| /* |
| * 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 "modules/audio_coding/test/TestVADDTX.h" |
| |
| #include <string> |
| |
| #include "absl/strings/match.h" |
| #include "absl/strings/string_view.h" |
| #include "api/audio_codecs/audio_decoder_factory_template.h" |
| #include "api/audio_codecs/audio_encoder_factory_template.h" |
| #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" |
| #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" |
| #include "api/audio_codecs/opus/audio_decoder_opus.h" |
| #include "api/audio_codecs/opus/audio_encoder_opus.h" |
| #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" |
| #include "modules/audio_coding/test/PCMFile.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/file_utils.h" |
| |
| namespace webrtc { |
| |
| MonitoringAudioPacketizationCallback::MonitoringAudioPacketizationCallback( |
| AudioPacketizationCallback* next) |
| : next_(next) { |
| ResetStatistics(); |
| } |
| |
| int32_t MonitoringAudioPacketizationCallback::SendData( |
| AudioFrameType frame_type, |
| uint8_t payload_type, |
| uint32_t timestamp, |
| const uint8_t* payload_data, |
| size_t payload_len_bytes, |
| int64_t absolute_capture_timestamp_ms) { |
| counter_[static_cast<int>(frame_type)]++; |
| return next_->SendData(frame_type, payload_type, timestamp, payload_data, |
| payload_len_bytes, absolute_capture_timestamp_ms); |
| } |
| |
| void MonitoringAudioPacketizationCallback::PrintStatistics() { |
| printf("\n"); |
| printf("kEmptyFrame %u\n", |
| counter_[static_cast<int>(AudioFrameType::kEmptyFrame)]); |
| printf("kAudioFrameSpeech %u\n", |
| counter_[static_cast<int>(AudioFrameType::kAudioFrameSpeech)]); |
| printf("kAudioFrameCN %u\n", |
| counter_[static_cast<int>(AudioFrameType::kAudioFrameCN)]); |
| printf("\n\n"); |
| } |
| |
| void MonitoringAudioPacketizationCallback::ResetStatistics() { |
| memset(counter_, 0, sizeof(counter_)); |
| } |
| |
| void MonitoringAudioPacketizationCallback::GetStatistics(uint32_t* counter) { |
| memcpy(counter, counter_, sizeof(counter_)); |
| } |
| |
| TestVadDtx::TestVadDtx() |
| : encoder_factory_( |
| CreateAudioEncoderFactory<AudioEncoderIlbc, AudioEncoderOpus>()), |
| decoder_factory_( |
| CreateAudioDecoderFactory<AudioDecoderIlbc, AudioDecoderOpus>()), |
| acm_send_(AudioCodingModule::Create()), |
| acm_receive_(std::make_unique<acm2::AcmReceiver>( |
| acm2::AcmReceiver::Config(decoder_factory_))), |
| channel_(std::make_unique<Channel>()), |
| packetization_callback_( |
| std::make_unique<MonitoringAudioPacketizationCallback>( |
| channel_.get())) { |
| EXPECT_EQ( |
| 0, acm_send_->RegisterTransportCallback(packetization_callback_.get())); |
| channel_->RegisterReceiverACM(acm_receive_.get()); |
| } |
| |
| bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format, |
| absl::optional<Vad::Aggressiveness> vad_mode) { |
| constexpr int payload_type = 17, cn_payload_type = 117; |
| bool added_comfort_noise = false; |
| |
| auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format, |
| absl::nullopt); |
| if (vad_mode.has_value() && |
| !absl::EqualsIgnoreCase(codec_format.name, "opus")) { |
| AudioEncoderCngConfig config; |
| config.speech_encoder = std::move(encoder); |
| config.num_channels = 1; |
| config.payload_type = cn_payload_type; |
| config.vad_mode = vad_mode.value(); |
| encoder = CreateComfortNoiseEncoder(std::move(config)); |
| added_comfort_noise = true; |
| } |
| channel_->SetIsStereo(encoder->NumChannels() > 1); |
| acm_send_->SetEncoder(std::move(encoder)); |
| |
| std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}}; |
| acm_receive_->SetCodecs(receive_codecs); |
| |
| return added_comfort_noise; |
| } |
| |
| // Encoding a file and see if the numbers that various packets occur follow |
| // the expectation. |
| void TestVadDtx::Run(absl::string_view in_filename, |
| int frequency, |
| int channels, |
| absl::string_view out_filename, |
| bool append, |
| const int* expects) { |
| packetization_callback_->ResetStatistics(); |
| |
| PCMFile in_file; |
| in_file.Open(in_filename, frequency, "rb"); |
| in_file.ReadStereo(channels > 1); |
| // Set test length to 1000 ms (100 blocks of 10 ms each). |
| in_file.SetNum10MsBlocksToRead(100); |
| // Fast-forward both files 500 ms (50 blocks). The first second of the file is |
| // silence, but we want to keep half of that to test silence periods. |
| in_file.FastForward(50); |
| |
| PCMFile out_file; |
| if (append) { |
| out_file.Open(out_filename, kOutputFreqHz, "ab"); |
| } else { |
| out_file.Open(out_filename, kOutputFreqHz, "wb"); |
| } |
| |
| uint16_t frame_size_samples = in_file.PayloadLength10Ms(); |
| AudioFrame audio_frame; |
| while (!in_file.EndOfFile()) { |
| in_file.Read10MsData(audio_frame); |
| audio_frame.timestamp_ = time_stamp_; |
| time_stamp_ += frame_size_samples; |
| EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0); |
| bool muted; |
| acm_receive_->GetAudio(kOutputFreqHz, &audio_frame, &muted); |
| ASSERT_FALSE(muted); |
| out_file.Write10MsData(audio_frame); |
| } |
| |
| in_file.Close(); |
| out_file.Close(); |
| |
| #ifdef PRINT_STAT |
| packetization_callback_->PrintStatistics(); |
| #endif |
| |
| uint32_t stats[3]; |
| packetization_callback_->GetStatistics(stats); |
| packetization_callback_->ResetStatistics(); |
| |
| for (const auto& st : stats) { |
| int i = &st - stats; // Calculate the current position in stats. |
| switch (expects[i]) { |
| case 0: { |
| EXPECT_EQ(0u, st) << "stats[" << i << "] error."; |
| break; |
| } |
| case 1: { |
| EXPECT_GT(st, 0u) << "stats[" << i << "] error."; |
| break; |
| } |
| } |
| } |
| } |
| |
| // Following is the implementation of TestWebRtcVadDtx. |
| TestWebRtcVadDtx::TestWebRtcVadDtx() : output_file_num_(0) {} |
| |
| void TestWebRtcVadDtx::Perform() { |
| // TODO(bugs.webrtc.org/345525069): Either fix/enable or remove iLBC. |
| #if defined(__has_feature) && !__has_feature(undefined_behavior_sanitizer) |
| RunTestCases({"ILBC", 8000, 1}); |
| #endif |
| RunTestCases({"opus", 48000, 2}); |
| } |
| |
| // Test various configurations on VAD/DTX. |
| void TestWebRtcVadDtx::RunTestCases(const SdpAudioFormat& codec_format) { |
| Test(/*new_outfile=*/true, |
| /*expect_dtx_enabled=*/RegisterCodec(codec_format, absl::nullopt)); |
| |
| Test(/*new_outfile=*/false, |
| /*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadAggressive)); |
| |
| Test(/*new_outfile=*/false, |
| /*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadLowBitrate)); |
| |
| Test(/*new_outfile=*/false, /*expect_dtx_enabled=*/RegisterCodec( |
| codec_format, Vad::kVadVeryAggressive)); |
| |
| Test(/*new_outfile=*/false, |
| /*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadNormal)); |
| } |
| |
| // Set the expectation and run the test. |
| void TestWebRtcVadDtx::Test(bool new_outfile, bool expect_dtx_enabled) { |
| int expects[] = {-1, 1, expect_dtx_enabled, 0, 0}; |
| if (new_outfile) { |
| output_file_num_++; |
| } |
| rtc::StringBuilder out_filename; |
| out_filename << webrtc::test::OutputPath() << "testWebRtcVadDtx_outFile_" |
| << output_file_num_ << ".pcm"; |
| Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, |
| out_filename.str(), !new_outfile, expects); |
| } |
| |
| // Following is the implementation of TestOpusDtx. |
| void TestOpusDtx::Perform() { |
| int expects[] = {0, 1, 0, 0, 0}; |
| |
| // Register Opus as send codec |
| std::string out_filename = |
| webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm"; |
| RegisterCodec({"opus", 48000, 2}, absl::nullopt); |
| acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) { |
| (*encoder_ptr)->SetDtx(false); |
| }); |
| |
| Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, |
| out_filename, false, expects); |
| |
| acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) { |
| (*encoder_ptr)->SetDtx(true); |
| }); |
| expects[static_cast<int>(AudioFrameType::kEmptyFrame)] = 1; |
| expects[static_cast<int>(AudioFrameType::kAudioFrameCN)] = 1; |
| Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, |
| out_filename, true, expects); |
| } |
| |
| } // namespace webrtc |