Remove voe_auto_test and add new tests to cover the missing cases.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/3007383002
Cr-Commit-Position: refs/heads/master@{#19865}
diff --git a/BUILD.gn b/BUILD.gn
index ed80cbe..e590644 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -338,9 +338,6 @@
} else {
deps += [ "modules/video_capture:video_capture_tests" ]
}
- if (!is_ios) {
- deps += [ "voice_engine:voe_auto_test" ]
- }
if (rtc_enable_protobuf) {
deps += [
"audio:low_bandwidth_audio_test",
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index aa33192..04dec03 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -95,6 +95,7 @@
sources = [
"audio_receive_stream_unittest.cc",
+ "audio_send_stream_tests.cc",
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
"time_interval_unittest.cc",
diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc
new file mode 100644
index 0000000..4283b73
--- /dev/null
+++ b/audio/audio_send_stream_tests.cc
@@ -0,0 +1,238 @@
+/*
+ * Copyright (c) 2017 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 "test/call_test.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+class AudioSendTest : public SendTest {
+ public:
+ AudioSendTest() : SendTest(CallTest::kDefaultTimeoutMs) {}
+
+ size_t GetNumVideoStreams() const override {
+ return 0;
+ }
+ size_t GetNumAudioStreams() const override {
+ return 1;
+ }
+ size_t GetNumFlexfecStreams() const override {
+ return 0;
+ }
+};
+} // namespace
+
+using AudioSendStreamCallTest = CallTest;
+
+TEST_F(AudioSendStreamCallTest, SupportsCName) {
+ static std::string kCName = "PjqatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
+ class CNameObserver : public AudioSendTest {
+ public:
+ CNameObserver() = default;
+
+ private:
+ Action OnSendRtcp(const uint8_t* packet, size_t length) override {
+ RtcpPacketParser parser;
+ EXPECT_TRUE(parser.Parse(packet, length));
+ if (parser.sdes()->num_packets() > 0) {
+ EXPECT_EQ(1u, parser.sdes()->chunks().size());
+ EXPECT_EQ(kCName, parser.sdes()->chunks()[0].cname);
+
+ observation_complete_.Set();
+ }
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.c_name = kCName;
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP with CNAME.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, NoExtensionsByDefault) {
+ class NoExtensionsObserver : public AudioSendTest {
+ public:
+ NoExtensionsObserver() = default;
+
+ private:
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
+ EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
+ EXPECT_FALSE(header.extension.hasTransportSequenceNumber);
+ EXPECT_FALSE(header.extension.hasAudioLevel);
+ EXPECT_FALSE(header.extension.hasVideoRotation);
+ EXPECT_FALSE(header.extension.hasVideoContentType);
+ observation_complete_.Set();
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) {
+ class AudioLevelObserver : public AudioSendTest {
+ public:
+ AudioLevelObserver() : AudioSendTest() {
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel, test::kAudioLevelExtensionId));
+ }
+
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ EXPECT_TRUE(header.extension.hasAudioLevel);
+ if (header.extension.audioLevel != 0) {
+ // Wait for at least one packet with a non-zero level.
+ observation_complete_.Set();
+ } else {
+ LOG(LS_WARNING) << "Got a packet with zero audioLevel - waiting"
+ " for another packet...";
+ }
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAudioLevelUri, test::kAudioLevelExtensionId));
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SupportsTransportWideSequenceNumbers) {
+ static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
+ class TransportWideSequenceNumberObserver : public AudioSendTest {
+ public:
+ TransportWideSequenceNumberObserver() : AudioSendTest() {
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionTransportSequenceNumber, kExtensionId));
+ }
+
+ private:
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+ EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
+ EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
+
+ observation_complete_.Set();
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SendDtmf) {
+ static const uint8_t kDtmfPayloadType = 120;
+ static const int kDtmfPayloadFrequency = 8000;
+ static const int kDtmfEventFirst = 12;
+ static const int kDtmfEventLast = 31;
+ static const int kDtmfDuration = 50;
+ class DtmfObserver : public AudioSendTest {
+ public:
+ DtmfObserver() = default;
+
+ private:
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ if (header.payloadType == kDtmfPayloadType) {
+ EXPECT_EQ(12u, header.headerLength);
+ EXPECT_EQ(16u, length);
+ const int event = packet[12];
+ if (event != expected_dtmf_event_) {
+ ++expected_dtmf_event_;
+ EXPECT_EQ(event, expected_dtmf_event_);
+ if (expected_dtmf_event_ == kDtmfEventLast) {
+ observation_complete_.Set();
+ }
+ }
+ }
+
+ return SEND_PACKET;
+ }
+
+ void OnAudioStreamsCreated(
+ AudioSendStream* send_stream,
+ const std::vector<AudioReceiveStream*>& receive_streams) override {
+ // Need to start stream here, else DTMF events are dropped.
+ send_stream->Start();
+ for (int event = kDtmfEventFirst; event <= kDtmfEventLast; ++event) {
+ send_stream->SendTelephoneEvent(kDtmfPayloadType, kDtmfPayloadFrequency,
+ event, kDtmfDuration);
+ }
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for DTMF stream.";
+ }
+
+ int expected_dtmf_event_ = kDtmfEventFirst;
+ } test;
+
+ RunBaseTest(&test);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/constants.cc b/test/constants.cc
index 1a010c5..0835777 100644
--- a/test/constants.cc
+++ b/test/constants.cc
@@ -13,6 +13,7 @@
namespace webrtc {
namespace test {
+const int kAudioLevelExtensionId = 5;
const int kTOffsetExtensionId = 6;
const int kAbsSendTimeExtensionId = 7;
const int kTransportSequenceNumberExtensionId = 8;
diff --git a/test/constants.h b/test/constants.h
index e41e0da..85e8c18 100644
--- a/test/constants.h
+++ b/test/constants.h
@@ -11,6 +11,7 @@
namespace webrtc {
namespace test {
+extern const int kAudioLevelExtensionId;
extern const int kTOffsetExtensionId;
extern const int kAbsSendTimeExtensionId;
extern const int kTransportSequenceNumberExtensionId;
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 482197a..b61f909 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -240,71 +240,4 @@
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
-
- if (!is_ios) {
- rtc_executable("voe_auto_test") {
- testonly = true
-
- deps = [
- ":voice_engine",
- "..:webrtc_common",
- "../logging:rtc_event_log_api",
- "../modules:module_api",
- "../modules/audio_device:audio_device",
- "../modules/audio_processing:audio_processing",
- "../modules/rtp_rtcp:rtp_rtcp",
- "../modules/video_capture",
- "../rtc_base:rtc_base_approved",
- "../system_wrappers",
- "../system_wrappers/:system_wrappers_default",
- "../test/:test_common",
- "../test/:test_support",
- "../test/:video_test_common",
- "//testing/gmock",
- "//testing/gtest",
- ]
-
- sources = [
- "test/auto_test/automated_mode.cc",
- "test/auto_test/fixtures/after_initialization_fixture.cc",
- "test/auto_test/fixtures/after_initialization_fixture.h",
- "test/auto_test/fixtures/after_streaming_fixture.cc",
- "test/auto_test/fixtures/after_streaming_fixture.h",
- "test/auto_test/fixtures/before_initialization_fixture.cc",
- "test/auto_test/fixtures/before_initialization_fixture.h",
- "test/auto_test/fixtures/before_streaming_fixture.cc",
- "test/auto_test/fixtures/before_streaming_fixture.h",
- "test/auto_test/standard/codec_before_streaming_test.cc",
- "test/auto_test/standard/codec_test.cc",
- "test/auto_test/standard/dtmf_test.cc",
- "test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
- "test/auto_test/standard/rtp_rtcp_extensions.cc",
- "test/auto_test/standard/rtp_rtcp_test.cc",
- "test/auto_test/voe_standard_test.cc",
- "test/auto_test/voe_standard_test.h",
- "test/auto_test/voe_test_defines.h",
- ]
-
- defines = []
-
- if (rtc_enable_protobuf) {
- defines = [ "ENABLE_RTC_EVENT_LOG" ]
- }
-
- if (is_win) {
- defines += [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
-
- cflags = [
- "/wd4267", # size_t to int truncation.
- "/wd4373", # Virtual function override.
- # TODO(kjellander): Bug 261: fix this warning.
- ]
- }
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
- }
- }
}
diff --git a/voice_engine/test/auto_test/automated_mode.cc b/voice_engine/test/auto_test/automated_mode.cc
deleted file mode 100644
index 2893295..0000000
--- a/voice_engine/test/auto_test/automated_mode.cc
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * 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.
- */
-
-#include "test/gtest.h"
-#include "test/testsupport/fileutils.h"
-
-namespace webrtc {
-namespace voetest {
-
-void InitializeGoogleTest(int* argc, char** argv) {
- // Initialize WebRTC testing framework so paths to resources can be resolved.
- webrtc::test::SetExecutablePath(argv[0]);
- testing::InitGoogleTest(argc, argv);
-}
-
-int RunInAutomatedMode() {
- return RUN_ALL_TESTS();
-}
-
-} // namespace voetest
-} // namespace webrtc
diff --git a/voice_engine/test/auto_test/automated_mode.h b/voice_engine/test/auto_test/automated_mode.h
deleted file mode 100644
index 0d673a4..0000000
--- a/voice_engine/test/auto_test/automated_mode.h
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * 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_AUTOMATED_MODE_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
-
-namespace webrtc {
-namespace voetest {
-
-void InitializeGoogleTest(int* argc, char** argv);
-int RunInAutomatedMode();
-
-} // namespace voetest
-} // namespace webrtc
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
diff --git a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc b/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
deleted file mode 100644
index 6aa6d6e..0000000
--- a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_processing/include/audio_processing.h"
-#include "voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-
-class TestErrorObserver : public webrtc::VoiceEngineObserver {
- public:
- TestErrorObserver() {}
- virtual ~TestErrorObserver() {}
- void CallbackOnError(int channel, int error_code) {
- ADD_FAILURE() << "Unexpected error on channel " << channel <<
- ": error code " << error_code;
- }
-};
-
-AfterInitializationFixture::AfterInitializationFixture()
- : error_observer_(new TestErrorObserver()) {
- webrtc::Config config;
- config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(false));
- webrtc::AudioProcessing* audioproc = webrtc::AudioProcessing::Create(config);
-
- EXPECT_EQ(0, voe_base_->Init(NULL, audioproc));
- EXPECT_EQ(0, voe_base_->RegisterVoiceEngineObserver(*error_observer_));
-}
-
-AfterInitializationFixture::~AfterInitializationFixture() {
- EXPECT_EQ(0, voe_base_->DeRegisterVoiceEngineObserver());
-}
diff --git a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
deleted file mode 100644
index 4ce0b0f..0000000
--- a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ /dev/null
@@ -1,169 +0,0 @@
-/*
- * 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 "common_types.h" // NOLINT(build/include)
-#include "modules/rtp_rtcp/source/byte_io.h"
-#include "rtc_base/criticalsection.h"
-#include "rtc_base/platform_thread.h"
-#include "system_wrappers/include/atomic32.h"
-#include "system_wrappers/include/event_wrapper.h"
-#include "system_wrappers/include/sleep.h"
-#include "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_
diff --git a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc b/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
deleted file mode 100644
index dec014b..0000000
--- a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * Copyright (c) 2012 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 "voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
-#include "voice_engine/voice_engine_impl.h"
-
-AfterStreamingFixture::AfterStreamingFixture()
- : BeforeStreamingFixture() {
- webrtc::VoiceEngineImpl* voe_impl =
- static_cast<webrtc::VoiceEngineImpl*>(voice_engine_);
- channel_proxy_ = voe_impl->GetChannelProxy(channel_);
- channel_proxy_->RegisterLegacyReceiveCodecs();
- ResumePlaying();
-}
diff --git a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h b/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h
deleted file mode 100644
index 2164328..0000000
--- a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright (c) 2012 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_AFTER_STREAMING_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
-
-#include <memory>
-
-#include "voice_engine/channel_proxy.h"
-#include "voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
-
-// This fixture will, in addition to the work done by its superclasses,
-// start play back on construction.
-class AfterStreamingFixture : public BeforeStreamingFixture {
- public:
- AfterStreamingFixture();
- virtual ~AfterStreamingFixture() {}
-
- protected:
- std::unique_ptr<webrtc::voe::ChannelProxy> channel_proxy_;
-};
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
diff --git a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
deleted file mode 100644
index b647e88..0000000
--- a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2012 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 "voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
-
-#include "system_wrappers/include/sleep.h"
-
-BeforeInitializationFixture::BeforeInitializationFixture()
- : voice_engine_(webrtc::VoiceEngine::Create()) {
- EXPECT_TRUE(voice_engine_ != NULL);
-
- voe_base_ = webrtc::VoEBase::GetInterface(voice_engine_);
- voe_codec_ = webrtc::VoECodec::GetInterface(voice_engine_);
- voe_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(voice_engine_);
- voe_network_ = webrtc::VoENetwork::GetInterface(voice_engine_);
- voe_file_ = webrtc::VoEFile::GetInterface(voice_engine_);
-}
-
-BeforeInitializationFixture::~BeforeInitializationFixture() {
- voe_base_->Release();
- voe_codec_->Release();
- voe_rtp_rtcp_->Release();
- voe_network_->Release();
- voe_file_->Release();
-
- EXPECT_TRUE(webrtc::VoiceEngine::Delete(voice_engine_));
-}
-
-void BeforeInitializationFixture::Sleep(long milliseconds) {
- webrtc::SleepMs(milliseconds);
-}
diff --git a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
deleted file mode 100644
index 974c82e..0000000
--- a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * 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_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
-
-#include "common_types.h" // NOLINT(build/include)
-#include "test/gmock.h"
-#include "test/gtest.h"
-#include "typedefs.h" // NOLINT(build/include)
-#include "voice_engine/include/voe_base.h"
-#include "voice_engine/include/voe_codec.h"
-#include "voice_engine/include/voe_errors.h"
-#include "voice_engine/include/voe_file.h"
-#include "voice_engine/include/voe_network.h"
-#include "voice_engine/include/voe_rtp_rtcp.h"
-#include "voice_engine/test/auto_test/voe_test_common.h"
-
-// This convenient fixture sets up all voice engine interfaces automatically for
-// use by testing subclasses. It allocates each interface and releases it once
-// which means that if a tests allocates additional interfaces from the voice
-// engine and forgets to release it, this test will fail in the destructor.
-// It will not call any init methods.
-//
-// Implementation note:
-// The interface fetching is done in the constructor and not SetUp() since
-// this relieves our subclasses from calling SetUp in the superclass if they
-// choose to override SetUp() themselves. This is fine as googletest will
-// construct new test objects for each method.
-class BeforeInitializationFixture : public testing::Test {
- public:
- BeforeInitializationFixture();
- virtual ~BeforeInitializationFixture();
-
- protected:
- // Use this sleep function to sleep in tests.
- void Sleep(long milliseconds);
-
- webrtc::VoiceEngine* voice_engine_;
- webrtc::VoEBase* voe_base_;
- webrtc::VoECodec* voe_codec_;
- webrtc::VoERTP_RTCP* voe_rtp_rtcp_;
- webrtc::VoENetwork* voe_network_;
- webrtc::VoEFile* voe_file_;
-};
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
diff --git a/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc b/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
deleted file mode 100644
index 554face..0000000
--- a/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * 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 "test/testsupport/fileutils.h"
-#include "voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
-
-BeforeStreamingFixture::BeforeStreamingFixture()
- : channel_(voe_base_->CreateChannel()),
- transport_(NULL) {
- EXPECT_GE(channel_, 0);
-
- fake_microphone_input_file_ =
- webrtc::test::ResourcePath("voice_engine/audio_long16", "pcm");
-
- SetUpLocalPlayback();
- RestartFakeMicrophone();
-}
-
-BeforeStreamingFixture::~BeforeStreamingFixture() {
- voe_file_->StopPlayingFileAsMicrophone(channel_);
- PausePlaying();
-
- EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
- voe_base_->DeleteChannel(channel_);
- delete transport_;
-}
-
-void BeforeStreamingFixture::SwitchToManualMicrophone() {
- EXPECT_EQ(0, voe_file_->StopPlayingFileAsMicrophone(channel_));
-
- TEST_LOG("You need to speak manually into the microphone for this test.\n");
- TEST_LOG("Please start speaking now.\n");
- Sleep(1000);
-}
-
-void BeforeStreamingFixture::RestartFakeMicrophone() {
- EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
- channel_, fake_microphone_input_file_.c_str(), true, true));
-}
-
-void BeforeStreamingFixture::PausePlaying() {
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
-}
-
-void BeforeStreamingFixture::ResumePlaying() {
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-}
-
-void BeforeStreamingFixture::WaitForTransmittedPackets(int32_t packet_count) {
- transport_->WaitForTransmittedPackets(packet_count);
-}
-
-void BeforeStreamingFixture::SetUpLocalPlayback() {
- transport_ = new LoopBackTransport(voe_network_, channel_);
- EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, *transport_));
-
- webrtc::CodecInst codec;
- codec.channels = 1;
- codec.pacsize = 160;
- codec.plfreq = 8000;
- codec.pltype = 0;
- codec.rate = 64000;
-#if defined(_MSC_VER) && defined(_WIN32)
- _snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE - 1, "PCMU");
-#else
- snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE, "PCMU");
-#endif
- voe_codec_->SetSendCodec(channel_, codec);
-}
diff --git a/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h b/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h
deleted file mode 100644
index 4d49258..0000000
--- a/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_BEFORE_STREAMING_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_BEFORE_STREAMING_H_
-
-#include <string>
-#include "voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-
-// This fixture will, in addition to the work done by its superclasses,
-// create a channel and prepare playing a file through the fake microphone
-// to simulate microphone input. The purpose is to make it convenient
-// to write tests that require microphone input.
-class BeforeStreamingFixture : public AfterInitializationFixture {
- public:
- BeforeStreamingFixture();
- virtual ~BeforeStreamingFixture();
-
- protected:
- int channel_;
- std::string fake_microphone_input_file_;
-
- // Shuts off the fake microphone for this test.
- void SwitchToManualMicrophone();
-
- // Restarts the fake microphone if it's been shut off earlier.
- void RestartFakeMicrophone();
-
- // Stops all sending and playout.
- void PausePlaying();
-
- // Resumes all sending and playout.
- void ResumePlaying();
-
- // Waits until packet_count packetes have been processed by recipient.
- void WaitForTransmittedPackets(int32_t packet_count);
-
- private:
- void SetUpLocalPlayback();
-
- LoopBackTransport* transport_;
-};
-
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_BEFORE_STREAMING_H_
diff --git a/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc b/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
deleted file mode 100644
index 969aad1..0000000
--- a/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * 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.
- */
-
-#include "voice_engine/channel_proxy.h"
-#include "voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-#include "voice_engine/voice_engine_impl.h"
-
-class CodecBeforeStreamingTest : public AfterInitializationFixture {
- protected:
- void SetUp() {
- memset(&codec_instance_, 0, sizeof(codec_instance_));
- codec_instance_.channels = 1;
- codec_instance_.plfreq = 16000;
- codec_instance_.pacsize = 480;
-
- channel_ = voe_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(voice_engine_)
- ->GetChannelProxy(channel_)
- ->RegisterLegacyReceiveCodecs();
- }
-
- void TearDown() {
- voe_base_->DeleteChannel(channel_);
- }
-
- webrtc::CodecInst codec_instance_;
- int channel_;
-};
-
-// TODO(phoglund): add test which verifies default pltypes for various codecs.
-
-TEST_F(CodecBeforeStreamingTest, GetRecPayloadTypeFailsForInvalidCodecName) {
- strcpy(codec_instance_.plname, "SomeInvalidCodecName");
-
- // Should fail since the codec name is invalid.
- EXPECT_NE(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-}
-
-TEST_F(CodecBeforeStreamingTest, GetRecPayloadTypeRecognizesISAC) {
- strcpy(codec_instance_.plname, "iSAC");
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
- strcpy(codec_instance_.plname, "ISAC");
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-}
-
-TEST_F(CodecBeforeStreamingTest, SetRecPayloadTypeCanChangeISACPayloadType) {
- strcpy(codec_instance_.plname, "ISAC");
- codec_instance_.rate = 32000;
-
- codec_instance_.pltype = 123;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(123, codec_instance_.pltype);
-
- codec_instance_.pltype = 104;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-
- EXPECT_EQ(104, codec_instance_.pltype);
-}
-
-TEST_F(CodecBeforeStreamingTest, SetRecPayloadTypeCanChangeILBCPayloadType) {
- strcpy(codec_instance_.plname, "iLBC");
- codec_instance_.plfreq = 8000;
- codec_instance_.pacsize = 240;
- codec_instance_.rate = 13300;
-
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
- int original_pltype = codec_instance_.pltype;
- codec_instance_.pltype = 123;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-
- EXPECT_EQ(123, codec_instance_.pltype);
-
- codec_instance_.pltype = original_pltype;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-
- EXPECT_EQ(original_pltype, codec_instance_.pltype);
-}
diff --git a/voice_engine/test/auto_test/standard/codec_test.cc b/voice_engine/test/auto_test/standard/codec_test.cc
deleted file mode 100644
index 2d979e7..0000000
--- a/voice_engine/test/auto_test/standard/codec_test.cc
+++ /dev/null
@@ -1,216 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-#include <string>
-
-#include "test/testsupport/fileutils.h"
-#include "voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
-#include "voice_engine/voice_engine_defines.h"
-
-class CodecTest : public AfterStreamingFixture {
- protected:
- void SetUp() {
- memset(&codec_instance_, 0, sizeof(codec_instance_));
- apm_ = webrtc::AudioProcessing::Create();
- voe_base_->Init(nullptr, apm_.get(), nullptr);
- }
-
- void SetArbitrarySendCodec() {
- // Just grab the first codec.
- EXPECT_EQ(0, voe_codec_->GetCodec(0, codec_instance_));
- EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, codec_instance_));
- }
-
- rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
- webrtc::CodecInst codec_instance_;
-};
-
-static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) {
- if (!STR_CASE_CMP(codec_instance->plname, "ilbc")) {
- if (packet_size == 160 || packet_size == 320) {
- codec_instance->rate = 15200;
- } else {
- codec_instance->rate = 13300;
- }
- }
-}
-
-static bool IsNotViableSendCodec(const char* codec_name) {
- return !STR_CASE_CMP(codec_name, "CN") ||
- !STR_CASE_CMP(codec_name, "telephone-event") ||
- !STR_CASE_CMP(codec_name, "red");
-}
-
-TEST_F(CodecTest, PcmuIsDefaultCodecAndHasTheRightValues) {
- EXPECT_EQ(0, voe_codec_->GetSendCodec(channel_, codec_instance_));
- EXPECT_EQ(1u, codec_instance_.channels);
- EXPECT_EQ(160, codec_instance_.pacsize);
- EXPECT_EQ(8000, codec_instance_.plfreq);
- EXPECT_EQ(0, codec_instance_.pltype);
- EXPECT_EQ(64000, codec_instance_.rate);
- EXPECT_STRCASEEQ("PCMU", codec_instance_.plname);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionIsOffByDefault) {
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
-
- EXPECT_FALSE(vad_enabled);
- EXPECT_TRUE(dtx_disabled);
- EXPECT_EQ(webrtc::kVadConventional, vad_mode);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionCanBeEnabled) {
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
-
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
-
- EXPECT_TRUE(vad_enabled);
- EXPECT_EQ(webrtc::kVadConventional, vad_mode);
- EXPECT_FALSE(dtx_disabled);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionTypeSettingsCanBeChanged) {
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, true, webrtc::kVadAggressiveLow, false));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadAggressiveLow);
- EXPECT_FALSE(dtx_disabled);
-
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, true, webrtc::kVadAggressiveMid, false));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadAggressiveMid);
- EXPECT_FALSE(dtx_disabled);
-
- // The fourth argument is the DTX disable flag, which is always supposed to
- // be false.
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true,
- webrtc::kVadAggressiveHigh, false));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadAggressiveHigh);
- EXPECT_FALSE(dtx_disabled);
-
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true,
- webrtc::kVadConventional, false));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadConventional);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionCanBeTurnedOff) {
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
-
- // VAD is always on when DTX is on, so we need to turn off DTX too.
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, false, webrtc::kVadConventional, true));
-
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
-
- EXPECT_FALSE(vad_enabled);
- EXPECT_TRUE(dtx_disabled);
- EXPECT_EQ(webrtc::kVadConventional, vad_mode);
-}
-
-TEST_F(CodecTest, OpusMaxPlaybackRateCanBeSet) {
- for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
- voe_codec_->GetCodec(i, codec_instance_);
- if (STR_CASE_CMP("opus", codec_instance_.plname)) {
- continue;
- }
- voe_codec_->SetSendCodec(channel_, codec_instance_);
- // SetOpusMaxPlaybackRate can handle any integer as the bandwidth. Following
- // tests some most commonly used numbers.
- EXPECT_EQ(0, voe_codec_->SetOpusMaxPlaybackRate(channel_, 48000));
- EXPECT_EQ(0, voe_codec_->SetOpusMaxPlaybackRate(channel_, 32000));
- EXPECT_EQ(0, voe_codec_->SetOpusMaxPlaybackRate(channel_, 16000));
- EXPECT_EQ(0, voe_codec_->SetOpusMaxPlaybackRate(channel_, 8000));
- }
-}
-
-TEST_F(CodecTest, OpusDtxCanBeSetForOpus) {
- for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
- voe_codec_->GetCodec(i, codec_instance_);
- if (STR_CASE_CMP("opus", codec_instance_.plname)) {
- continue;
- }
- voe_codec_->SetSendCodec(channel_, codec_instance_);
- EXPECT_EQ(0, voe_codec_->SetOpusDtx(channel_, false));
- EXPECT_EQ(0, voe_codec_->SetOpusDtx(channel_, true));
- }
-}
-
-TEST_F(CodecTest, OpusDtxCannotBeSetForNonOpus) {
- for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
- voe_codec_->GetCodec(i, codec_instance_);
- if (!STR_CASE_CMP("opus", codec_instance_.plname)) {
- continue;
- }
- voe_codec_->SetSendCodec(channel_, codec_instance_);
- EXPECT_EQ(-1, voe_codec_->SetOpusDtx(channel_, true));
- }
-}
-
-// TODO(xians, phoglund): Re-enable when issue 372 is resolved.
-TEST_F(CodecTest, DISABLED_ManualVerifySendCodecsForAllPacketSizes) {
- for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
- voe_codec_->GetCodec(i, codec_instance_);
- if (IsNotViableSendCodec(codec_instance_.plname)) {
- TEST_LOG("Skipping %s.\n", codec_instance_.plname);
- continue;
- }
- EXPECT_NE(-1, codec_instance_.pltype) <<
- "The codec database should suggest a payload type.";
-
- // Test with default packet size:
- TEST_LOG("%s (pt=%d): default packet size(%d), accepts sizes ",
- codec_instance_.plname, codec_instance_.pltype,
- codec_instance_.pacsize);
- voe_codec_->SetSendCodec(channel_, codec_instance_);
- Sleep(CODEC_TEST_TIME);
-
- // Now test other reasonable packet sizes:
- bool at_least_one_succeeded = false;
- for (int packet_size = 80; packet_size < 1000; packet_size += 80) {
- SetRateIfILBC(&codec_instance_, packet_size);
- codec_instance_.pacsize = packet_size;
-
- if (voe_codec_->SetSendCodec(channel_, codec_instance_) != -1) {
- // Note that it's fine for SetSendCodec to fail - what packet sizes
- // it accepts depends on the codec. It should accept one at minimum.
- TEST_LOG("%d ", packet_size);
- TEST_LOG_FLUSH;
- at_least_one_succeeded = true;
- Sleep(CODEC_TEST_TIME);
- }
- }
- TEST_LOG("\n");
- EXPECT_TRUE(at_least_one_succeeded);
- }
-}
diff --git a/voice_engine/test/auto_test/standard/dtmf_test.cc b/voice_engine/test/auto_test/standard/dtmf_test.cc
deleted file mode 100644
index cc5fec2..0000000
--- a/voice_engine/test/auto_test/standard/dtmf_test.cc
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * 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.
- */
-
-#include "voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
-#include "voice_engine/voice_engine_defines.h"
-
-class DtmfTest : public AfterStreamingFixture {
- protected:
- void RunSixteenDtmfEvents() {
- TEST_LOG("Sending telephone events:\n");
- for (int i = 0; i < 16; i++) {
- TEST_LOG("%d ", i);
- TEST_LOG_FLUSH;
- EXPECT_TRUE(channel_proxy_->SendTelephoneEventOutband(i, 160));
- Sleep(500);
- }
- TEST_LOG("\n");
- }
-};
-
-TEST_F(DtmfTest, ManualSuccessfullySendsOutOfBandTelephoneEvents) {
- RunSixteenDtmfEvents();
-}
-
-TEST_F(DtmfTest, TestTwoNonDtmfEvents) {
- EXPECT_TRUE(channel_proxy_->SendTelephoneEventOutband(32, 160));
- EXPECT_TRUE(channel_proxy_->SendTelephoneEventOutband(110, 160));
-}
-
-// This test modifies the DTMF payload type from the default 106 to 88
-// and then runs through 16 DTMF out.of-band events.
-TEST_F(DtmfTest, ManualCanChangeDtmfPayloadType) {
- webrtc::CodecInst codec_instance = webrtc::CodecInst();
-
- TEST_LOG("Changing DTMF payload type.\n");
-
- // Start by modifying the receiving side.
- for (int i = 0; i < voe_codec_->NumOfCodecs(); i++) {
- EXPECT_EQ(0, voe_codec_->GetCodec(i, codec_instance));
- if (!STR_CASE_CMP("telephone-event", codec_instance.plname)) {
- codec_instance.pltype = 88; // Use 88 instead of default 106.
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance));
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- break;
- }
- }
-
- Sleep(500);
-
- // Next, we must modify the sending side as well.
- EXPECT_TRUE(
- channel_proxy_->SetSendTelephoneEventPayloadType(codec_instance.pltype,
- codec_instance.plfreq));
-
- RunSixteenDtmfEvents();
-}
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
deleted file mode 100644
index dc01d90..0000000
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (c) 2012 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 "voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-
-using namespace webrtc;
-using namespace testing;
-
-class RtpRtcpBeforeStreamingTest : public AfterInitializationFixture {
- protected:
- void SetUp();
- void TearDown();
-
- int channel_;
-};
-
-void RtpRtcpBeforeStreamingTest::SetUp() {
- EXPECT_THAT(channel_ = voe_base_->CreateChannel(), Not(Lt(0)));
-}
-
-void RtpRtcpBeforeStreamingTest::TearDown() {
- EXPECT_EQ(0, voe_base_->DeleteChannel(channel_));
-}
-
-TEST_F(RtpRtcpBeforeStreamingTest,
- GetRtcpStatusReturnsTrueByDefaultAndObeysSetRtcpStatus) {
- bool on = false;
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
- EXPECT_TRUE(on);
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, false));
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
- EXPECT_FALSE(on);
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, true));
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
- EXPECT_TRUE(on);
-}
-
-TEST_F(RtpRtcpBeforeStreamingTest, GetLocalSsrcObeysSetLocalSsrc) {
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, 1234));
- unsigned int result = 0;
- EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, result));
- EXPECT_EQ(1234u, result);
-}
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
deleted file mode 100644
index d4692f5..0000000
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * 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());
-}
-
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
deleted file mode 100644
index 2e19527..0000000
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2012 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 "rtc_base/criticalsection.h"
-#include "rtc_base/flags.h"
-#include "system_wrappers/include/event_wrapper.h"
-#include "test/testsupport/fileutils.h"
-#include "voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
-#include "voice_engine/test/auto_test/voe_standard_test.h"
-
-DECLARE_bool(include_timing_dependent_tests);
-
-class TestRtpObserver : public webrtc::VoERTPObserver {
- public:
- TestRtpObserver() : changed_ssrc_event_(webrtc::EventWrapper::Create()) {}
- virtual ~TestRtpObserver() {}
- virtual void OnIncomingCSRCChanged(int channel,
- unsigned int CSRC,
- bool added) {}
- virtual void OnIncomingSSRCChanged(int channel,
- unsigned int SSRC);
- void WaitForChangedSsrc() {
- // 10 seconds should be enough.
- EXPECT_EQ(webrtc::kEventSignaled, changed_ssrc_event_->Wait(10*1000));
- }
- void SetIncomingSsrc(unsigned int ssrc) {
- rtc::CritScope lock(&crit_);
- incoming_ssrc_ = ssrc;
- }
- public:
- rtc::CriticalSection crit_;
- unsigned int incoming_ssrc_;
- std::unique_ptr<webrtc::EventWrapper> changed_ssrc_event_;
-};
-
-void TestRtpObserver::OnIncomingSSRCChanged(int channel,
- unsigned int SSRC) {
- char msg[128];
- sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel,
- SSRC);
- TEST_LOG("%s", msg);
-
- {
- rtc::CritScope lock(&crit_);
- if (incoming_ssrc_ == SSRC)
- changed_ssrc_event_->Set();
- }
-}
-
-static const char* const RTCP_CNAME = "Whatever";
-
-class RtpRtcpTest : public AfterStreamingFixture {
- protected:
- void SetUp() {
- // We need a second channel for this test, so set it up.
- second_channel_ = voe_base_->CreateChannel();
- EXPECT_GE(second_channel_, 0);
-
- transport_ = new LoopBackTransport(voe_network_, second_channel_);
- EXPECT_EQ(0, voe_network_->RegisterExternalTransport(second_channel_,
- *transport_));
-
- EXPECT_EQ(0, voe_base_->StartPlayout(second_channel_));
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(second_channel_, 5678));
- EXPECT_EQ(0, voe_base_->StartSend(second_channel_));
-
- // We'll set up the RTCP CNAME and SSRC to something arbitrary here.
- voe_rtp_rtcp_->SetRTCP_CNAME(channel_, RTCP_CNAME);
- }
-
- void TearDown() {
- EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(second_channel_));
- voe_base_->DeleteChannel(second_channel_);
- delete transport_;
- }
-
- int second_channel_;
- LoopBackTransport* transport_;
-};
-
-TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
- if (!FLAG_include_timing_dependent_tests) {
- TEST_LOG("Skipping test - running in slow execution environment...\n");
- return;
- }
-
- // We need to sleep a bit here for the name to propagate. For
- // instance, 200 milliseconds is not enough, 1 second still flaky,
- // so we'll go with five seconds here.
- Sleep(5000);
-
- char char_buffer[256];
- voe_rtp_rtcp_->GetRemoteRTCP_CNAME(channel_, char_buffer);
- EXPECT_STREQ(RTCP_CNAME, char_buffer);
-}
-
-TEST_F(RtpRtcpTest, SSRCPropagatesCorrectly) {
- unsigned int local_ssrc = 1234;
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, local_ssrc));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- Sleep(1000);
-
- unsigned int ssrc;
- EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc));
- EXPECT_EQ(local_ssrc, ssrc);
-
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc));
- EXPECT_EQ(local_ssrc, ssrc);
-}
diff --git a/voice_engine/test/auto_test/voe_standard_test.cc b/voice_engine/test/auto_test/voe_standard_test.cc
deleted file mode 100644
index c9dba31..0000000
--- a/voice_engine/test/auto_test/voe_standard_test.cc
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2012 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 "voice_engine/test/auto_test/voe_standard_test.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <string.h>
-
-#include "rtc_base/flags.h"
-#include "system_wrappers/include/event_wrapper.h"
-#include "typedefs.h" // NOLINT(build/include)
-#include "voice_engine/test/auto_test/automated_mode.h"
-#include "voice_engine/test/auto_test/voe_test_defines.h"
-#include "voice_engine/voice_engine_defines.h"
-
-DEFINE_bool(include_timing_dependent_tests, true,
- "If true, we will include tests / parts of tests that are known "
- "to break in slow execution environments (such as valgrind).");
-DEFINE_bool(automated, false,
- "If true, we'll run the automated tests we have in noninteractive "
- "mode.");
-DEFINE_bool(help, false, "Print this message.");
-
-namespace webrtc {
-namespace voetest {
-
-int dummy = 0; // Dummy used in different functions to avoid warnings
-
-void SubAPIManager::DisplayStatus() const {
- TEST_LOG("Supported sub APIs:\n\n");
- if (_base)
- TEST_LOG(" Base\n");
- if (_codec)
- TEST_LOG(" Codec\n");
- if (_file)
- TEST_LOG(" File\n");
- if (_hardware)
- TEST_LOG(" Hardware\n");
- if (_network)
- TEST_LOG(" Network\n");
- if (_rtp_rtcp)
- TEST_LOG(" RTP_RTCP\n");
- if (_apm)
- TEST_LOG(" AudioProcessing\n");
- ANL();
- TEST_LOG("Excluded sub APIs:\n\n");
- if (!_base)
- TEST_LOG(" Base\n");
- if (!_codec)
- TEST_LOG(" Codec\n");
- if (!_file)
- TEST_LOG(" File\n");
- if (!_hardware)
- TEST_LOG(" Hardware\n");
- if (!_network)
- TEST_LOG(" Network\n");
- if (!_rtp_rtcp)
- TEST_LOG(" RTP_RTCP\n");
- if (!_apm)
- TEST_LOG(" AudioProcessing\n");
- ANL();
-}
-
-int RunInManualMode() {
- SubAPIManager api_manager;
- api_manager.DisplayStatus();
-
- printf("----------------------------\n");
- printf("Select type of test\n\n");
- printf(" (0) Quit\n");
- printf(" (1) Standard test\n");
- printf("\n: ");
-
- int selection(0);
- dummy = scanf("%d", &selection);
-
- switch (selection) {
- case 0:
- return 0;
- case 1:
- TEST_LOG("\n\n+++ Running standard tests +++\n\n");
- // Currently, all googletest-rewritten tests are in the "automated" suite.
- return RunInAutomatedMode();
- default:
- TEST_LOG("Invalid selection!\n");
- return 0;
- }
-}
-
-} // namespace voetest
-} // namespace webrtc
-
-#if !defined(WEBRTC_IOS)
-int main(int argc, char** argv) {
- // This function and RunInAutomatedMode is defined in automated_mode.cc
- // to avoid macro clashes with googletest (for instance ASSERT_TRUE).
- webrtc::voetest::InitializeGoogleTest(&argc, argv);
- if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
- return 1;
- }
- if (FLAG_help) {
- rtc::FlagList::Print(nullptr, false);
- return 0;
- }
-
- if (FLAG_automated) {
- return webrtc::voetest::RunInAutomatedMode();
- }
-
- return webrtc::voetest::RunInManualMode();
-}
-#endif //#if !defined(WEBRTC_IOS)
diff --git a/voice_engine/test/auto_test/voe_standard_test.h b/voice_engine/test/auto_test/voe_standard_test.h
deleted file mode 100644
index 8888d53..0000000
--- a/voice_engine/test/auto_test/voe_standard_test.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2012 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 VOICE_ENGINE_VOE_STANDARD_TEST_H_
-#define VOICE_ENGINE_VOE_STANDARD_TEST_H_
-
-#include <stdio.h>
-#include <string>
-
-#include "test/testsupport/fileutils.h"
-#include "voice_engine/test/auto_test/voe_test_common.h"
-
-namespace webrtc {
-namespace voetest {
-
-class SubAPIManager {
- public:
- SubAPIManager()
- : _base(true),
- _codec(false),
- _file(false),
- _hardware(false),
- _network(false),
- _rtp_rtcp(false),
- _apm(false) {
- _codec = true;
- _file = true;
- _hardware = true;
- _network = true;
- _rtp_rtcp = true;
- _apm = true;
- }
-
- void DisplayStatus() const;
-
- private:
- bool _base, _codec;
- bool _file, _hardware;
- bool _network, _rtp_rtcp, _apm;
-};
-
-} // namespace voetest
-} // namespace webrtc
-
-#endif // VOICE_ENGINE_VOE_STANDARD_TEST_H_
diff --git a/voice_engine/test/auto_test/voe_test_common.h b/voice_engine/test/auto_test/voe_test_common.h
deleted file mode 100644
index 5af710d..0000000
--- a/voice_engine/test/auto_test/voe_test_common.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (c) 2012 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 VOICE_ENGINE_VOE_TEST_COMMON_H_
-#define VOICE_ENGINE_VOE_TEST_COMMON_H_
-
-#ifdef WEBRTC_ANDROID
-#include <android/log.h>
-#define ANDROID_LOG_TAG "VoiceEngine Auto Test"
-#define TEST_LOG(...) \
- __android_log_print(ANDROID_LOG_DEBUG, ANDROID_LOG_TAG, __VA_ARGS__)
-#define TEST_LOG_ERROR(...) \
- __android_log_print(ANDROID_LOG_ERROR, ANDROID_LOG_TAG, __VA_ARGS__)
-#define TEST_LOG_FLUSH
-#else
-#define TEST_LOG printf
-#define TEST_LOG_ERROR printf
-#define TEST_LOG_FLUSH fflush(NULL)
-#endif
-
-// Time in ms to test each packet size for each codec
-#define CODEC_TEST_TIME 400
-
-#endif // VOICE_ENGINE_VOE_TEST_COMMON_H_
diff --git a/voice_engine/test/auto_test/voe_test_defines.h b/voice_engine/test/auto_test/voe_test_defines.h
deleted file mode 100644
index 6b8c0c7..0000000
--- a/voice_engine/test/auto_test/voe_test_defines.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2012 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 VOICE_ENGINE_VOE_TEST_DEFINES_H_
-#define VOICE_ENGINE_VOE_TEST_DEFINES_H_
-
-#include "voice_engine/test/auto_test/voe_test_common.h"
-
-// Select the tests to execute, list order below is same as they will be
-// executed. Note that, all settings below will be overriden by sub-API
-// settings in voice_engine_configurations.h.
-#define _TEST_BASE_
-#define _TEST_RTP_RTCP_
-#define _TEST_CODEC_
-#define _TEST_FILE_
-#define _TEST_NETWORK_
-
-// Enable this when running instrumentation of some kind to exclude tests
-// that will not pass due to slowed down execution.
-// #define _INSTRUMENTATION_TESTING_
-
-// Some parts can cause problems while running Insure
-#ifdef __INSURE__
-#define _INSTRUMENTATION_TESTING_
-#endif
-
-#define MARK() TEST_LOG("."); fflush(NULL); // Add test marker
-#define ANL() TEST_LOG("\n") // Add New Line
-#define AOK() TEST_LOG("[Test is OK]"); fflush(NULL); // Add OK
-#if defined(_WIN32)
-#define PAUSE \
- { \
- TEST_LOG("Press any key to continue..."); \
- _getch(); \
- TEST_LOG("\n"); \
- }
-#else
-#define PAUSE \
- { \
- TEST_LOG("Continuing (pause not supported)\n"); \
- }
-#endif
-
-#define TEST(s) \
- { \
- TEST_LOG("Testing: %s", #s); \
- } \
-
-#ifdef _INSTRUMENTATION_TESTING_
-// Don't stop execution if error occurs
-#define TEST_MUSTPASS(expr) \
- { \
- if ((expr)) \
- { \
- TEST_LOG_ERROR("Error at line:%i, %s \n",__LINE__, #expr); \
- TEST_LOG_ERROR("Error code: %i\n",voe_base_->LastError()); \
- } \
- }
-#define TEST_ERROR(code) \
- { \
- int err = voe_base_->LastError(); \
- if (err != code) \
- { \
- TEST_LOG_ERROR("Invalid error code (%d, should be %d) at line %d\n",
- code, err, __LINE__);
-}
-}
-#else
-#define ASSERT_TRUE(expr) TEST_MUSTPASS(!(expr))
-#define ASSERT_FALSE(expr) TEST_MUSTPASS(expr)
-#define TEST_MUSTFAIL(expr) TEST_MUSTPASS(!((expr) == -1))
-#define TEST_MUSTPASS(expr) \
- { \
- if ((expr)) \
- { \
- TEST_LOG_ERROR("\nError at line:%i, %s \n",__LINE__, #expr); \
- TEST_LOG_ERROR("Error code: %i\n", voe_base_->LastError()); \
- PAUSE \
- return -1; \
- } \
- }
-#define TEST_ERROR(code) \
- { \
- int err = voe_base_->LastError(); \
- if (err != code) \
- { \
- TEST_LOG_ERROR("Invalid error code (%d, should be %d) at line %d\n", \
- err, code, __LINE__); \
- PAUSE \
- return -1; \
- } \
- }
-#endif // #ifdef _INSTRUMENTATION_TESTING_
-#define EXCLUDE() \
- { \
- TEST_LOG("\n>>> Excluding test at line: %i <<<\n\n",__LINE__); \
- }
-
-#define INCOMPLETE() \
- { \
- TEST_LOG("\n>>> Incomplete test at line: %i <<<\n\n",__LINE__); \
- }
-
-#endif // VOICE_ENGINE_VOE_TEST_DEFINES_H_