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Diffstat (limited to 'webrtc/modules/audio_coding/test/delay_test.cc')
| -rw-r--r-- | webrtc/modules/audio_coding/test/delay_test.cc | 265 |
1 files changed, 265 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc new file mode 100644 index 0000000000..a8c137f501 --- /dev/null +++ b/webrtc/modules/audio_coding/test/delay_test.cc @@ -0,0 +1,265 @@ +/* + * 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 <iostream> + +#include "gflags/gflags.h" +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/common.h" +#include "webrtc/common_types.h" +#include "webrtc/engine_configurations.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" +#include "webrtc/system_wrappers/include/event_wrapper.h" +#include "webrtc/test/testsupport/fileutils.h" + +DEFINE_string(codec, "isac", "Codec Name"); +DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz."); +DEFINE_int32(num_channels, 1, "Number of Channels."); +DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional."); +DEFINE_int32(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)."); + +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(0)), + acm_b_(AudioCodingModule::Create(1)), + 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 (FLAGS_input_file.size() > 0) + file_name = FLAGS_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 (FLAGS_delay > 0) { + ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_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(0, acm_b_->RegisterReceiveCodec(my_codec_param)) << + "Couldn't register receive codec.\n"; + } + + // 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_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz + << "Hz" << "_" << FLAGS_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 playout_ts; + 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); + ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame)); + out_file_b_.Write10MsData( + audio_frame.data_, + audio_frame.samples_per_channel_ * audio_frame.num_channels_); + acm_b_->PlayoutTimestamp(&playout_ts); + received_ts = channel_a2b_->LastInTimestamp(); + inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts) + / 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(); + } + + rtc::scoped_ptr<AudioCodingModule> acm_a_; + rtc::scoped_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[]) { + google::ParseCommandLineFlags(&argc, &argv, true); + webrtc::TestSettings test_setting; + strcpy(test_setting.codec.name, FLAGS_codec.c_str()); + + if (FLAGS_sample_rate_hz != 8000 && + FLAGS_sample_rate_hz != 16000 && + FLAGS_sample_rate_hz != 32000 && + FLAGS_sample_rate_hz != 48000) { + std::cout << "Invalid sampling rate.\n"; + return 1; + } + test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz; + if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) { + std::cout << "Only mono and stereo are supported.\n"; + return 1; + } + test_setting.codec.num_channels = FLAGS_num_channels; + test_setting.acm.dtx = FLAGS_dtx; + test_setting.acm.fec = FLAGS_fec; + test_setting.packet_loss = FLAGS_packet_loss; + + webrtc::DelayTest delay_test; + delay_test.Initialize(); + delay_test.Perform(&test_setting, 1, 240, "delay_test"); + return 0; +} |