Replace voe_auto_test

webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h

-/*
- *  Copyright (c) 2011 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 SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
-
-#include <deque>
-#include <memory>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/platform_thread.h"
-#include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
-
-class TestErrorObserver;
-
-class LoopBackTransport : public webrtc::Transport {
- public:
-  LoopBackTransport(webrtc::VoENetwork* voe_network, int channel)
-      : packet_event_(webrtc::EventWrapper::Create()),
-        thread_(NetworkProcess, this, "LoopBackTransport"),
-        channel_(channel),
-        voe_network_(voe_network),
-        transmitted_packets_(0) {
-    thread_.Start();
-  }
-
-  ~LoopBackTransport() { thread_.Stop(); }
-
-  bool SendRtp(const uint8_t* data,
-               size_t len,
-               const webrtc::PacketOptions& options) override {
-    StorePacket(Packet::Rtp, data, len);
-    return true;
-  }
-
-  bool SendRtcp(const uint8_t* data, size_t len) override {
-    StorePacket(Packet::Rtcp, data, len);
-    return true;
-  }
-
-  void WaitForTransmittedPackets(int32_t packet_count) {
-    enum {
-      kSleepIntervalMs = 10
-    };
-    int32_t limit = transmitted_packets_.Value() + packet_count;
-    while (transmitted_packets_.Value() < limit) {
-      webrtc::SleepMs(kSleepIntervalMs);
-    }
-  }
-
-  void AddChannel(uint32_t ssrc, int channel) {
-    rtc::CritScope lock(&crit_);
-    channels_[ssrc] = channel;
-  }
-
- private:
-  struct Packet {
-    enum Type { Rtp, Rtcp, } type;
-
-    Packet() : len(0) {}
-    Packet(Type type, const void* data, size_t len)
-        : type(type), len(len) {
-      assert(len <= 1500);
-      memcpy(this->data, data, len);
-    }
-
-    uint8_t data[1500];
-    size_t len;
-  };
-
-  void StorePacket(Packet::Type type,
-                   const void* data,
-                   size_t len) {
-    {
-      rtc::CritScope lock(&crit_);
-      packet_queue_.push_back(Packet(type, data, len));
-    }
-    packet_event_->Set();
-  }
-
-  static bool NetworkProcess(void* transport) {
-    return static_cast<LoopBackTransport*>(transport)->SendPackets();
-  }
-
-  bool SendPackets() {
-    switch (packet_event_->Wait(10)) {
-      case webrtc::kEventSignaled:
-        break;
-      case webrtc::kEventTimeout:
-        break;
-      case webrtc::kEventError:
-        // TODO(pbos): Log a warning here?
-        return true;
-    }
-
-    while (true) {
-      Packet p;
-      int channel = channel_;
-      {
-        rtc::CritScope lock(&crit_);
-        if (packet_queue_.empty())
-          break;
-        p = packet_queue_.front();
-        packet_queue_.pop_front();
-
-        if (p.type == Packet::Rtp) {
-          uint32_t ssrc =
-              webrtc::ByteReader<uint32_t>::ReadBigEndian(&p.data[8]);
-          if (channels_[ssrc] != 0)
-            channel = channels_[ssrc];
-        }
-        // TODO(pbos): Add RTCP SSRC muxing/demuxing if anything requires it.
-      }
-
-      // Minimum RTP header size.
-      if (p.len < 12)
-        continue;
-
-      switch (p.type) {
-        case Packet::Rtp:
-          voe_network_->ReceivedRTPPacket(channel, p.data, p.len,
-                                          webrtc::PacketTime());
-          break;
-        case Packet::Rtcp:
-          voe_network_->ReceivedRTCPPacket(channel, p.data, p.len);
-          break;
-      }
-      ++transmitted_packets_;
-    }
-    return true;
-  }
-
-  rtc::CriticalSection crit_;
-  const std::unique_ptr<webrtc::EventWrapper> packet_event_;
-  rtc::PlatformThread thread_;
-  std::deque<Packet> packet_queue_ RTC_GUARDED_BY(crit_);
-  const int channel_;
-  std::map<uint32_t, int> channels_ RTC_GUARDED_BY(crit_);
-  webrtc::VoENetwork* const voe_network_;
-  webrtc::Atomic32 transmitted_packets_;
-};
-
-// This fixture initializes the voice engine in addition to the work
-// done by the before-initialization fixture. It also registers an error
-// observer which will fail tests on error callbacks. This fixture is
-// useful to tests that want to run before we have started any form of
-// streaming through the voice engine.
-class AfterInitializationFixture : public BeforeInitializationFixture {
- public:
-  AfterInitializationFixture();
-  virtual ~AfterInitializationFixture();
-
- protected:
-  std::unique_ptr<TestErrorObserver> error_observer_;
-};
-
-#endif  // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_