diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index a3ff4d40bf..fe9ca70033 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -1263,7 +1263,6 @@ if (rtc_include_tests) { ":audio_codec_speed_tests", ":audio_decoder_unittests", ":audio_decoder_unittests", - ":delay_test", ":g711_test", ":g722_test", ":ilbc_test", @@ -1422,38 +1421,6 @@ if (rtc_include_tests) { ] } - rtc_executable("delay_test") { - testonly = true - sources = [ - "test/Channel.cc", - "test/Channel.h", - "test/PCMFile.cc", - "test/PCMFile.h", - "test/delay_test.cc", - "test/utility.cc", - "test/utility.h", - ] - - deps = [ - ":audio_coding", - ":audio_coding_module_typedefs", - ":audio_format_conversion", - "..:module_api", - "../../:webrtc_common", - "../../api/audio:audio_frame_api", - "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../rtc_base:checks", - "../../rtc_base:rtc_base_approved", - "../../system_wrappers", - "../../system_wrappers:system_wrappers_default", - "../../test:fileutils", - "../../test:test_support", - "../rtp_rtcp", - "//testing/gtest", - "//third_party/abseil-cpp/absl/types:optional", - ] - } # delay_test - audio_decoder_unittests_resources = [ "../../resources/audio_coding/testfile32kHz.pcm" ] diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc deleted file mode 100644 index 6402e05780..0000000000 --- a/modules/audio_coding/test/delay_test.cc +++ /dev/null @@ -1,282 +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 -#include -#include - -#include -#include - -#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/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 "rtc_base/strings/string_builder.h" -#include "system_wrappers/include/event_wrapper.h" -#include "test/gtest.h" -#include "test/testsupport/fileutils.h" - -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( - AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))), - acm_b_(AudioCodingModule::Create( - AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))), - 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) { - rtc::StringBuilder 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(); - absl::optional playout_timestamp = acm_b_->PlayoutTimestamp(); - ASSERT_TRUE(playout_timestamp); - inst_delay_sec = static_cast(received_ts - *playout_timestamp) / - static_cast(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 acm_a_; - std::unique_ptr 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; -}