Replace voe_conference_test.

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

-/*
- *  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 <memory>
-#include <utility>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/platform_thread.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/gtest.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"
-
-namespace webrtc {
-namespace voetest {
-
-static const size_t kMaxPacketSizeByte = 1500;
-
-// 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, int64_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_;
-    int64_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();
-
-  rtc::CriticalSection pq_crit_;
-  rtc::CriticalSection stream_crit_;
-  const std::unique_ptr<webrtc::EventWrapper> packet_event_;
-  rtc::PlatformThread thread_;
-
-  unsigned int rtt_ms_;
-  unsigned int stream_count_;
-
-  std::map<unsigned int, std::pair<int, int>> streams_
-      RTC_GUARDED_BY(stream_crit_);
-  std::deque<Packet> packet_queue_ RTC_GUARDED_BY(pq_crit_);
-
-  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_;
-  rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_;
-
-  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_;
-  rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_;
-  LoudestFilter loudest_filter_;
-
-  const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
-};
-
-}  // namespace voetest
-}  // namespace webrtc
-
-#endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_