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, 196 insertions, 0 deletions

diff --git a/webrtc/video/vie_encoder.h b/webrtc/video/vie_encoder.h
new file mode 100644
index 0000000000..a15fd8920b
--- /dev/null
+++ b/webrtc/video/vie_encoder.h
@@ -0,0 +1,196 @@
+/*
+ *  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_ENCODER_H_
+#define WEBRTC_VIDEO_VIE_ENCODER_H_
+
+#include <map>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/common_types.h"
+#include "webrtc/frame_callback.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video/video_capture_input.h"
+
+namespace webrtc {
+
+class BitrateAllocator;
+class BitrateObserver;
+class Config;
+class CriticalSectionWrapper;
+class EncodedImageCallback;
+class PacedSender;
+class PayloadRouter;
+class ProcessThread;
+class QMVideoSettingsCallback;
+class SendStatisticsProxy;
+class ViEBitrateObserver;
+class ViEEffectFilter;
+class VideoCodingModule;
+
+class ViEEncoder : public RtcpIntraFrameObserver,
+                   public VideoEncoderRateObserver,
+                   public VCMPacketizationCallback,
+                   public VCMSendStatisticsCallback,
+                   public VideoCaptureCallback {
+ public:
+  friend class ViEBitrateObserver;
+
+  ViEEncoder(uint32_t number_of_cores,
+             ProcessThread* module_process_thread,
+             SendStatisticsProxy* stats_proxy,
+             I420FrameCallback* pre_encode_callback,
+             PacedSender* pacer,
+             BitrateAllocator* bitrate_allocator);
+  ~ViEEncoder();
+
+  bool Init();
+
+  // This function is assumed to be called before any frames are delivered and
+  // only once.
+  // Ideally this would be done in Init, but the dependencies between ViEEncoder
+  // and ViEChannel makes it really hard to do in a good way.
+  void StartThreadsAndSetSharedMembers(
+      rtc::scoped_refptr<PayloadRouter> send_payload_router,
+      VCMProtectionCallback* vcm_protection_callback);
+
+  // This function must be called before the corresponding ViEChannel is
+  // deleted.
+  void StopThreadsAndRemoveSharedMembers();
+
+  void SetNetworkTransmissionState(bool is_transmitting);
+
+  // Returns the id of the owning channel.
+  int Owner() const;
+
+  // Drops incoming packets before they get to the encoder.
+  void Pause();
+  void Restart();
+
+  // Codec settings.
+  int32_t RegisterExternalEncoder(VideoEncoder* encoder,
+                                  uint8_t pl_type,
+                                  bool internal_source);
+  int32_t DeRegisterExternalEncoder(uint8_t pl_type);
+  int32_t SetEncoder(const VideoCodec& video_codec);
+
+  // Implementing VideoCaptureCallback.
+  void DeliverFrame(VideoFrame video_frame) override;
+
+  int32_t SendKeyFrame();
+
+  uint32_t LastObservedBitrateBps() const;
+  int CodecTargetBitrate(uint32_t* bitrate) const;
+  // Loss protection. Must be called before SetEncoder() to have max packet size
+  // updated according to protection.
+  // TODO(pbos): Set protection method on construction or extract vcm_ outside
+  // this class and set it on construction there.
+  void SetProtectionMethod(bool nack, bool fec);
+
+  // Buffering mode.
+  void SetSenderBufferingMode(int target_delay_ms);
+
+  // Implements VideoEncoderRateObserver.
+  void OnSetRates(uint32_t bitrate_bps, int framerate) override;
+
+  // Implements VCMPacketizationCallback.
+  int32_t SendData(uint8_t payload_type,
+                   const EncodedImage& encoded_image,
+                   const RTPFragmentationHeader& fragmentation_header,
+                   const RTPVideoHeader* rtp_video_hdr) override;
+  void OnEncoderImplementationName(const char* implementation_name) override;
+
+  // Implements VideoSendStatisticsCallback.
+  int32_t SendStatistics(const uint32_t bit_rate,
+                         const uint32_t frame_rate) override;
+
+  // Implements RtcpIntraFrameObserver.
+  void OnReceivedIntraFrameRequest(uint32_t ssrc) override;
+  void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) override;
+  void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) override;
+  void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) override;
+
+  // Sets SSRCs for all streams.
+  void SetSsrcs(const std::vector<uint32_t>& ssrcs);
+
+  void SetMinTransmitBitrate(int min_transmit_bitrate_kbps);
+
+  // Lets the sender suspend video when the rate drops below
+  // |threshold_bps|, and turns back on when the rate goes back up above
+  // |threshold_bps| + |window_bps|.
+  void SuspendBelowMinBitrate();
+
+  // New-style callbacks, used by VideoSendStream.
+  void RegisterPostEncodeImageCallback(
+        EncodedImageCallback* post_encode_callback);
+
+  int GetPaddingNeededBps() const;
+
+ protected:
+  // Called by BitrateObserver.
+  void OnNetworkChanged(uint32_t bitrate_bps,
+                        uint8_t fraction_lost,
+                        int64_t round_trip_time_ms);
+
+ private:
+  bool EncoderPaused() const EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
+  void TraceFrameDropStart() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
+  void TraceFrameDropEnd() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
+
+  const uint32_t number_of_cores_;
+
+  const rtc::scoped_ptr<VideoProcessing> vp_;
+  const rtc::scoped_ptr<QMVideoSettingsCallback> qm_callback_;
+  const rtc::scoped_ptr<VideoCodingModule> vcm_;
+  rtc::scoped_refptr<PayloadRouter> send_payload_router_;
+
+  rtc::scoped_ptr<CriticalSectionWrapper> data_cs_;
+  rtc::scoped_ptr<BitrateObserver> bitrate_observer_;
+
+  SendStatisticsProxy* const stats_proxy_;
+  I420FrameCallback* const pre_encode_callback_;
+  PacedSender* const pacer_;
+  BitrateAllocator* const bitrate_allocator_;
+
+  // The time we last received an input frame or encoded frame. This is used to
+  // track when video is stopped long enough that we also want to stop sending
+  // padding.
+  int64_t time_of_last_frame_activity_ms_ GUARDED_BY(data_cs_);
+  VideoCodec encoder_config_ GUARDED_BY(data_cs_);
+  int min_transmit_bitrate_kbps_ GUARDED_BY(data_cs_);
+  uint32_t last_observed_bitrate_bps_ GUARDED_BY(data_cs_);
+  int target_delay_ms_ GUARDED_BY(data_cs_);
+  bool network_is_transmitting_ GUARDED_BY(data_cs_);
+  bool encoder_paused_ GUARDED_BY(data_cs_);
+  bool encoder_paused_and_dropped_frame_ GUARDED_BY(data_cs_);
+  std::map<unsigned int, int64_t> time_last_intra_request_ms_
+      GUARDED_BY(data_cs_);
+
+  ProcessThread* module_process_thread_;
+
+  bool has_received_sli_ GUARDED_BY(data_cs_);
+  uint8_t picture_id_sli_ GUARDED_BY(data_cs_);
+  bool has_received_rpsi_ GUARDED_BY(data_cs_);
+  uint64_t picture_id_rpsi_ GUARDED_BY(data_cs_);
+  std::map<uint32_t, int> ssrc_streams_ GUARDED_BY(data_cs_);
+
+  bool video_suspended_ GUARDED_BY(data_cs_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_VIE_ENCODER_H_