diff options
Diffstat (limited to 'webrtc/modules/audio_coding/main/test/TestAllCodecs.cc')
-rw-r--r-- | webrtc/modules/audio_coding/main/test/TestAllCodecs.cc | 486 |
1 files changed, 0 insertions, 486 deletions
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc deleted file mode 100644 index 19189b6b8f..0000000000 --- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc +++ /dev/null @@ -1,486 +0,0 @@ -/* - * 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 "webrtc/modules/audio_coding/main/test/TestAllCodecs.h" - -#include <cstdio> -#include <limits> -#include <string> - -#include "testing/gtest/include/gtest/gtest.h" - -#include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" -#include "webrtc/system_wrappers/include/trace.h" -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/typedefs.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<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(0)), - acm_b_(AudioCodingModule::Create(1)), - 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) { - WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1, - "---------- 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); - } - - // 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. -#ifdef WEBRTC_CODEC_G722 - 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(); -#endif -#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_G722 - printf(" G.722\n"); -#endif -#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; - 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. - CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame)); - - // 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; - acm_a_->SendCodec(&my_codec_param); - printf("%s -> ", my_codec_param.plname); - acm_b_->ReceiveCodec(&my_codec_param); - printf("%s\n", my_codec_param.plname); -} - -} // namespace webrtc |