Replace voe_auto_test

voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc

-/*
- *  Copyright (c) 2014 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.
- */
-
-#include <memory>
-
-#include "modules/include/module_common_types.h"
-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "system_wrappers/include/atomic32.h"
-#include "system_wrappers/include/sleep.h"
-#include "voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
-
-using ::testing::_;
-using ::testing::AtLeast;
-using ::testing::Eq;
-using ::testing::Field;
-
-class ExtensionVerifyTransport : public webrtc::Transport {
- public:
-  ExtensionVerifyTransport()
-      : parser_(webrtc::RtpHeaderParser::Create()),
-        received_packets_(0),
-        bad_packets_(0),
-        audio_level_id_(-1),
-        absolute_sender_time_id_(-1) {}
-
-  bool SendRtp(const uint8_t* data,
-               size_t len,
-               const webrtc::PacketOptions& options) override {
-    webrtc::RTPHeader header;
-    if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
-      bool ok = true;
-      if (audio_level_id_ >= 0 &&
-          !header.extension.hasAudioLevel) {
-        ok = false;
-      }
-      if (absolute_sender_time_id_ >= 0 &&
-          !header.extension.hasAbsoluteSendTime) {
-        ok = false;
-      }
-      if (!ok) {
-        // bad_packets_ count packets we expected to have an extension but
-        // didn't have one.
-        ++bad_packets_;
-      }
-    }
-    // received_packets_ count all packets we receive.
-    ++received_packets_;
-    return true;
-  }
-
-  bool SendRtcp(const uint8_t* data, size_t len) override {
-    return true;
-  }
-
-  void SetAudioLevelId(int id) {
-    audio_level_id_ = id;
-    parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, id);
-  }
-
-  void SetAbsoluteSenderTimeId(int id) {
-    absolute_sender_time_id_ = id;
-    parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAbsoluteSendTime,
-                                        id);
-  }
-
-  bool Wait() {
-    // Wait until we've received to specified number of packets.
-    while (received_packets_.Value() < kPacketsExpected) {
-      webrtc::SleepMs(kSleepIntervalMs);
-    }
-    // Check whether any were 'bad' (didn't contain an extension when they
-    // where supposed to).
-    return bad_packets_.Value() == 0;
-  }
-
- private:
-  enum {
-    kPacketsExpected = 10,
-    kSleepIntervalMs = 10
-  };
-  std::unique_ptr<webrtc::RtpHeaderParser> parser_;
-  webrtc::Atomic32 received_packets_;
-  webrtc::Atomic32 bad_packets_;
-  int audio_level_id_;
-  int absolute_sender_time_id_;
-};
-
-class SendRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture {
- protected:
-  void SetUp() override {
-    EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
-    EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_,
-                                                         verifying_transport_));
-  }
-  void TearDown() override { PausePlaying(); }
-
-  ExtensionVerifyTransport verifying_transport_;
-};
-
-TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeNoAudioLevel) {
-  verifying_transport_.SetAudioLevelId(0);
-  ResumePlaying();
-  EXPECT_FALSE(verifying_transport_.Wait());
-}
-
-TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAudioLevel) {
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
-                                                                9));
-  verifying_transport_.SetAudioLevelId(9);
-  ResumePlaying();
-  EXPECT_TRUE(verifying_transport_.Wait());
-}
-