| /* |
| * 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 <cstdio> |
| #include <limits> |
| #include <string> |
| |
| #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 "test/gtest.h" |
| #include "test/testsupport/fileutils.h" |
| #include "typedefs.h" // NOLINT(build/include) |
| |
| // 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<size_t>::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 == kAudioFrameCN) { |
| rtp_info.type.Audio.isCNG = true; |
| } else { |
| rtp_info.type.Audio.isCNG = false; |
| } |
| if (frame_type == kEmptyFrame) { |
| // Skip this frame. |
| return 0; |
| } |
| |
| // Only run mono for all test cases. |
| rtp_info.type.Audio.channel = 1; |
| 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()), |
| acm_b_(AudioCodingModule::Create()), |
| 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<size_t>( |
| static_cast<float>(packet_size * rate) / |
| static_cast<float>(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; |
| CHECK_ERROR(my_acm->RegisterSendCodec(my_codec_param)); |
| } |
| |
| 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<int>(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(); |
| std::ostringstream 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 |