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Unified Diff: webrtc/modules/audio_coding/main/test/TestAllCodecs.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
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Index: webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
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 e9e4f2bceab7fb4436250c936025f5b1a5abae0c..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ /dev/null
@@ -1,485 +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;
- printf("%s -> ", acm_a_->SendCodec()->plname);
- acm_b_->ReceiveCodec(&my_codec_param);
- printf("%s\n", my_codec_param.plname);
-}
-
-} // namespace webrtc
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