| /* |
| * 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 "TwoWayCommunication.h" |
| |
| #include <ctype.h> |
| #include <stdio.h> |
| #include <string.h> |
| |
| #include <memory> |
| |
| #ifdef WIN32 |
| #include <Windows.h> |
| #endif |
| |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "common_types.h" // NOLINT(build/include) |
| #include "modules/audio_coding/codecs/audio_format_conversion.h" |
| #include "modules/audio_coding/test/PCMFile.h" |
| #include "modules/audio_coding/test/utility.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/fileutils.h" |
| |
| namespace webrtc { |
| |
| #define MAX_FILE_NAME_LENGTH_BYTE 500 |
| |
| TwoWayCommunication::TwoWayCommunication() |
| : _acmA(AudioCodingModule::Create( |
| AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))), |
| _acmRefA(AudioCodingModule::Create( |
| AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) { |
| AudioCodingModule::Config config; |
| // The clicks will be more obvious if time-stretching is not allowed. |
| // TODO(henrik.lundin) Really? |
| config.neteq_config.for_test_no_time_stretching = true; |
| config.decoder_factory = CreateBuiltinAudioDecoderFactory(); |
| _acmB.reset(AudioCodingModule::Create(config)); |
| _acmRefB.reset(AudioCodingModule::Create(config)); |
| } |
| |
| TwoWayCommunication::~TwoWayCommunication() { |
| delete _channel_A2B; |
| delete _channel_B2A; |
| delete _channelRef_A2B; |
| delete _channelRef_B2A; |
| _inFileA.Close(); |
| _inFileB.Close(); |
| _outFileA.Close(); |
| _outFileB.Close(); |
| _outFileRefA.Close(); |
| _outFileRefB.Close(); |
| } |
| |
| void TwoWayCommunication::SetUpAutotest() { |
| CodecInst codecInst_A; |
| CodecInst codecInst_B; |
| CodecInst dummyCodec; |
| |
| EXPECT_EQ(0, _acmA->Codec("ISAC", &codecInst_A, 16000, 1)); |
| EXPECT_EQ(0, _acmB->Codec("L16", &codecInst_B, 8000, 1)); |
| EXPECT_EQ(0, _acmA->Codec(6, &dummyCodec)); |
| |
| //--- Set A codecs |
| EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A)); |
| EXPECT_EQ(true, _acmA->RegisterReceiveCodec(codecInst_B.pltype, |
| CodecInstToSdp(codecInst_B))); |
| |
| //--- Set ref-A codecs |
| EXPECT_GT(_acmRefA->RegisterSendCodec(codecInst_A), -1); |
| EXPECT_EQ(true, _acmRefA->RegisterReceiveCodec(codecInst_B.pltype, |
| CodecInstToSdp(codecInst_B))); |
| |
| //--- Set B codecs |
| EXPECT_GT(_acmB->RegisterSendCodec(codecInst_B), -1); |
| EXPECT_EQ(true, _acmB->RegisterReceiveCodec(codecInst_A.pltype, |
| CodecInstToSdp(codecInst_A))); |
| |
| //--- Set ref-B codecs |
| EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B)); |
| EXPECT_EQ(true, _acmRefB->RegisterReceiveCodec(codecInst_A.pltype, |
| CodecInstToSdp(codecInst_A))); |
| |
| uint16_t frequencyHz; |
| |
| //--- Input A and B |
| std::string in_file_name = |
| webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
| frequencyHz = 16000; |
| _inFileA.Open(in_file_name, frequencyHz, "rb"); |
| _inFileB.Open(in_file_name, frequencyHz, "rb"); |
| |
| //--- Output A |
| std::string output_file_a = webrtc::test::OutputPath() + "outAutotestA.pcm"; |
| frequencyHz = 16000; |
| _outFileA.Open(output_file_a, frequencyHz, "wb"); |
| std::string output_ref_file_a = |
| webrtc::test::OutputPath() + "ref_outAutotestA.pcm"; |
| _outFileRefA.Open(output_ref_file_a, frequencyHz, "wb"); |
| |
| //--- Output B |
| std::string output_file_b = webrtc::test::OutputPath() + "outAutotestB.pcm"; |
| frequencyHz = 16000; |
| _outFileB.Open(output_file_b, frequencyHz, "wb"); |
| std::string output_ref_file_b = |
| webrtc::test::OutputPath() + "ref_outAutotestB.pcm"; |
| _outFileRefB.Open(output_ref_file_b, frequencyHz, "wb"); |
| |
| //--- Set A-to-B channel |
| _channel_A2B = new Channel; |
| _acmA->RegisterTransportCallback(_channel_A2B); |
| _channel_A2B->RegisterReceiverACM(_acmB.get()); |
| //--- Do the same for the reference |
| _channelRef_A2B = new Channel; |
| _acmRefA->RegisterTransportCallback(_channelRef_A2B); |
| _channelRef_A2B->RegisterReceiverACM(_acmRefB.get()); |
| |
| //--- Set B-to-A channel |
| _channel_B2A = new Channel; |
| _acmB->RegisterTransportCallback(_channel_B2A); |
| _channel_B2A->RegisterReceiverACM(_acmA.get()); |
| //--- Do the same for reference |
| _channelRef_B2A = new Channel; |
| _acmRefB->RegisterTransportCallback(_channelRef_B2A); |
| _channelRef_B2A->RegisterReceiverACM(_acmRefA.get()); |
| } |
| |
| void TwoWayCommunication::Perform() { |
| SetUpAutotest(); |
| unsigned int msecPassed = 0; |
| unsigned int secPassed = 0; |
| |
| int32_t outFreqHzA = _outFileA.SamplingFrequency(); |
| int32_t outFreqHzB = _outFileB.SamplingFrequency(); |
| |
| AudioFrame audioFrame; |
| |
| auto codecInst_B = _acmB->SendCodec(); |
| ASSERT_TRUE(codecInst_B); |
| |
| // In the following loop we tests that the code can handle misuse of the APIs. |
| // In the middle of a session with data flowing between two sides, called A |
| // and B, APIs will be called, and the code should continue to run, and be |
| // able to recover. |
| while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) { |
| msecPassed += 10; |
| EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0); |
| EXPECT_GE(_acmA->Add10MsData(audioFrame), 0); |
| EXPECT_GE(_acmRefA->Add10MsData(audioFrame), 0); |
| |
| EXPECT_GT(_inFileB.Read10MsData(audioFrame), 0); |
| |
| EXPECT_GE(_acmB->Add10MsData(audioFrame), 0); |
| EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0); |
| bool muted; |
| EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted)); |
| ASSERT_FALSE(muted); |
| _outFileA.Write10MsData(audioFrame); |
| EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted)); |
| ASSERT_FALSE(muted); |
| _outFileRefA.Write10MsData(audioFrame); |
| EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted)); |
| ASSERT_FALSE(muted); |
| _outFileB.Write10MsData(audioFrame); |
| EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted)); |
| ASSERT_FALSE(muted); |
| _outFileRefB.Write10MsData(audioFrame); |
| |
| // Update time counters each time a second of data has passed. |
| if (msecPassed >= 1000) { |
| msecPassed = 0; |
| secPassed++; |
| } |
| // Re-register send codec on side B. |
| if (((secPassed % 5) == 4) && (msecPassed >= 990)) { |
| EXPECT_EQ(0, _acmB->RegisterSendCodec(*codecInst_B)); |
| EXPECT_TRUE(_acmB->SendCodec()); |
| } |
| // Initialize receiver on side A. |
| if (((secPassed % 7) == 6) && (msecPassed == 0)) |
| EXPECT_EQ(0, _acmA->InitializeReceiver()); |
| // Re-register codec on side A. |
| if (((secPassed % 7) == 6) && (msecPassed >= 990)) { |
| EXPECT_EQ(true, _acmA->RegisterReceiveCodec( |
| codecInst_B->pltype, CodecInstToSdp(*codecInst_B))); |
| } |
| } |
| } |
| |
| } // namespace webrtc |