Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 194 insertions, 0 deletions

diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
new file mode 100644
index 0000000000..c42647e2d3
--- /dev/null
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2015 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_REMOTE_BITRATE_ESTIMATOR_TEST_PACKET_SENDER_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_PACKET_SENDER_H_
+
+#include <list>
+#include <limits>
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+class MetricRecorder;
+
+class PacketSender : public PacketProcessor {
+ public:
+  PacketSender(PacketProcessorListener* listener, int flow_id)
+      : PacketProcessor(listener, flow_id, kSender),
+        running_(true),
+        // For Packet::send_time_us() to be comparable with timestamps from
+        // clock_, the clock of the PacketSender and the Source must be aligned.
+        // We assume that both start at time 0.
+        clock_(0),
+        metric_recorder_(nullptr) {}
+  virtual ~PacketSender() {}
+  // Call GiveFeedback() with the returned interval in milliseconds, provided
+  // there is a new estimate available.
+  // Note that changing the feedback interval affects the timing of when the
+  // output of the estimators is sampled and therefore the baseline files may
+  // have to be regenerated.
+  virtual int GetFeedbackIntervalMs() const = 0;
+  void SetSenderTimestamps(Packets* in_out);
+
+  virtual uint32_t TargetBitrateKbps() { return 0; }
+
+  virtual void Pause();
+  virtual void Resume(int64_t paused_time_ms);
+
+  void set_metric_recorder(MetricRecorder* metric_recorder);
+  virtual void RecordBitrate();
+
+ protected:
+  bool running_;  // Initialized by default as true.
+  SimulatedClock clock_;
+
+ private:
+  MetricRecorder* metric_recorder_;
+};
+
+class VideoSender : public PacketSender, public BitrateObserver {
+ public:
+  VideoSender(PacketProcessorListener* listener,
+              VideoSource* source,
+              BandwidthEstimatorType estimator);
+  virtual ~VideoSender();
+
+  int GetFeedbackIntervalMs() const override;
+  void RunFor(int64_t time_ms, Packets* in_out) override;
+
+  virtual VideoSource* source() const { return source_; }
+
+  uint32_t TargetBitrateKbps() override;
+
+  // Implements BitrateObserver.
+  void OnNetworkChanged(uint32_t target_bitrate_bps,
+                        uint8_t fraction_lost,
+                        int64_t rtt) override;
+
+  void Pause() override;
+  void Resume(int64_t paused_time_ms) override;
+
+ protected:
+  void ProcessFeedbackAndGeneratePackets(int64_t time_ms,
+                                         std::list<FeedbackPacket*>* feedbacks,
+                                         Packets* generated);
+
+  VideoSource* source_;
+  rtc::scoped_ptr<BweSender> bwe_;
+  int64_t start_of_run_ms_;
+  std::list<Module*> modules_;
+
+ private:
+  uint32_t previous_sending_bitrate_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(VideoSender);
+};
+
+class PacedVideoSender : public VideoSender, public PacedSender::Callback {
+ public:
+  PacedVideoSender(PacketProcessorListener* listener,
+                   VideoSource* source,
+                   BandwidthEstimatorType estimator);
+  virtual ~PacedVideoSender();
+
+  void RunFor(int64_t time_ms, Packets* in_out) override;
+
+  // Implements PacedSender::Callback.
+  bool TimeToSendPacket(uint32_t ssrc,
+                        uint16_t sequence_number,
+                        int64_t capture_time_ms,
+                        bool retransmission) override;
+  size_t TimeToSendPadding(size_t bytes) override;
+
+  // Implements BitrateObserver.
+  void OnNetworkChanged(uint32_t target_bitrate_bps,
+                        uint8_t fraction_lost,
+                        int64_t rtt) override;
+
+ private:
+  int64_t TimeUntilNextProcess(const std::list<Module*>& modules);
+  void CallProcess(const std::list<Module*>& modules);
+  void QueuePackets(Packets* batch, int64_t end_of_batch_time_us);
+
+  PacedSender pacer_;
+  Packets queue_;
+  Packets pacer_queue_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacedVideoSender);
+};
+
+class TcpSender : public PacketSender {
+ public:
+  TcpSender(PacketProcessorListener* listener, int flow_id, int64_t offset_ms);
+  TcpSender(PacketProcessorListener* listener,
+            int flow_id,
+            int64_t offset_ms,
+            int send_limit_bytes);
+  virtual ~TcpSender() {}
+
+  void RunFor(int64_t time_ms, Packets* in_out) override;
+  int GetFeedbackIntervalMs() const override { return 10; }
+
+  uint32_t TargetBitrateKbps() override;
+
+ private:
+  struct InFlight {
+   public:
+    InFlight(const MediaPacket& packet)
+        : sequence_number(packet.header().sequenceNumber),
+          time_ms(packet.send_time_ms()) {}
+
+    InFlight(uint16_t seq_num, int64_t now_ms)
+        : sequence_number(seq_num), time_ms(now_ms) {}
+
+    bool operator<(const InFlight& rhs) const {
+      return sequence_number < rhs.sequence_number;
+    }
+
+    uint16_t sequence_number;  // Sequence number of a packet in flight, or a
+                               // packet which has just been acked.
+    int64_t time_ms;  // Time of when the packet left the sender, or when the
+                      // ack was received.
+  };
+
+  void SendPackets(Packets* in_out);
+  void UpdateCongestionControl(const FeedbackPacket* fb);
+  int TriggerTimeouts();
+  void HandleLoss();
+  Packets GeneratePackets(size_t num_packets);
+  void UpdateSendBitrateEstimate(size_t num_packets);
+
+  float cwnd_;
+  int ssthresh_;
+  std::set<InFlight> in_flight_;
+  bool ack_received_;
+  uint16_t last_acked_seq_num_;
+  uint16_t next_sequence_number_;
+  int64_t offset_ms_;
+  int64_t last_reduction_time_ms_;
+  int64_t last_rtt_ms_;
+  int total_sent_bytes_;
+  int send_limit_bytes_;  // Initialized by default as kNoLimit.
+  int64_t last_generated_packets_ms_;
+  size_t num_recent_sent_packets_;
+  uint32_t bitrate_kbps_;
+};
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_PACKET_SENDER_H_