| /* |
| * 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 <assert.h> |
| #include <math.h> |
| #include <string.h> |
| |
| #include <iostream> |
| #include <memory> |
| |
| #include "common_types.h" // NOLINT(build/include) |
| #include "modules/audio_coding/codecs/audio_format_conversion.h" |
| #include "modules/audio_coding/include/audio_coding_module.h" |
| #include "modules/audio_coding/include/audio_coding_module_typedefs.h" |
| #include "modules/audio_coding/test/Channel.h" |
| #include "modules/audio_coding/test/PCMFile.h" |
| #include "modules/audio_coding/test/utility.h" |
| #include "rtc_base/flags.h" |
| #include "system_wrappers/include/event_wrapper.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/fileutils.h" |
| #include "typedefs.h" // NOLINT(build/include) |
| |
| DEFINE_string(codec, "isac", "Codec Name"); |
| DEFINE_int(sample_rate_hz, 16000, "Sampling rate in Hertz."); |
| DEFINE_int(num_channels, 1, "Number of Channels."); |
| DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional."); |
| DEFINE_int(delay, 0, "Delay in millisecond."); |
| DEFINE_bool(dtx, false, "Enable DTX at the sender side."); |
| DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}."); |
| DEFINE_bool(fec, false, "Use Forward Error Correction (FEC)."); |
| DEFINE_bool(help, false, "Print this message."); |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| struct CodecSettings { |
| char name[50]; |
| int sample_rate_hz; |
| int num_channels; |
| }; |
| |
| struct AcmSettings { |
| bool dtx; |
| bool fec; |
| }; |
| |
| struct TestSettings { |
| CodecSettings codec; |
| AcmSettings acm; |
| bool packet_loss; |
| }; |
| |
| } // namespace |
| |
| class DelayTest { |
| public: |
| DelayTest() |
| : acm_a_(AudioCodingModule::Create()), |
| acm_b_(AudioCodingModule::Create()), |
| channel_a2b_(new Channel), |
| test_cntr_(0), |
| encoding_sample_rate_hz_(8000) {} |
| |
| ~DelayTest() { |
| if (channel_a2b_ != NULL) { |
| delete channel_a2b_; |
| channel_a2b_ = NULL; |
| } |
| in_file_a_.Close(); |
| } |
| |
| void Initialize() { |
| test_cntr_ = 0; |
| std::string file_name = webrtc::test::ResourcePath( |
| "audio_coding/testfile32kHz", "pcm"); |
| if (strlen(FLAG_input_file) > 0) |
| file_name = FLAG_input_file; |
| in_file_a_.Open(file_name, 32000, "rb"); |
| ASSERT_EQ(0, acm_a_->InitializeReceiver()) << |
| "Couldn't initialize receiver.\n"; |
| ASSERT_EQ(0, acm_b_->InitializeReceiver()) << |
| "Couldn't initialize receiver.\n"; |
| |
| if (FLAG_delay > 0) { |
| ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAG_delay)) << |
| "Failed to set minimum delay.\n"; |
| } |
| |
| int num_encoders = acm_a_->NumberOfCodecs(); |
| CodecInst my_codec_param; |
| for (int n = 0; n < num_encoders; n++) { |
| EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) << |
| "Failed to get codec."; |
| if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0) |
| my_codec_param.channels = 1; |
| else if (my_codec_param.channels > 1) |
| continue; |
| if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 && |
| my_codec_param.plfreq == 48000) |
| continue; |
| if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0) |
| continue; |
| ASSERT_EQ(true, |
| acm_b_->RegisterReceiveCodec(my_codec_param.pltype, |
| CodecInstToSdp(my_codec_param))); |
| } |
| |
| // Create and connect the channel |
| ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) << |
| "Couldn't register Transport callback.\n"; |
| channel_a2b_->RegisterReceiverACM(acm_b_.get()); |
| } |
| |
| void Perform(const TestSettings* config, size_t num_tests, int duration_sec, |
| const char* output_prefix) { |
| for (size_t n = 0; n < num_tests; ++n) { |
| ApplyConfig(config[n]); |
| Run(duration_sec, output_prefix); |
| } |
| } |
| |
| private: |
| void ApplyConfig(const TestSettings& config) { |
| printf("====================================\n"); |
| printf("Test %d \n" |
| "Codec: %s, %d kHz, %d channel(s)\n" |
| "ACM: DTX %s, FEC %s\n" |
| "Channel: %s\n", |
| ++test_cntr_, config.codec.name, config.codec.sample_rate_hz, |
| config.codec.num_channels, config.acm.dtx ? "on" : "off", |
| config.acm.fec ? "on" : "off", |
| config.packet_loss ? "with packet-loss" : "no packet-loss"); |
| SendCodec(config.codec); |
| ConfigAcm(config.acm); |
| ConfigChannel(config.packet_loss); |
| } |
| |
| void SendCodec(const CodecSettings& config) { |
| CodecInst my_codec_param; |
| ASSERT_EQ(0, AudioCodingModule::Codec( |
| config.name, &my_codec_param, config.sample_rate_hz, |
| config.num_channels)) << "Specified codec is not supported.\n"; |
| |
| encoding_sample_rate_hz_ = my_codec_param.plfreq; |
| ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) << |
| "Failed to register send-codec.\n"; |
| } |
| |
| void ConfigAcm(const AcmSettings& config) { |
| ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) << |
| "Failed to set VAD.\n"; |
| ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) << |
| "Failed to set RED.\n"; |
| } |
| |
| void ConfigChannel(bool packet_loss) { |
| channel_a2b_->SetFECTestWithPacketLoss(packet_loss); |
| } |
| |
| void OpenOutFile(const char* output_id) { |
| std::stringstream file_stream; |
| file_stream << "delay_test_" << FLAG_codec << "_" << FLAG_sample_rate_hz |
| << "Hz" << "_" << FLAG_delay << "ms.pcm"; |
| std::cout << "Output file: " << file_stream.str() << std::endl << std::endl; |
| std::string file_name = webrtc::test::OutputPath() + file_stream.str(); |
| out_file_b_.Open(file_name.c_str(), 32000, "wb"); |
| } |
| |
| void Run(int duration_sec, const char* output_prefix) { |
| OpenOutFile(output_prefix); |
| AudioFrame audio_frame; |
| uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency(); |
| |
| int num_frames = 0; |
| int in_file_frames = 0; |
| uint32_t received_ts; |
| double average_delay = 0; |
| double inst_delay_sec = 0; |
| while (num_frames < (duration_sec * 100)) { |
| if (in_file_a_.EndOfFile()) { |
| in_file_a_.Rewind(); |
| } |
| |
| // Print delay information every 16 frame |
| if ((num_frames & 0x3F) == 0x3F) { |
| NetworkStatistics statistics; |
| acm_b_->GetNetworkStatistics(&statistics); |
| fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d" |
| " ts-based average = %6.3f, " |
| "curr buff-lev = %4u opt buff-lev = %4u \n", |
| statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs, |
| statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs, |
| average_delay, statistics.currentBufferSize, |
| statistics.preferredBufferSize); |
| fflush (stdout); |
| } |
| |
| in_file_a_.Read10MsData(audio_frame); |
| ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0); |
| bool muted; |
| ASSERT_EQ(0, |
| acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted)); |
| RTC_DCHECK(!muted); |
| out_file_b_.Write10MsData( |
| audio_frame.data(), |
| audio_frame.samples_per_channel_ * audio_frame.num_channels_); |
| received_ts = channel_a2b_->LastInTimestamp(); |
| rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp(); |
| ASSERT_TRUE(playout_timestamp); |
| inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) / |
| static_cast<double>(encoding_sample_rate_hz_); |
| |
| if (num_frames > 10) |
| average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec; |
| |
| ++num_frames; |
| ++in_file_frames; |
| } |
| out_file_b_.Close(); |
| } |
| |
| std::unique_ptr<AudioCodingModule> acm_a_; |
| std::unique_ptr<AudioCodingModule> acm_b_; |
| |
| Channel* channel_a2b_; |
| |
| PCMFile in_file_a_; |
| PCMFile out_file_b_; |
| int test_cntr_; |
| int encoding_sample_rate_hz_; |
| }; |
| |
| } // namespace webrtc |
| |
| int main(int argc, char* argv[]) { |
| if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) { |
| return 1; |
| } |
| if (FLAG_help) { |
| rtc::FlagList::Print(nullptr, false); |
| return 0; |
| } |
| |
| webrtc::TestSettings test_setting; |
| strcpy(test_setting.codec.name, FLAG_codec); |
| |
| if (FLAG_sample_rate_hz != 8000 && |
| FLAG_sample_rate_hz != 16000 && |
| FLAG_sample_rate_hz != 32000 && |
| FLAG_sample_rate_hz != 48000) { |
| std::cout << "Invalid sampling rate.\n"; |
| return 1; |
| } |
| test_setting.codec.sample_rate_hz = FLAG_sample_rate_hz; |
| if (FLAG_num_channels < 1 || FLAG_num_channels > 2) { |
| std::cout << "Only mono and stereo are supported.\n"; |
| return 1; |
| } |
| test_setting.codec.num_channels = FLAG_num_channels; |
| test_setting.acm.dtx = FLAG_dtx; |
| test_setting.acm.fec = FLAG_fec; |
| test_setting.packet_loss = FLAG_packet_loss; |
| |
| webrtc::DelayTest delay_test; |
| delay_test.Initialize(); |
| delay_test.Perform(&test_setting, 1, 240, "delay_test"); |
| return 0; |
| } |
