Remove interface directories kept to avoid breaking downstream.

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

Index: webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
deleted file mode 100644
index ac3954ba9be003c14cc0f3a929a667bc3abfac63..0000000000000000000000000000000000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ /dev/null
@@ -1,442 +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_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include <stddef.h>
-#include <list>
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/typedefs.h"
-
-#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
-#define IP_PACKET_SIZE 1500    // we assume ethernet
-#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
-#define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
-
-namespace webrtc {
-namespace rtcp {
-class TransportFeedback;
-}
-
-const int kVideoPayloadTypeFrequency = 90000;
-
-// Minimum RTP header size in bytes.
-const uint8_t kRtpHeaderSize = 12;
-
-struct AudioPayload
-{
-    uint32_t    frequency;
-    uint8_t     channels;
-    uint32_t    rate;
-};
-
-struct VideoPayload
-{
-    RtpVideoCodecTypes   videoCodecType;
-    uint32_t       maxRate;
-};
-
-union PayloadUnion
-{
-    AudioPayload Audio;
-    VideoPayload Video;
-};
-
-enum RTPAliveType
-{
-    kRtpDead   = 0,
-    kRtpNoRtp = 1,
-    kRtpAlive  = 2
-};
-
-enum ProtectionType {
-  kUnprotectedPacket,
-  kProtectedPacket
-};
-
-enum StorageType {
-  kDontRetransmit,
-  kAllowRetransmission
-};
-
-enum RTPExtensionType {
-  kRtpExtensionNone,
-  kRtpExtensionTransmissionTimeOffset,
-  kRtpExtensionAudioLevel,
-  kRtpExtensionAbsoluteSendTime,
-  kRtpExtensionVideoRotation,
-  kRtpExtensionTransportSequenceNumber,
-};
-
-enum RTCPAppSubTypes
-{
-    kAppSubtypeBwe     = 0x00
-};
-
-// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
-enum RTCPPacketType : uint32_t {
-  kRtcpReport = 0x0001,
-  kRtcpSr = 0x0002,
-  kRtcpRr = 0x0004,
-  kRtcpSdes = 0x0008,
-  kRtcpBye = 0x0010,
-  kRtcpPli = 0x0020,
-  kRtcpNack = 0x0040,
-  kRtcpFir = 0x0080,
-  kRtcpTmmbr = 0x0100,
-  kRtcpTmmbn = 0x0200,
-  kRtcpSrReq = 0x0400,
-  kRtcpXrVoipMetric = 0x0800,
-  kRtcpApp = 0x1000,
-  kRtcpSli = 0x4000,
-  kRtcpRpsi = 0x8000,
-  kRtcpRemb = 0x10000,
-  kRtcpTransmissionTimeOffset = 0x20000,
-  kRtcpXrReceiverReferenceTime = 0x40000,
-  kRtcpXrDlrrReportBlock = 0x80000,
-  kRtcpTransportFeedback = 0x100000,
-};
-
-enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
-
-enum RtpRtcpPacketType
-{
-    kPacketRtp        = 0,
-    kPacketKeepAlive = 1
-};
-
-enum NACKMethod
-{
-    kNackOff      = 0,
-    kNackRtcp     = 2
-};
-
-enum RetransmissionMode : uint8_t {
-  kRetransmitOff = 0x0,
-  kRetransmitFECPackets = 0x1,
-  kRetransmitBaseLayer = 0x2,
-  kRetransmitHigherLayers = 0x4,
-  kRetransmitAllPackets = 0xFF
-};
-
-enum RtxMode {
-  kRtxOff                 = 0x0,
-  kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
-  kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
-                                  // instead of padding.
-};
-
-const size_t kRtxHeaderSize = 2;
-
-struct RTCPSenderInfo
-{
-    uint32_t NTPseconds;
-    uint32_t NTPfraction;
-    uint32_t RTPtimeStamp;
-    uint32_t sendPacketCount;
-    uint32_t sendOctetCount;
-};
-
-struct RTCPReportBlock {
-  RTCPReportBlock()
-      : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
-        extendedHighSeqNum(0), jitter(0), lastSR(0),
-        delaySinceLastSR(0) {}
-
-  RTCPReportBlock(uint32_t remote_ssrc,
-                  uint32_t source_ssrc,
-                  uint8_t fraction_lost,
-                  uint32_t cumulative_lost,
-                  uint32_t extended_high_sequence_number,
-                  uint32_t jitter,
-                  uint32_t last_sender_report,
-                  uint32_t delay_since_last_sender_report)
-      : remoteSSRC(remote_ssrc),
-        sourceSSRC(source_ssrc),
-        fractionLost(fraction_lost),
-        cumulativeLost(cumulative_lost),
-        extendedHighSeqNum(extended_high_sequence_number),
-        jitter(jitter),
-        lastSR(last_sender_report),
-        delaySinceLastSR(delay_since_last_sender_report) {}
-
-  // Fields as described by RFC 3550 6.4.2.
-  uint32_t remoteSSRC;  // SSRC of sender of this report.
-  uint32_t sourceSSRC;  // SSRC of the RTP packet sender.
-  uint8_t fractionLost;
-  uint32_t cumulativeLost;  // 24 bits valid.
-  uint32_t extendedHighSeqNum;
-  uint32_t jitter;
-  uint32_t lastSR;
-  uint32_t delaySinceLastSR;
-};
-
-struct RtcpReceiveTimeInfo {
-  // Fields as described by RFC 3611 4.5.
-  uint32_t sourceSSRC;
-  uint32_t lastRR;
-  uint32_t delaySinceLastRR;
-};
-
-typedef std::list<RTCPReportBlock> ReportBlockList;
-
-struct RtpState {
-  RtpState()
-      : sequence_number(0),
-        start_timestamp(0),
-        timestamp(0),
-        capture_time_ms(-1),
-        last_timestamp_time_ms(-1),
-        media_has_been_sent(false) {}
-  uint16_t sequence_number;
-  uint32_t start_timestamp;
-  uint32_t timestamp;
-  int64_t capture_time_ms;
-  int64_t last_timestamp_time_ms;
-  bool media_has_been_sent;
-};
-
-class RtpData
-{
-public:
-    virtual ~RtpData() {}
-
-    virtual int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const WebRtcRTPHeader* rtpHeader) = 0;
-
-    virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                   size_t packet_length) = 0;
-};
-
-class RtpFeedback
-{
-public:
-    virtual ~RtpFeedback() {}
-
-    // Receiving payload change or SSRC change. (return success!)
-    /*
-    *   channels    - number of channels in codec (1 = mono, 2 = stereo)
-    */
-    virtual int32_t OnInitializeDecoder(
-        const int8_t payloadType,
-        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-        const int frequency,
-        const uint8_t channels,
-        const uint32_t rate) = 0;
-
-    virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
-
-    virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
-                                       const bool added) = 0;
-};
-
-class RtpAudioFeedback {
- public:
-  virtual void OnPlayTelephoneEvent(const uint8_t event,
-                                    const uint16_t lengthMs,
-                                    const uint8_t volume) = 0;
-
- protected:
-  virtual ~RtpAudioFeedback() {}
-};
-
-class RtcpIntraFrameObserver {
- public:
-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
-
-  virtual void OnReceivedSLI(uint32_t ssrc,
-                             uint8_t picture_id) = 0;
-
-  virtual void OnReceivedRPSI(uint32_t ssrc,
-                              uint64_t picture_id) = 0;
-
-  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0;
-
-  virtual ~RtcpIntraFrameObserver() {}
-};
-
-class RtcpBandwidthObserver {
- public:
-  // REMB or TMMBR
-  virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
-
-  virtual void OnReceivedRtcpReceiverReport(
-      const ReportBlockList& report_blocks,
-      int64_t rtt,
-      int64_t now_ms) = 0;
-
-  virtual ~RtcpBandwidthObserver() {}
-};
-
-struct PacketInfo {
-  PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
-      : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {}
-
-  PacketInfo(int64_t arrival_time_ms,
-             int64_t send_time_ms,
-             uint16_t sequence_number,
-             size_t payload_size,
-             bool was_paced)
-      : PacketInfo(-1,
-                   arrival_time_ms,
-                   send_time_ms,
-                   sequence_number,
-                   payload_size,
-                   was_paced) {}
-
-  PacketInfo(int64_t creation_time_ms,
-             int64_t arrival_time_ms,
-             int64_t send_time_ms,
-             uint16_t sequence_number,
-             size_t payload_size,
-             bool was_paced)
-      : creation_time_ms(creation_time_ms),
-        arrival_time_ms(arrival_time_ms),
-        send_time_ms(send_time_ms),
-        sequence_number(sequence_number),
-        payload_size(payload_size),
-        was_paced(was_paced) {}
-
-  // Time corresponding to when this object was created.
-  int64_t creation_time_ms;
-  // Time corresponding to when the packet was received. Timestamped with the
-  // receiver's clock.
-  int64_t arrival_time_ms;
-  // Time corresponding to when the packet was sent, timestamped with the
-  // sender's clock.
-  int64_t send_time_ms;
-  // Packet identifier, incremented with 1 for every packet generated by the
-  // sender.
-  uint16_t sequence_number;
-  // Size of the packet excluding RTP headers.
-  size_t payload_size;
-  // True if the packet was paced out by the pacer.
-  bool was_paced;
-};
-
-class TransportFeedbackObserver {
- public:
-  TransportFeedbackObserver() {}
-  virtual ~TransportFeedbackObserver() {}
-
-  // Note: Transport-wide sequence number as sequence number. Arrival time
-  // must be set to 0.
-  virtual void AddPacket(uint16_t sequence_number,
-                         size_t length,
-                         bool was_paced) = 0;
-
-  virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
-};
-
-class RtcpRttStats {
- public:
-  virtual void OnRttUpdate(int64_t rtt) = 0;
-
-  virtual int64_t LastProcessedRtt() const = 0;
-
-  virtual ~RtcpRttStats() {};
-};
-
-// Null object version of RtpFeedback.
-class NullRtpFeedback : public RtpFeedback {
- public:
-  virtual ~NullRtpFeedback() {}
-
-  int32_t OnInitializeDecoder(const int8_t payloadType,
-                              const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                              const int frequency,
-                              const uint8_t channels,
-                              const uint32_t rate) override {
-    return 0;
-  }
-
-  void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
-  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
-};
-
-// Null object version of RtpData.
-class NullRtpData : public RtpData {
- public:
-  virtual ~NullRtpData() {}
-
-  int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                const size_t payloadSize,
-                                const WebRtcRTPHeader* rtpHeader) override {
-    return 0;
-  }
-
-  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
-    return true;
-  }
-};
-
-// Null object version of RtpAudioFeedback.
-class NullRtpAudioFeedback : public RtpAudioFeedback {
- public:
-  virtual ~NullRtpAudioFeedback() {}
-
-  void OnPlayTelephoneEvent(const uint8_t event,
-                            const uint16_t lengthMs,
-                            const uint8_t volume) override {}
-};
-
-// Statistics about packet loss for a single directional connection. All values
-// are totals since the connection initiated.
-struct RtpPacketLossStats {
-  // The number of packets lost in events where no adjacent packets were also
-  // lost.
-  uint64_t single_packet_loss_count;
-  // The number of events in which more than one adjacent packet was lost.
-  uint64_t multiple_packet_loss_event_count;
-  // The number of packets lost in events where more than one adjacent packet
-  // was lost.
-  uint64_t multiple_packet_loss_packet_count;
-};
-
-class RtpPacketSender {
- public:
-  RtpPacketSender() {}
-  virtual ~RtpPacketSender() {}
-
-  enum Priority {
-    kHighPriority = 0,    // Pass through; will be sent immediately.
-    kNormalPriority = 2,  // Put in back of the line.
-    kLowPriority = 3,     // Put in back of the low priority line.
-  };
-  // Low priority packets are mixed with the normal priority packets
-  // while we are paused.
-
-  // Returns true if we send the packet now, else it will add the packet
-  // information to the queue and call TimeToSendPacket when it's time to send.
-  virtual void InsertPacket(Priority priority,
-                            uint32_t ssrc,
-                            uint16_t sequence_number,
-                            int64_t capture_time_ms,
-                            size_t bytes,
-                            bool retransmission) = 0;
-};
-
-class TransportSequenceNumberAllocator {
- public:
-  TransportSequenceNumberAllocator() {}
-  virtual ~TransportSequenceNumberAllocator() {}
-
-  virtual uint16_t AllocateSequenceNumber() = 0;
-};
-
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_