diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc new file mode 100644 index 0000000000..0f3e73ef51 --- /dev/null +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc @@ -0,0 +1,172 @@ +/* + * 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 "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" +#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/test/testsupport/gtest_disable.h" + +namespace webrtc { + +const int kSampleRateHz = 16000; +const int kNumSamples10ms = kSampleRateHz / 100; +const int kFrameSizeMs = 10; // Multiple of 10. +const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; +const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); +const uint8_t kPayloadType = 111; + +class RtpUtility { + public: + RtpUtility(int samples_per_packet, uint8_t payload_type) + : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} + + virtual ~RtpUtility() {} + + void Populate(WebRtcRTPHeader* rtp_header) { + rtp_header->header.sequenceNumber = 0xABCD; + rtp_header->header.timestamp = 0xABCDEF01; + rtp_header->header.payloadType = payload_type_; + rtp_header->header.markerBit = false; + rtp_header->header.ssrc = 0x1234; + rtp_header->header.numCSRCs = 0; + rtp_header->frameType = kAudioFrameSpeech; + + rtp_header->header.payload_type_frequency = kSampleRateHz; + rtp_header->type.Audio.channel = 1; + rtp_header->type.Audio.isCNG = false; + } + + void Forward(WebRtcRTPHeader* rtp_header) { + ++rtp_header->header.sequenceNumber; + rtp_header->header.timestamp += samples_per_packet_; + } + + private: + int samples_per_packet_; + uint8_t payload_type_; +}; + +class AudioCodingModuleTest : public ::testing::Test { + protected: + AudioCodingModuleTest() + : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), + acm2_(AudioCodingModule::Create(1 /*id*/)) {} + + ~AudioCodingModuleTest() {} + + void TearDown() {} + + void SetUp() { + CodecInst codec; + AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1); + codec.pltype = kPayloadType; + + // Register L16 codec in ACMs. + ASSERT_EQ(0, acm2_->RegisterReceiveCodec(codec)); + + rtp_utility_->Populate(&rtp_header_); + } + + void InsertPacketAndPullAudio() { + AudioFrame audio_frame; + const uint8_t kPayload[kPayloadSizeBytes] = {0}; + + ASSERT_EQ(0, + acm2_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); + + ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame)); + rtp_utility_->Forward(&rtp_header_); + } + + void JustPullAudio() { + AudioFrame audio_frame; + ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame)); + } + + scoped_ptr rtp_utility_; + scoped_ptr acm2_; + + WebRtcRTPHeader rtp_header_; +}; + +// Check if the statistics are initialized correctly. Before any call to ACM +// all fields have to be zero. +TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) { + AudioDecodingCallStats stats; + acm2_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(0, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(0, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); +} + +// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms() +// should result in generating silence, check the associated field. +TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) { + AudioDecodingCallStats stats; + const int kInitialDelay = 100; + + acm2_->SetInitialPlayoutDelay(kInitialDelay); + + AudioFrame audio_frame; + int num_calls = 0; + for (int time_ms = 0; time_ms < kInitialDelay; + time_ms += kFrameSizeMs, ++num_calls) { + InsertPacketAndPullAudio(); + } + acm2_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(0, stats.calls_to_neteq); + EXPECT_EQ(num_calls, stats.calls_to_silence_generator); + EXPECT_EQ(0, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); +} + +// Insert some packets and pull audio. Check statistics are valid. Then, +// simulate packet loss and check if PLC and PLC-to-CNG statistics are +// correctly updated. +TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) { + AudioDecodingCallStats stats; + const int kNumNormalCalls = 10; + + AudioFrame audio_frame; + for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) { + InsertPacketAndPullAudio(); + } + acm2_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); + + const int kNumPlc = 3; + const int kNumPlcCng = 5; + + // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG. + for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) { + JustPullAudio(); + } + acm2_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(kNumPlc, stats.decoded_plc); + EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng); +} + +} // namespace webrtc diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index dafe6bde8b..8fd0714ed7 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -103,6 +103,7 @@ ], 'sources': [ 'audio_coding/main/acm2/acm_receiver_unittest.cc', + 'audio_coding/main/acm2/audio_coding_module_unittest.cc', 'audio_coding/main/acm2/call_statistics_unittest.cc', 'audio_coding/main/acm2/initial_delay_manager_unittest.cc', 'audio_coding/main/acm2/nack_unittest.cc', diff --git a/webrtc/voice_engine/voe_neteq_stats_unittest.cc b/webrtc/voice_engine/voe_neteq_stats_unittest.cc deleted file mode 100644 index 4bb4d4a0fb..0000000000 --- a/webrtc/voice_engine/voe_neteq_stats_unittest.cc +++ /dev/null @@ -1,285 +0,0 @@ -/* - * Copyright (c) 2013 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 "webrtc/voice_engine/include/voe_neteq_stats.h" - -#include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h" -#include "webrtc/modules/audio_device/include/fake_audio_device.h" -#include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/test/testsupport/gtest_disable.h" -#include "webrtc/voice_engine/include/voe_base.h" -#include "webrtc/voice_engine/include/voe_hardware.h" -#include "webrtc/voice_engine/voice_engine_defines.h" - -namespace webrtc { -namespace voe { -namespace { - -const int kSampleRateHz = 16000; -const int kNumSamples10ms = kSampleRateHz / 100; -const int kFrameSizeMs = 10; // Multiple of 10. -const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; -const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); -const uint8_t kPayloadType = 111; - -class RtpUtility { - public: - RtpUtility(int samples_per_packet, uint8_t payload_type) - : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} - - virtual ~RtpUtility() {} - - void Populate(WebRtcRTPHeader* rtp_header) { - rtp_header->header.sequenceNumber = 0xABCD; - rtp_header->header.timestamp = 0xABCDEF01; - rtp_header->header.payloadType = payload_type_; - rtp_header->header.markerBit = false; - rtp_header->header.ssrc = 0x1234; - rtp_header->header.numCSRCs = 0; - rtp_header->frameType = kAudioFrameSpeech; - - rtp_header->header.payload_type_frequency = kSampleRateHz; - rtp_header->type.Audio.channel = 1; - rtp_header->type.Audio.isCNG = false; - } - - void Forward(WebRtcRTPHeader* rtp_header) { - ++rtp_header->header.sequenceNumber; - rtp_header->header.timestamp += samples_per_packet_; - } - - private: - int samples_per_packet_; - uint8_t payload_type_; -}; - -// This factory method allows access to ACM of a channel, facilitating insertion -// of packets to and pulling audio of ACM. -struct InsertAcm : AudioCodingModuleFactory { - explicit InsertAcm(AudioCodingModule* acm) : acm_(acm) {} - ~InsertAcm() {} - virtual AudioCodingModule* Create(int /*id*/) const { return acm_; } - - AudioCodingModule* acm_; -}; - -class VoENetEqStatsTest : public ::testing::Test { - protected: - VoENetEqStatsTest() - : acm1_(new acm1::AudioCodingModuleImpl(1, Clock::GetRealTimeClock())), - acm2_(new acm2::AudioCodingModuleImpl(2)), - voe_(VoiceEngine::Create()), - base_(VoEBase::GetInterface(voe_)), - voe_neteq_stats_(VoENetEqStats::GetInterface(voe_)), - channel_acm1_(-1), - channel_acm2_(-1), - adm_(new FakeAudioDeviceModule), - rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) {} - - ~VoENetEqStatsTest() {} - - void TearDown() { - voe_neteq_stats_->Release(); - base_->DeleteChannel(channel_acm1_); - base_->DeleteChannel(channel_acm2_); - base_->Terminate(); - base_->Release(); - VoiceEngine::Delete(voe_); - } - - void SetUp() { - // Check if all components are valid. - ASSERT_TRUE(voe_ != NULL); - ASSERT_TRUE(base_ != NULL); - ASSERT_TRUE(adm_.get() != NULL); - ASSERT_EQ(0, base_->Init(adm_.get())); - - // Set configs. - config_acm1_.Set(new InsertAcm(acm1_)); - config_acm2_.Set(new InsertAcm(acm2_)); - - // Create channe1s; - channel_acm1_ = base_->CreateChannel(config_acm1_); - ASSERT_NE(-1, channel_acm1_); - - channel_acm2_ = base_->CreateChannel(config_acm2_); - ASSERT_NE(-1, channel_acm2_); - - CodecInst codec; - AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1); - codec.pltype = kPayloadType; - - // Register L16 codec in ACMs. - ASSERT_EQ(0, acm1_->RegisterReceiveCodec(codec)); - ASSERT_EQ(0, acm2_->RegisterReceiveCodec(codec)); - - rtp_utility_->Populate(&rtp_header_); - } - - void InsertPacketAndPullAudio() { - AudioFrame audio_frame; - const uint8_t kPayload[kPayloadSizeBytes] = {0}; - - ASSERT_EQ(0, - acm1_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); - ASSERT_EQ(0, - acm2_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); - - ASSERT_EQ(0, acm1_->PlayoutData10Ms(-1, &audio_frame)); - ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame)); - rtp_utility_->Forward(&rtp_header_); - } - - void JustPullAudio() { - AudioFrame audio_frame; - ASSERT_EQ(0, acm1_->PlayoutData10Ms(-1, &audio_frame)); - ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame)); - } - - Config config_acm1_; - Config config_acm2_; - - // ACMs are inserted into VoE channels, and this class is not the owner of - // them. Therefore, they should not be deleted, not even in destructor. - AudioCodingModule* acm1_; - AudioCodingModule* acm2_; - - VoiceEngine* voe_; - VoEBase* base_; - VoENetEqStats* voe_neteq_stats_; - int channel_acm1_; - int channel_acm2_; - scoped_ptr adm_; - scoped_ptr rtp_utility_; - WebRtcRTPHeader rtp_header_; -}; - -// Check if the statistics are initialized correctly. Before any call to ACM -// all fields have to be zero. -TEST_F(VoENetEqStatsTest, DISABLED_ON_ANDROID(InitializedToZero)) { - AudioDecodingCallStats stats; - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats)); - EXPECT_EQ(0, stats.calls_to_neteq); - EXPECT_EQ(0, stats.calls_to_silence_generator); - EXPECT_EQ(0, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(0, stats.decoded_plc); - EXPECT_EQ(0, stats.decoded_plc_cng); - - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats)); - EXPECT_EQ(0, stats.calls_to_neteq); - EXPECT_EQ(0, stats.calls_to_silence_generator); - EXPECT_EQ(0, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(0, stats.decoded_plc); - EXPECT_EQ(0, stats.decoded_plc_cng); -} - -// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms() -// should result in generating silence, check the associated field. -TEST_F(VoENetEqStatsTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) { - AudioDecodingCallStats stats; - const int kInitialDelay = 100; - - acm1_->SetInitialPlayoutDelay(kInitialDelay); - acm2_->SetInitialPlayoutDelay(kInitialDelay); - - AudioFrame audio_frame; - int num_calls = 0; - for (int time_ms = 0; time_ms < kInitialDelay; - time_ms += kFrameSizeMs, ++num_calls) { - InsertPacketAndPullAudio(); - } - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats)); - EXPECT_EQ(0, stats.calls_to_neteq); - EXPECT_EQ(num_calls, stats.calls_to_silence_generator); - EXPECT_EQ(0, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(0, stats.decoded_plc); - EXPECT_EQ(0, stats.decoded_plc_cng); - - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats)); - EXPECT_EQ(0, stats.calls_to_neteq); - EXPECT_EQ(num_calls, stats.calls_to_silence_generator); - EXPECT_EQ(0, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(0, stats.decoded_plc); - EXPECT_EQ(0, stats.decoded_plc_cng); -} - -// Insert some packets and pull audio. Check statistics are valid. Then, -// simulate packet loss and check if PLC and PLC-to-CNG statistics are -// correctly updated. -TEST_F(VoENetEqStatsTest, DISABLED_ON_ANDROID(NetEqCalls)) { - AudioDecodingCallStats stats; - const int kNumNormalCalls = 10; - - AudioFrame audio_frame; - for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) { - InsertPacketAndPullAudio(); - } - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats)); - EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq); - EXPECT_EQ(0, stats.calls_to_silence_generator); - EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(0, stats.decoded_plc); - EXPECT_EQ(0, stats.decoded_plc_cng); - - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats)); - EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq); - EXPECT_EQ(0, stats.calls_to_silence_generator); - EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(0, stats.decoded_plc); - EXPECT_EQ(0, stats.decoded_plc_cng); - - const int kNumPlc = 3; - const int kNumPlcCng = 5; - - // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG. - for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) { - JustPullAudio(); - } - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm1_, &stats)); - EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq); - EXPECT_EQ(0, stats.calls_to_silence_generator); - EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(kNumPlc, stats.decoded_plc); - EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng); - - ASSERT_EQ(0, - voe_neteq_stats_->GetDecodingCallStatistics(channel_acm2_, &stats)); - EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq); - EXPECT_EQ(0, stats.calls_to_silence_generator); - EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); - EXPECT_EQ(0, stats.decoded_cng); - EXPECT_EQ(kNumPlc, stats.decoded_plc); - EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng); -} - -} // namespace - -} // namespace voe - -} // namespace webrtc diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp index 699677cfb7..9c4f97e134 100644 --- a/webrtc/voice_engine/voice_engine.gyp +++ b/webrtc/voice_engine/voice_engine.gyp @@ -126,7 +126,6 @@ 'voe_audio_processing_unittest.cc', 'voe_base_unittest.cc', 'voe_codec_unittest.cc', - 'voe_neteq_stats_unittest.cc', ], 'conditions': [ # TODO(henrike): remove build_with_chromium==1 when the bots are