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webrtc/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
henrik.lundin@webrtc.org fa58745445 Delete all codec-specific subclasses of ACMGenericCodec 
They have all been replaced by AudioEncoder subclasses, accessed throgh
ACMGenericCodecWrapper objects. After this change, the only subclass of
ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated
in a future cl.)

This CL also deletes acm_opus_unittest.cc. This test file was already
replaced audio_encoder_opus_unittest.cc	in r8244.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40729004

Cr-Commit-Position: refs/heads/master@{#8457}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 09:26:51 +00:00

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/*
* 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/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/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 == kFrameEmpty) {
// 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
#ifdef WEBRTC_CODEC_PCM16
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();
#endif
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
#ifndef WEBRTC_CODEC_PCM16
printf(" PCM16\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));
// Run sender side of ACM.
CHECK_ERROR(acm_a_->Process());
// 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
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