPayloadType(
-         int8_t& payloadType) const = 0;
-
-     /*
-     * Store the audio level in dBov for header-extension-for-audio-level-
-     * indication.
-     * This API shall be called before transmision of an RTP packet to ensure
-     * that the |level| part of the extended RTP header is updated.
-     *
-     * return -1 on failure else 0.
-     */
-     virtual int32_t SetAudioLevel(uint8_t level_dBov) = 0;
-
-    /**************************************************************************
-    *
-    *   Video
-    *
-    ***************************************************************************/
-
-    /*
-    *   Set the target send bitrate
-    */
-    virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0;
-
-    /*
-    *   Turn on/off generic FEC
-    */
-    virtual void SetGenericFECStatus(bool enable,
-                                     uint8_t payload_type_red,
-                                     uint8_t payload_type_fec) = 0;
-
-    /*
-    *   Get generic FEC setting
-    */
-    virtual void GenericFECStatus(bool& enable,
-                                     uint8_t& payloadTypeRED,
-                                     uint8_t& payloadTypeFEC) = 0;
-
-
-    virtual int32_t SetFecParameters(
-        const FecProtectionParams* delta_params,
-        const FecProtectionParams* key_params) = 0;
-
-    /*
-    *   Set method for requestion a new key frame
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
-
-    /*
-    *   send a request for a keyframe
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RequestKeyFrame() = 0;
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_