Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 132 insertions, 0 deletions

diff --git a/webrtc/video/vie_receiver.h b/webrtc/video/vie_receiver.h
new file mode 100644
index 0000000000..8204888bbb
--- /dev/null
+++ b/webrtc/video/vie_receiver.h
@@ -0,0 +1,132 @@
+/*
+ *  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_VIDEO_VIE_RECEIVER_H_
+#define WEBRTC_VIDEO_VIE_RECEIVER_H_
+
+#include <list>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class FecReceiver;
+class RemoteNtpTimeEstimator;
+class ReceiveStatistics;
+class RemoteBitrateEstimator;
+class RtpHeaderParser;
+class RTPPayloadRegistry;
+class RtpReceiver;
+class RtpRtcp;
+class VideoCodingModule;
+struct ReceiveBandwidthEstimatorStats;
+
+class ViEReceiver : public RtpData {
+ public:
+  ViEReceiver(VideoCodingModule* module_vcm,
+              RemoteBitrateEstimator* remote_bitrate_estimator,
+              RtpFeedback* rtp_feedback);
+  ~ViEReceiver();
+
+  bool SetReceiveCodec(const VideoCodec& video_codec);
+  bool RegisterPayload(const VideoCodec& video_codec);
+
+  void SetNackStatus(bool enable, int max_nack_reordering_threshold);
+  void SetRtxPayloadType(int payload_type, int associated_payload_type);
+  // If set to true, the RTX payload type mapping supplied in
+  // |SetRtxPayloadType| will be used when restoring RTX packets. Without it,
+  // RTX packets will always be restored to the last non-RTX packet payload type
+  // received.
+  void SetUseRtxPayloadMappingOnRestore(bool val);
+  void SetRtxSsrc(uint32_t ssrc);
+  bool GetRtxSsrc(uint32_t* ssrc) const;
+
+  bool IsFecEnabled() const;
+
+  uint32_t GetRemoteSsrc() const;
+  int GetCsrcs(uint32_t* csrcs) const;
+
+  void SetRtpRtcpModule(RtpRtcp* module);
+
+  RtpReceiver* GetRtpReceiver() const;
+
+  void RegisterRtpRtcpModules(const std::vector<RtpRtcp*>& rtp_modules);
+
+  bool SetReceiveTimestampOffsetStatus(bool enable, int id);
+  bool SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
+  bool SetReceiveVideoRotationStatus(bool enable, int id);
+  bool SetReceiveTransportSequenceNumber(bool enable, int id);
+
+  void StartReceive();
+  void StopReceive();
+
+  // Receives packets from external transport.
+  int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length,
+                        const PacketTime& packet_time);
+  int ReceivedRTCPPacket(const void* rtcp_packet, size_t rtcp_packet_length);
+
+  // Implements RtpData.
+  int32_t OnReceivedPayloadData(const uint8_t* payload_data,
+                                const size_t payload_size,
+                                const WebRtcRTPHeader* rtp_header) override;
+  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
+
+  ReceiveStatistics* GetReceiveStatistics() const;
+
+ private:
+  int InsertRTPPacket(const uint8_t* rtp_packet, size_t rtp_packet_length,
+                      const PacketTime& packet_time);
+  bool ReceivePacket(const uint8_t* packet,
+                     size_t packet_length,
+                     const RTPHeader& header,
+                     bool in_order);
+  // Parses and handles for instance RTX and RED headers.
+  // This function assumes that it's being called from only one thread.
+  bool ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
+                                         size_t packet_length,
+                                         const RTPHeader& header);
+  void NotifyReceiverOfFecPacket(const RTPHeader& header);
+  int InsertRTCPPacket(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
+  bool IsPacketInOrder(const RTPHeader& header) const;
+  bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
+  void UpdateHistograms();
+
+  rtc::scoped_ptr<CriticalSectionWrapper> receive_cs_;
+  Clock* clock_;
+  rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
+  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
+  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
+  const rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
+  rtc::scoped_ptr<FecReceiver> fec_receiver_;
+  RtpRtcp* rtp_rtcp_;
+  std::vector<RtpRtcp*> rtp_rtcp_simulcast_;
+  VideoCodingModule* vcm_;
+  RemoteBitrateEstimator* remote_bitrate_estimator_;
+
+  rtc::scoped_ptr<RemoteNtpTimeEstimator> ntp_estimator_;
+
+  bool receiving_;
+  uint8_t restored_packet_[IP_PACKET_SIZE];
+  bool restored_packet_in_use_;
+  bool receiving_ast_enabled_;
+  bool receiving_cvo_enabled_;
+  bool receiving_tsn_enabled_;
+  int64_t last_packet_log_ms_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_VIE_RECEIVER_H_