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Adding Opus frame length test

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BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3672 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2013-03-15 13:29:17 +00:00
parent d613c207cc
commit 73222cff1a
11 changed files with 396 additions and 68 deletions
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1
webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
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@ -134,6 +134,7 @@
'../test/dual_stream_unittest.cc', '../test/dual_stream_unittest.cc',
'../test/EncodeDecodeTest.cc', '../test/EncodeDecodeTest.cc',
'../test/iSACTest.cc', '../test/iSACTest.cc',
'../test/opus_test.cc',
'../test/PCMFile.cc', '../test/PCMFile.cc',
'../test/RTPFile.cc', '../test/RTPFile.cc',
'../test/SpatialAudio.cc', '../test/SpatialAudio.cc',

24
webrtc/modules/audio_coding/main/test/APITest.cc
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@ -8,25 +8,27 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/modules/audio_coding/main/test/APITest.h"
#include <stdio.h> #include <stdio.h>
#include <stdlib.h> #include <stdlib.h>
#include <string.h> #include <string.h>
#include <cctype> #include <cctype>
#include <iostream> #include <iostream>
#include <ostream> #include <ostream>
#include <string> #include <string>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "APITest.h" #include "webrtc/engine_configurations.h"
#include "common_types.h" #include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
#include "engine_configurations.h" #include "webrtc/modules/audio_coding/main/test/utility.h"
#include "event_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "thread_wrapper.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "testsupport/fileutils.h" #include "webrtc/system_wrappers/interface/tick_util.h"
#include "tick_util.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "trace.h" #include "webrtc/test/testsupport/fileutils.h"
#include "utility.h"
namespace webrtc { namespace webrtc {

19
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
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@ -8,21 +8,22 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "EncodeDecodeTest.h" #include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
#include <sstream>
#include <stdio.h> #include <stdio.h>
#include <stdlib.h> #include <stdlib.h>
#include <string.h> #include <string.h>
#include <sstream>
#include <string> #include <string>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "audio_coding_module.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "common_types.h" #include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
#include "testsupport/fileutils.h" #include "webrtc/modules/audio_coding/main/test/utility.h"
#include "trace.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "utility.h" #include "webrtc/test/testsupport/fileutils.h"
namespace webrtc { namespace webrtc {

15
webrtc/modules/audio_coding/main/test/TestVADDTX.cc
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@ -8,16 +8,17 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "TestVADDTX.h" #include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
#include <iostream> #include <iostream>
#include "audio_coding_module_typedefs.h" #include "webrtc/common_types.h"
#include "common_types.h" #include "webrtc/engine_configurations.h"
#include "engine_configurations.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "testsupport/fileutils.h" #include "webrtc/modules/audio_coding/main/test/utility.h"
#include "trace.h" #include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
#include "utility.h" #include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc { namespace webrtc {

34
webrtc/modules/audio_coding/main/test/Tester.cc
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@ -12,19 +12,19 @@
#include <string> #include <string>
#include <vector> #include <vector>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "APITest.h" #include "webrtc/modules/audio_coding/main/test/APITest.h"
#include "audio_coding_module.h" #include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
#include "EncodeDecodeTest.h" #include "webrtc/modules/audio_coding/main/test/iSACTest.h"
#include "iSACTest.h" #include "webrtc/modules/audio_coding/main/test/opus_test.h"
#include "TestAllCodecs.h" #include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
#include "TestFEC.h" #include "webrtc/modules/audio_coding/main/test/TestFEC.h"
#include "TestStereo.h" #include "webrtc/modules/audio_coding/main/test/TestStereo.h"
#include "testsupport/fileutils.h" #include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
#include "TestVADDTX.h" #include "webrtc/modules/audio_coding/main/test/TwoWayCommunication.h"
#include "trace.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "TwoWayCommunication.h" #include "webrtc/test/testsupport/fileutils.h"
using webrtc::AudioCodingModule; using webrtc::AudioCodingModule;
using webrtc::Trace; using webrtc::Trace;
@ -128,6 +128,14 @@ TEST(AudioCodingModuleTest, TestAllCodecs) {
} }
#endif #endif
TEST(AudioCodingModuleTest, TestOpus) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_opus_trace.txt").c_str());
webrtc::OpusTest().Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, RunAllTests) { TEST(AudioCodingModuleTest, RunAllTests) {
std::vector<ACMTest*> tests; std::vector<ACMTest*> tests;
PopulateTests(&tests); PopulateTests(&tests);

5
webrtc/modules/audio_coding/main/test/delay_test.cc
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@ -16,16 +16,17 @@
#include <iostream> #include <iostream>
#include "gflags/gflags.h" #include "gflags/gflags.h"
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "testsupport/fileutils.h"
#include "webrtc/common_types.h" #include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h" #include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h" #include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h" #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h" #include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
DEFINE_string(codec, "isac", "Codec Name"); DEFINE_string(codec, "isac", "Codec Name");
DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz."); DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");

14
webrtc/modules/audio_coding/main/test/iSACTest.cc
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@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
#include <cctype> #include <cctype>
#include <stdio.h> #include <stdio.h>
#include <string.h> #include <string.h>
@ -21,12 +23,12 @@
#include <time.h> #include <time.h>
#endif #endif
#include "event_wrapper.h" #include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
#include "iSACTest.h" #include "webrtc/modules/audio_coding/main/test/utility.h"
#include "utility.h" #include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "trace.h" #include "webrtc/system_wrappers/interface/tick_util.h"
#include "testsupport/fileutils.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "tick_util.h" #include "webrtc/test/testsupport/fileutils.h"
namespace webrtc { namespace webrtc {

270
webrtc/modules/audio_coding/main/test/opus_test.cc Normal file
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@ -0,0 +1,270 @@
/*
* Copyright (c) 2013 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/opus_test.h"
#include <cassert>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
OpusTest::OpusTest()
: acm_receiver_(NULL),
channel_a2b_(NULL),
counter_(0),
payload_type_(255),
rtp_timestamp_(0) {
}
OpusTest::~OpusTest() {
if (acm_receiver_ != NULL) {
AudioCodingModule::Destroy(acm_receiver_);
acm_receiver_ = NULL;
}
if (channel_a2b_ != NULL) {
delete channel_a2b_;
channel_a2b_ = NULL;
}
if (opus_mono_encoder_ != NULL) {
WebRtcOpus_EncoderFree(opus_mono_encoder_);
opus_mono_encoder_ = NULL;
}
if (opus_stereo_encoder_ != NULL) {
WebRtcOpus_EncoderFree(opus_stereo_encoder_);
opus_stereo_encoder_ = NULL;
}
}
void OpusTest::Perform() {
#ifndef WEBRTC_CODEC_OPUS
// Opus isn't defined, exit.
return;
#else
uint16_t frequency_hz;
int audio_channels;
int16_t test_cntr = 0;
// Open both mono and stereo test files in 32 kHz.
const std::string file_name_stereo =
webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
const std::string file_name_mono =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
frequency_hz = 32000;
in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
in_file_stereo_.ReadStereo(true);
in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
in_file_mono_.ReadStereo(false);
// Create Opus encoders for mono and stereo.
ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1), -1);
ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2), -1);
// Create and initialize one ACM, to be used as receiver.
acm_receiver_ = AudioCodingModule::Create(0);
ASSERT_TRUE(acm_receiver_ != NULL);
EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
// Register Opus stereo as receiving codec.
CodecInst opus_codec_param;
int codec_id = acm_receiver_->Codec("opus", 48000, 2);
EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
payload_type_ = opus_codec_param.pltype;
EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
channel_a2b_->RegisterReceiverACM(acm_receiver_);
//
// Test Stereo.
//
channel_a2b_->set_codec_mode(kStereo);
audio_channels = 2;
test_cntr++;
OpenOutFile(test_cntr);
// Run Opus with 2.5 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 120);
// Run Opus with 5 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 240);
// Run Opus with 10 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 480);
// Run Opus with 20 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 960);
// Run Opus with 40 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 1920);
// Run Opus with 60 ms frame size.
Run(channel_a2b_, audio_channels, 64000, 2880);
out_file_.Close();
//
// Test Mono.
//
channel_a2b_->set_codec_mode(kMono);
audio_channels = 1;
test_cntr++;
OpenOutFile(test_cntr);
// Register Opus mono as receiving codec.
opus_codec_param.channels = 1;
EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
// Run Opus with 2.5 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 120);
// Run Opus with 5 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 240);
// Run Opus with 10 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 480);
// Run Opus with 20 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 960);
// Run Opus with 40 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 1920);
// Run Opus with 60 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 2880);
// Close the files.
in_file_stereo_.Close();
in_file_mono_.Close();
out_file_.Close();
#endif
}
void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
int frame_length, int percent_loss) {
AudioFrame audio_frame;
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
int16_t audio[480 * 12 * 2]; // Can hold 120 ms stereo audio.
int written_samples = 0;
int read_samples = 0;
channel->reset_payload_size();
// Set encoder rate.
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
while (1) {
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
if (percent_loss > 0) {
if (counter_ == floor((100 / percent_loss) + 0.5)) {
counter_ = 0;
channel->set_lost_packet(true);
} else {
channel->set_lost_packet(false);
}
counter_++;
}
// Get 10 msec of audio.
if (channels == 1) {
if (in_file_mono_.EndOfFile()) {
break;
}
in_file_mono_.Read10MsData(audio_frame);
} else {
if (in_file_stereo_.EndOfFile()) {
break;
}
in_file_stereo_.Read10MsData(audio_frame);
}
// Input audio is sampled at 32 kHz, but Opus operates at 48 kHz.
// Resampling is required.
EXPECT_EQ(480, resampler_.Resample10Msec(audio_frame.data_, 32000,
&audio[written_samples], 48000,
channels));
written_samples += 480 * channels;
// Sometimes we need to loop over the audio vector to produce the right
// number of packets.
int loop_encode = (written_samples - read_samples) /
(channels * frame_length);
if (loop_encode > 0) {
const int kMaxBytes = 1000; // Maximum number of bytes for one packet.
int16_t bitstream_len_byte;
uint8_t bitstream[kMaxBytes];
for (int i = 0; i < loop_encode; i++) {
if (channels == 1) {
bitstream_len_byte = WebRtcOpus_Encode(
opus_mono_encoder_, &audio[read_samples],
frame_length, kMaxBytes, bitstream);
ASSERT_GT(bitstream_len_byte, -1);
} else {
bitstream_len_byte = WebRtcOpus_Encode(
opus_stereo_encoder_, &audio[read_samples],
frame_length, kMaxBytes, bitstream);
ASSERT_GT(bitstream_len_byte, -1);
}
channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
bitstream, bitstream_len_byte, NULL);
rtp_timestamp_ += frame_length;
read_samples += frame_length * channels;
}
if (read_samples == written_samples) {
read_samples = 0;
written_samples = 0;
}
}
// Run received side of ACM.
CHECK_ERROR(acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
// Write output speech to file.
out_file_.Write10MsData(
audio_frame.data_,
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
}
if (in_file_mono_.EndOfFile()) {
in_file_mono_.Rewind();
}
if (in_file_stereo_.EndOfFile()) {
in_file_stereo_.Rewind();
}
// Reset in case we ended with a lost packet.
channel->set_lost_packet(false);
}
void OpusTest::OpenOutFile(int test_number) {
std::string file_name;
std::stringstream file_stream;
file_stream << webrtc::test::OutputPath() << "opustest_out_"
<< test_number << ".pcm";
file_name = file_stream.str();
out_file_.Open(file_name, 32000, "wb");
}
} // namespace webrtc

52
webrtc/modules/audio_coding/main/test/opus_test.h Normal file
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@ -0,0 +1,52 @@
/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
#include <math.h>
#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
namespace webrtc {
class OpusTest : public ACMTest {
public:
OpusTest();
~OpusTest();
void Perform();
private:
void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
int percent_loss = 0);
void OpenOutFile(int test_number);
AudioCodingModule* acm_receiver_;
TestPackStereo* channel_a2b_;
PCMFile in_file_stereo_;
PCMFile in_file_mono_;
PCMFile out_file_;
int counter_;
uint8_t payload_type_;
int rtp_timestamp_;
ACMResampler resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_

7
webrtc/modules/audio_coding/main/test/utility.cc
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@ -14,9 +14,10 @@
#include <stdio.h> #include <stdio.h>
#include <stdlib.h> #include <stdlib.h>
#include "audio_coding_module.h" #include "testing/gtest/include/gtest/gtest.h"
#include "common_types.h" #include "webrtc/common_types.h"
#include "gtest/gtest.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13 #define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13

21
webrtc/modules/audio_coding/main/test/utility.h
View File

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef ACM_TEST_UTILITY_H #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
#define ACM_TEST_UTILITY_H #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
#include "audio_coding_module.h" #include "testing/gtest/include/gtest/gtest.h"
#include "gtest/gtest.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
namespace webrtc { namespace webrtc {
@ -55,17 +55,6 @@ namespace webrtc {
}while(0) }while(0)
#ifdef WIN32
/* Exclude rarely-used stuff from Windows headers */
//#define WIN32_LEAN_AND_MEAN
/* OS-dependent case-insensitive string comparison */
#define STR_CASE_CMP(x,y) ::_stricmp(x,y)
#else
/* OS-dependent case-insensitive string comparison */
#define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
#endif
#define DESTROY_ACM(acm) \ #define DESTROY_ACM(acm) \
do { \ do { \
if(acm != NULL) { \ if(acm != NULL) { \
@ -192,4 +181,4 @@ private:
} // namespace webrtc } // namespace webrtc
#endif // ACM_TEST_UTILITY_H #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_