/* * 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/TestAllCodecs.h" #include #include #include #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_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/utility.h" #include "rtc_base/logging.h" #include "rtc_base/stringencode.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" #include "test/testsupport/fileutils.h" // Description of the test: // In this test we set up a one-way communication channel from a participant // called "a" to a participant called "b". // a -> channel_a_to_b -> b // // The test loops through all available mono codecs, encode at "a" sends over // the channel, and decodes at "b". namespace { const size_t kVariableSize = std::numeric_limits::max(); } namespace webrtc { // Class for simulating packet handling. TestPack::TestPack() : receiver_acm_(NULL), sequence_number_(0), timestamp_diff_(0), last_in_timestamp_(0), total_bytes_(0), payload_size_(0) {} TestPack::~TestPack() {} void TestPack::RegisterReceiverACM(AudioCodingModule* acm) { receiver_acm_ = acm; return; } int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation) { WebRtcRTPHeader rtp_info; int32_t status; rtp_info.header.markerBit = false; rtp_info.header.ssrc = 0; rtp_info.header.sequenceNumber = sequence_number_++; rtp_info.header.payloadType = payload_type; rtp_info.header.timestamp = timestamp; if (frame_type == kEmptyFrame) { // Skip this frame. return 0; } // Only run mono for all test cases. memcpy(payload_data_, payload_data, payload_size); status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info); payload_size_ = payload_size; timestamp_diff_ = timestamp - last_in_timestamp_; last_in_timestamp_ = timestamp; total_bytes_ += payload_size; return status; } size_t TestPack::payload_size() { return payload_size_; } uint32_t TestPack::timestamp_diff() { return timestamp_diff_; } void TestPack::reset_payload_size() { payload_size_ = 0; } TestAllCodecs::TestAllCodecs(int test_mode) : acm_a_(AudioCodingModule::Create( AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))), acm_b_(AudioCodingModule::Create( AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))), channel_a_to_b_(NULL), test_count_(0), packet_size_samples_(0), packet_size_bytes_(0) { // test_mode = 0 for silent test (auto test) test_mode_ = test_mode; } TestAllCodecs::~TestAllCodecs() { if (channel_a_to_b_ != NULL) { delete channel_a_to_b_; channel_a_to_b_ = NULL; } } void TestAllCodecs::Perform() { const std::string file_name = webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); infile_a_.Open(file_name, 32000, "rb"); if (test_mode_ == 0) { RTC_LOG(LS_INFO) << "---------- TestAllCodecs ----------"; } acm_a_->InitializeReceiver(); acm_b_->InitializeReceiver(); uint8_t num_encoders = acm_a_->NumberOfCodecs(); CodecInst my_codec_param; for (uint8_t n = 0; n < num_encoders; n++) { acm_b_->Codec(n, &my_codec_param); if (!strcmp(my_codec_param.plname, "opus")) { my_codec_param.channels = 1; } acm_b_->RegisterReceiveCodec(my_codec_param.pltype, CodecInstToSdp(my_codec_param)); } // Create and connect the channel channel_a_to_b_ = new TestPack; acm_a_->RegisterTransportCallback(channel_a_to_b_); channel_a_to_b_->RegisterReceiverACM(acm_b_.get()); // All codecs are tested for all allowed sampling frequencies, rates and // packet sizes. if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); char codec_g722[] = "G722"; RegisterSendCodec('A', codec_g722, 16000, 64000, 160, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_g722, 16000, 64000, 320, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_g722, 16000, 64000, 480, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_g722, 16000, 64000, 640, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_g722, 16000, 64000, 800, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0); Run(channel_a_to_b_); outfile_b_.Close(); #ifdef WEBRTC_CODEC_ILBC if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); char codec_ilbc[] = "ILBC"; RegisterSendCodec('A', codec_ilbc, 8000, 13300, 240, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_ilbc, 8000, 13300, 480, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_ilbc, 8000, 15200, 160, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0); Run(channel_a_to_b_); outfile_b_.Close(); #endif #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); char codec_isac[] = "ISAC"; RegisterSendCodec('A', codec_isac, 16000, -1, 480, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_isac, 16000, -1, 960, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_isac, 16000, 15000, 480, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_isac, 16000, 32000, 960, kVariableSize); Run(channel_a_to_b_); outfile_b_.Close(); #endif #ifdef WEBRTC_CODEC_ISAC if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_isac, 32000, 56000, 960, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_isac, 32000, 37000, 960, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_isac, 32000, 32000, 960, kVariableSize); Run(channel_a_to_b_); outfile_b_.Close(); #endif if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); char codec_l16[] = "L16"; RegisterSendCodec('A', codec_l16, 8000, 128000, 80, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_l16, 8000, 128000, 160, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_l16, 8000, 128000, 240, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0); Run(channel_a_to_b_); outfile_b_.Close(); if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_l16, 16000, 256000, 320, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_l16, 16000, 256000, 480, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0); Run(channel_a_to_b_); outfile_b_.Close(); if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0); Run(channel_a_to_b_); outfile_b_.Close(); if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); char codec_pcma[] = "PCMA"; RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0); Run(channel_a_to_b_); if (test_mode_ != 0) { printf("===============================================================\n"); } char codec_pcmu[] = "PCMU"; RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, 0); Run(channel_a_to_b_); RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, 0); Run(channel_a_to_b_); outfile_b_.Close(); #ifdef WEBRTC_CODEC_OPUS if (test_mode_ != 0) { printf("===============================================================\n"); } test_count_++; OpenOutFile(test_count_); char codec_opus[] = "OPUS"; RegisterSendCodec('A', codec_opus, 48000, 6000, 480, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_opus, 48000, 20000, 480 * 2, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_opus, 48000, 32000, 480 * 4, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_opus, 48000, 96000, 480 * 6, kVariableSize); Run(channel_a_to_b_); RegisterSendCodec('A', codec_opus, 48000, 500000, 480 * 2, kVariableSize); Run(channel_a_to_b_); outfile_b_.Close(); #endif if (test_mode_ != 0) { printf("===============================================================\n"); /* Print out all codecs that were not tested in the run */ printf("The following codecs was not included in the test:\n"); #ifndef WEBRTC_CODEC_ILBC printf(" iLBC\n"); #endif #ifndef WEBRTC_CODEC_ISAC printf(" ISAC float\n"); #endif #ifndef WEBRTC_CODEC_ISACFX printf(" ISAC fix\n"); #endif printf("\nTo complete the test, listen to the %d number of output files.\n", test_count_); } } // Register Codec to use in the test // // Input: side - which ACM to use, 'A' or 'B' // codec_name - name to use when register the codec // sampling_freq_hz - sampling frequency in Herz // rate - bitrate in bytes // packet_size - packet size in samples // extra_byte - if extra bytes needed compared to the bitrate // used when registering, can be an internal header // set to kVariableSize if the codec is a variable // rate codec void TestAllCodecs::RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz, int rate, int packet_size, size_t extra_byte) { if (test_mode_ != 0) { // Print out codec and settings. printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name, sampling_freq_hz, rate, packet_size); } // Store packet-size in samples, used to validate the received packet. // If G.722, store half the size to compensate for the timestamp bug in the // RFC for G.722. // If iSAC runs in adaptive mode, packet size in samples can change on the // fly, so we exclude this test by setting |packet_size_samples_| to -1. if (!strcmp(codec_name, "G722")) { packet_size_samples_ = packet_size / 2; } else if (!strcmp(codec_name, "ISAC") && (rate == -1)) { packet_size_samples_ = -1; } else { packet_size_samples_ = packet_size; } // Store the expected packet size in bytes, used to validate the received // packet. If variable rate codec (extra_byte == -1), set to -1. if (extra_byte != kVariableSize) { // Add 0.875 to always round up to a whole byte packet_size_bytes_ = static_cast(static_cast(packet_size * rate) / static_cast(sampling_freq_hz * 8) + 0.875) + extra_byte; } else { // Packets will have a variable size. packet_size_bytes_ = kVariableSize; } // Set pointer to the ACM where to register the codec. AudioCodingModule* my_acm = NULL; switch (side) { case 'A': { my_acm = acm_a_.get(); break; } case 'B': { my_acm = acm_b_.get(); break; } default: { break; } } ASSERT_TRUE(my_acm != NULL); // Get all codec parameters before registering CodecInst my_codec_param; CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param, sampling_freq_hz, 1)); my_codec_param.rate = rate; my_codec_param.pacsize = packet_size; auto factory = CreateBuiltinAudioEncoderFactory(); constexpr int payload_type = 17; SdpAudioFormat format = CodecInstToSdp(my_codec_param); format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact( packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000))); my_acm->SetEncoder( factory->MakeAudioEncoder(payload_type, format, absl::nullopt)); } void TestAllCodecs::Run(TestPack* channel) { AudioFrame audio_frame; int32_t out_freq_hz = outfile_b_.SamplingFrequency(); size_t receive_size; uint32_t timestamp_diff; channel->reset_payload_size(); int error_count = 0; int counter = 0; // Set test length to 500 ms (50 blocks of 10 ms each). infile_a_.SetNum10MsBlocksToRead(50); // Fast-forward 1 second (100 blocks) since the file starts with silence. infile_a_.FastForward(100); while (!infile_a_.EndOfFile()) { // Add 10 msec to ACM. infile_a_.Read10MsData(audio_frame); CHECK_ERROR(acm_a_->Add10MsData(audio_frame)); // Verify that the received packet size matches the settings. receive_size = channel->payload_size(); if (receive_size) { if ((receive_size != packet_size_bytes_) && (packet_size_bytes_ != kVariableSize)) { error_count++; } // Verify that the timestamp is updated with expected length. The counter // is used to avoid problems when switching codec or frame size in the // test. timestamp_diff = channel->timestamp_diff(); if ((counter > 10) && (static_cast(timestamp_diff) != packet_size_samples_) && (packet_size_samples_ > -1)) error_count++; } // Run received side of ACM. bool muted; CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame, &muted)); ASSERT_FALSE(muted); // Write output speech to file. outfile_b_.Write10MsData(audio_frame.data(), audio_frame.samples_per_channel_); // Update loop counter counter++; } EXPECT_EQ(0, error_count); if (infile_a_.EndOfFile()) { infile_a_.Rewind(); } } void TestAllCodecs::OpenOutFile(int test_number) { std::string filename = webrtc::test::OutputPath(); rtc::StringBuilder test_number_str; test_number_str << test_number; filename += "testallcodecs_out_"; filename += test_number_str.str(); filename += ".pcm"; outfile_b_.Open(filename, 32000, "wb"); } void TestAllCodecs::DisplaySendReceiveCodec() { CodecInst my_codec_param; printf("%s -> ", acm_a_->SendCodec()->plname); acm_b_->ReceiveCodec(&my_codec_param); printf("%s\n", my_codec_param.plname); } } // namespace webrtc