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, 164 insertions, 0 deletions

diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
new file mode 100644
index 0000000000..7b45e6d3e1
--- /dev/null
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -0,0 +1,164 @@
+/*
+ *  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_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
+#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
+
+#include <deque>
+#include <map>
+#include <utility>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/thread_wrapper.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_file.h"
+#include "webrtc/voice_engine/include/voe_network.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
+
+static const size_t kMaxPacketSizeByte = 1500;
+
+namespace voetest {
+
+// This class is to simulate a conference call. There are two Voice Engines, one
+// for local channels and the other for remote channels. There is a simulated
+// reflector, which exchanges RTCP with local channels. For simplicity, it
+// also uses the Voice Engine for remote channels. One can add streams by
+// calling AddStream(), which creates a remote sender channel and a local
+// receive channel. The remote sender channel plays a file as microphone in a
+// looped fashion. Received streams are mixed and played.
+
+class ConferenceTransport: public webrtc::Transport {
+ public:
+  ConferenceTransport();
+  virtual ~ConferenceTransport();
+
+  /* SetRtt()
+   * Set RTT between local channels and reflector.
+   *
+   * Input:
+   *   rtt_ms : RTT in milliseconds.
+   */
+  void SetRtt(unsigned int rtt_ms);
+
+  /* AddStream()
+   * Adds a stream in the conference.
+   *
+   * Input:
+   *   file_name : name of the file to be added as microphone input.
+   *   format    : format of the input file.
+   *
+   * Returns stream id.
+   */
+  unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
+
+  /* RemoveStream()
+   * Removes a stream with specified ID from the conference.
+   *
+   * Input:
+   *   id : stream id.
+   *
+   * Returns false if the specified stream does not exist, true if succeeds.
+   */
+  bool RemoveStream(unsigned int id);
+
+  /* StartPlayout()
+   * Starts playing out the stream with specified ID, using the default device.
+   *
+   * Input:
+   *   id : stream id.
+   *
+   * Returns false if the specified stream does not exist, true if succeeds.
+   */
+  bool StartPlayout(unsigned int id);
+
+  /* GetReceiverStatistics()
+   * Gets RTCP statistics of the stream with specified ID.
+   *
+   * Input:
+   *   id : stream id;
+   *   stats : pointer to a CallStatistics to store the result.
+   *
+   * Returns false if the specified stream does not exist, true if succeeds.
+   */
+  bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
+
+  // Inherit from class webrtc::Transport.
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override;
+  bool SendRtcp(const uint8_t *data, size_t len) override;
+
+ private:
+  struct Packet {
+    enum Type { Rtp, Rtcp, } type_;
+
+    Packet() : len_(0) {}
+    Packet(Type type, const void* data, size_t len, uint32_t time_ms)
+        : type_(type), len_(len), send_time_ms_(time_ms) {
+      EXPECT_LE(len_, kMaxPacketSizeByte);
+      memcpy(data_, data, len_);
+    }
+
+    uint8_t data_[kMaxPacketSizeByte];
+    size_t len_;
+    uint32_t send_time_ms_;
+  };
+
+  static bool Run(void* transport) {
+    return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
+  }
+
+  int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
+  void StorePacket(Packet::Type type, const void* data, size_t len);
+  void SendPacket(const Packet& packet);
+  bool DispatchPackets();
+
+  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
+  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
+  const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
+  const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
+
+  unsigned int rtt_ms_;
+  unsigned int stream_count_;
+
+  std::map<unsigned int, std::pair<int, int>> streams_
+    GUARDED_BY(stream_crit_.get());
+  std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get());
+
+  int local_sender_;  // Channel Id of local sender
+  int reflector_;
+
+  webrtc::VoiceEngine* local_voe_;
+  webrtc::VoEBase* local_base_;
+  webrtc::VoERTP_RTCP* local_rtp_rtcp_;
+  webrtc::VoENetwork* local_network_;
+
+  webrtc::VoiceEngine* remote_voe_;
+  webrtc::VoEBase* remote_base_;
+  webrtc::VoECodec* remote_codec_;
+  webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
+  webrtc::VoENetwork* remote_network_;
+  webrtc::VoEFile* remote_file_;
+
+  LoudestFilter loudest_filter_;
+
+  const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
+};
+}  // namespace voetest
+
+#endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_