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Reland "Converting five tests to use new AudioCoding interface" (r7258)

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This CL reverts r7264. The problem was that iSAC-SWB and iSAC-FB are
not supported on android. These are now disabled.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7273 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-09-23 12:05:34 +00:00
parent 4f6f22f0c6
commit 0e6e4d2ff2
12 changed files with 1575 additions and 207 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

160
webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
View File

@ -22,124 +22,67 @@
namespace webrtc {
namespace test {
namespace {
// Returns true if the codec should be registered, otherwise false. Changes
// the number of channels for the Opus codec to always be 1.
bool ModifyAndUseThisCodec(CodecInst* codec_param) {
if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
codec_param->plfreq == 48000)
return false; // Skip 48 kHz comfort noise.
if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
return false; // Skip DTFM.
return true;
}
// Remaps payload types from ACM's default to those used in the resource file
// neteq_universal_new.rtp. Returns true if the codec should be registered,
// otherwise false. The payload types are set as follows (all are mono codecs):
// PCMu = 0;
// PCMa = 8;
// Comfort noise 8 kHz = 13
// Comfort noise 16 kHz = 98
// Comfort noise 32 kHz = 99
// iLBC = 102
// iSAC wideband = 103
// iSAC super-wideband = 104
// iSAC fullband = 124
// AVT/DTMF = 106
// RED = 117
// PCM16b 8 kHz = 93
// PCM16b 16 kHz = 94
// PCM16b 32 kHz = 95
// G.722 = 94
bool RemapPltypeAndUseThisCodec(const char* plname,
int plfreq,
int channels,
int* pltype) {
if (channels != 1)
return false; // Don't use non-mono codecs.
// Re-map pltypes to those used in the NetEq test files.
if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
*pltype = 0;
} else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
*pltype = 8;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
*pltype = 13;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
*pltype = 98;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
*pltype = 99;
} else if (STR_CASE_CMP(plname, "ILBC") == 0) {
*pltype = 102;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
*pltype = 103;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
*pltype = 104;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 48000) {
*pltype = 124;
} else if (STR_CASE_CMP(plname, "telephone-event") == 0) {
*pltype = 106;
} else if (STR_CASE_CMP(plname, "red") == 0) {
*pltype = 117;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
*pltype = 93;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
*pltype = 94;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
*pltype = 95;
} else if (STR_CASE_CMP(plname, "G722") == 0) {
*pltype = 9;
} else {
// Don't use any other codecs.
return false;
}
return true;
}
} // namespace
AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels)
: clock_(0),
acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
packet_source_(packet_source),
audio_sink_(audio_sink),
output_freq_hz_(output_freq_hz),
exptected_output_channels_(exptected_output_channels) {
webrtc::AudioCoding::Config config;
config.clock = &clock_;
config.playout_frequency_hz = output_freq_hz_;
acm_.reset(webrtc::AudioCoding::Create(config));
}
void AcmReceiveTest::RegisterDefaultCodecs() {
CodecInst my_codec_param;
for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
if (ModifyAndUseThisCodec(&my_codec_param)) {
ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
<< "Couldn't register receive codec.\n";
}
}
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
#ifndef WEBRTC_ANDROID
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 105));
#endif
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 107));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 108));
ASSERT_TRUE(
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 109));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B_2ch, 111));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb_2ch, 112));
ASSERT_TRUE(
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 113));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU_2ch, 110));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA_2ch, 118));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722_2ch, 119));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 127));
}
void AcmReceiveTest::RegisterNetEqTestCodecs() {
CodecInst my_codec_param;
for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
if (!ModifyAndUseThisCodec(&my_codec_param)) {
// Skip this codec.
continue;
}
if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
my_codec_param.plfreq,
my_codec_param.channels,
&my_codec_param.pltype)) {
ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
<< "Couldn't register receive codec.\n";
}
}
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
#ifndef WEBRTC_ANDROID
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 124));
#endif
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 93));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 94));
ASSERT_TRUE(
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 95));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 117));
}
void AcmReceiveTest::Run() {
@ -148,7 +91,7 @@ void AcmReceiveTest::Run() {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
EXPECT_TRUE(acm_->Get10MsAudio(&output_frame));
EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
const int samples_per_block = output_freq_hz_ * 10 / 1000;
EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
@ -170,11 +113,10 @@ void AcmReceiveTest::Run() {
header.header = packet->header();
header.frameType = kAudioFrameSpeech;
memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
EXPECT_EQ(0,
acm_->IncomingPacket(
packet->payload(),
static_cast<int32_t>(packet->payload_length_bytes()),
header))
EXPECT_TRUE(
acm_->InsertPacket(packet->payload(),
static_cast<int32_t>(packet->payload_length_bytes()),
header))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(header.header.payloadType) << std::endl
<< " TS = " << header.header.timestamp << std::endl

4
webrtc/modules/audio_coding/main/acm2/acm_receive_test.h
View File

@ -16,7 +16,7 @@
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioCodingModule;
class AudioCoding;
struct CodecInst;
namespace test {
@ -50,7 +50,7 @@ class AcmReceiveTest {
private:
SimulatedClock clock_;
scoped_ptr<AudioCodingModule> acm_;
scoped_ptr<AudioCoding> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
const int output_freq_hz_;

187
webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc Normal file
View File

@ -0,0 +1,187 @@
/*
* Copyright (c) 2014 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/acm2/acm_receive_test_oldapi.h"
#include <assert.h>
#include <stdio.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
namespace webrtc {
namespace test {
namespace {
// Returns true if the codec should be registered, otherwise false. Changes
// the number of channels for the Opus codec to always be 1.
bool ModifyAndUseThisCodec(CodecInst* codec_param) {
if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
codec_param->plfreq == 48000)
return false; // Skip 48 kHz comfort noise.
if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
return false; // Skip DTFM.
return true;
}
// Remaps payload types from ACM's default to those used in the resource file
// neteq_universal_new.rtp. Returns true if the codec should be registered,
// otherwise false. The payload types are set as follows (all are mono codecs):
// PCMu = 0;
// PCMa = 8;
// Comfort noise 8 kHz = 13
// Comfort noise 16 kHz = 98
// Comfort noise 32 kHz = 99
// iLBC = 102
// iSAC wideband = 103
// iSAC super-wideband = 104
// iSAC fullband = 124
// AVT/DTMF = 106
// RED = 117
// PCM16b 8 kHz = 93
// PCM16b 16 kHz = 94
// PCM16b 32 kHz = 95
// G.722 = 94
bool RemapPltypeAndUseThisCodec(const char* plname,
int plfreq,
int channels,
int* pltype) {
if (channels != 1)
return false; // Don't use non-mono codecs.
// Re-map pltypes to those used in the NetEq test files.
if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
*pltype = 0;
} else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
*pltype = 8;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
*pltype = 13;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
*pltype = 98;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
*pltype = 99;
} else if (STR_CASE_CMP(plname, "ILBC") == 0) {
*pltype = 102;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
*pltype = 103;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
*pltype = 104;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 48000) {
*pltype = 124;
} else if (STR_CASE_CMP(plname, "telephone-event") == 0) {
*pltype = 106;
} else if (STR_CASE_CMP(plname, "red") == 0) {
*pltype = 117;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
*pltype = 93;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
*pltype = 94;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
*pltype = 95;
} else if (STR_CASE_CMP(plname, "G722") == 0) {
*pltype = 9;
} else {
// Don't use any other codecs.
return false;
}
return true;
}
} // namespace
AcmReceiveTestOldApi::AcmReceiveTestOldApi(
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels)
: clock_(0),
acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
packet_source_(packet_source),
audio_sink_(audio_sink),
output_freq_hz_(output_freq_hz),
exptected_output_channels_(exptected_output_channels) {
}
void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
CodecInst my_codec_param;
for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
if (ModifyAndUseThisCodec(&my_codec_param)) {
ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
<< "Couldn't register receive codec.\n";
}
}
}
void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
CodecInst my_codec_param;
for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
if (!ModifyAndUseThisCodec(&my_codec_param)) {
// Skip this codec.
continue;
}
if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
my_codec_param.plfreq,
my_codec_param.channels,
&my_codec_param.pltype)) {
ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
<< "Couldn't register receive codec.\n";
}
}
}
void AcmReceiveTestOldApi::Run() {
for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
packet.reset(packet_source_->NextPacket())) {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
const int samples_per_block = output_freq_hz_ * 10 / 1000;
EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
if (exptected_output_channels_ != kArbitraryChannels) {
if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
// Don't check number of channels for PLC output, since each test run
// usually starts with a short period of mono PLC before decoding the
// first packet.
} else {
EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
}
}
ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
clock_.AdvanceTimeMilliseconds(10);
}
// Insert packet after converting from RTPHeader to WebRtcRTPHeader.
WebRtcRTPHeader header;
header.header = packet->header();
header.frameType = kAudioFrameSpeech;
memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
EXPECT_EQ(0,
acm_->IncomingPacket(
packet->payload(),
static_cast<int32_t>(packet->payload_length_bytes()),
header))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(header.header.payloadType) << std::endl
<< " TS = " << header.header.timestamp << std::endl
<< " SN = " << header.header.sequenceNumber;
}
}
} // namespace test
} // namespace webrtc

63
webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h Normal file
View File

@ -0,0 +1,63 @@
/*
* Copyright (c) 2014 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_ACM2_ACM_RECEIVE_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioCodingModule;
struct CodecInst;
namespace test {
class AudioSink;
class PacketSource;
class AcmReceiveTestOldApi {
public:
enum NumOutputChannels {
kArbitraryChannels = 0,
kMonoOutput = 1,
kStereoOutput = 2
};
AcmReceiveTestOldApi(PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels);
virtual ~AcmReceiveTestOldApi() {}
// Registers the codecs with default parameters from ACM.
void RegisterDefaultCodecs();
// Registers codecs with payload types matching the pre-encoded NetEq test
// files.
void RegisterNetEqTestCodecs();
// Runs the test and returns true if successful.
void Run();
private:
SimulatedClock clock_;
scoped_ptr<AudioCodingModule> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
const int output_freq_hz_;
NumOutputChannels exptected_output_channels_;
DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_

20
webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
View File

@ -27,7 +27,6 @@ AcmSendTest::AcmSendTest(InputAudioFile* audio_source,
int source_rate_hz,
int test_duration_ms)
: clock_(0),
acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
audio_source_(audio_source),
source_rate_hz_(source_rate_hz),
input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
@ -37,24 +36,23 @@ AcmSendTest::AcmSendTest(InputAudioFile* audio_source,
payload_type_(0),
timestamp_(0),
sequence_number_(0) {
webrtc::AudioCoding::Config config;
config.clock = &clock_;
config.transport = this;
acm_.reset(webrtc::AudioCoding::Create(config));
input_frame_.sample_rate_hz_ = source_rate_hz_;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = input_block_size_samples_;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
acm_->RegisterTransportCallback(this);
}
bool AcmSendTest::RegisterCodec(const char* payload_name,
int sampling_freq_hz,
bool AcmSendTest::RegisterCodec(int codec_type,
int channels,
int payload_type,
int frame_size_samples) {
CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec_, sampling_freq_hz,
channels));
codec_.pltype = payload_type;
codec_.pacsize = frame_size_samples;
codec_registered_ = (acm_->RegisterSendCodec(codec_) == 0);
codec_registered_ =
acm_->RegisterSendCodec(codec_type, payload_type, frame_size_samples);
input_frame_.num_channels_ = channels;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
@ -79,9 +77,9 @@ Packet* AcmSendTest::NextPacket() {
input_frame_.num_channels_,
input_frame_.data_);
}
CHECK_EQ(0, acm_->Add10MsData(input_frame_));
int32_t encoded_bytes = acm_->Add10MsAudio(input_frame_);
EXPECT_GE(encoded_bytes, 0);
input_frame_.timestamp_ += input_block_size_samples_;
int32_t encoded_bytes = acm_->Process();
if (encoded_bytes > 0) {
// Encoded packet received.
return CreatePacket();

6
webrtc/modules/audio_coding/main/acm2/acm_send_test.h
View File

@ -33,8 +33,7 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
virtual ~AcmSendTest() {}
// Registers the send codec. Returns true on success, false otherwise.
bool RegisterCodec(const char* payload_name,
int sampling_freq_hz,
bool RegisterCodec(int codec_type,
int channels,
int payload_type,
int frame_size_samples);
@ -62,12 +61,11 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
Packet* CreatePacket();
SimulatedClock clock_;
scoped_ptr<AudioCodingModule> acm_;
scoped_ptr<AudioCoding> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
const int input_block_size_samples_;
AudioFrame input_frame_;
CodecInst codec_;
bool codec_registered_;
int test_duration_ms_;
// The following member variables are set whenever SendData() is called.

145
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc Normal file
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@ -0,0 +1,145 @@
/*
* Copyright (c) 2014 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/acm2/acm_send_test_oldapi.h"
#include <assert.h>
#include <stdio.h>
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
namespace webrtc {
namespace test {
AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
int source_rate_hz,
int test_duration_ms)
: clock_(0),
acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
audio_source_(audio_source),
source_rate_hz_(source_rate_hz),
input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
codec_registered_(false),
test_duration_ms_(test_duration_ms),
frame_type_(kAudioFrameSpeech),
payload_type_(0),
timestamp_(0),
sequence_number_(0) {
input_frame_.sample_rate_hz_ = source_rate_hz_;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = input_block_size_samples_;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
acm_->RegisterTransportCallback(this);
}
bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples) {
CHECK_EQ(0,
AudioCodingModule::Codec(
payload_name, &codec_, sampling_freq_hz, channels));
codec_.pltype = payload_type;
codec_.pacsize = frame_size_samples;
codec_registered_ = (acm_->RegisterSendCodec(codec_) == 0);
input_frame_.num_channels_ = channels;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
return codec_registered_;
}
Packet* AcmSendTestOldApi::NextPacket() {
assert(codec_registered_);
if (filter_.test(payload_type_)) {
// This payload type should be filtered out. Since the payload type is the
// same throughout the whole test run, no packet at all will be delivered.
// We can just as well signal that the test is over by returning NULL.
return NULL;
}
// Insert audio and process until one packet is produced.
while (clock_.TimeInMilliseconds() < test_duration_ms_) {
clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_));
if (input_frame_.num_channels_ > 1) {
InputAudioFile::DuplicateInterleaved(input_frame_.data_,
input_block_size_samples_,
input_frame_.num_channels_,
input_frame_.data_);
}
CHECK_EQ(0, acm_->Add10MsData(input_frame_));
input_frame_.timestamp_ += input_block_size_samples_;
int32_t encoded_bytes = acm_->Process();
if (encoded_bytes > 0) {
// Encoded packet received.
return CreatePacket();
}
}
// Test ended.
return NULL;
}
// This method receives the callback from ACM when a new packet is produced.
int32_t AcmSendTestOldApi::SendData(
FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) {
// Store the packet locally.
frame_type_ = frame_type;
payload_type_ = payload_type;
timestamp_ = timestamp;
last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
assert(last_payload_vec_.size() == payload_len_bytes);
return 0;
}
Packet* AcmSendTestOldApi::CreatePacket() {
const size_t kRtpHeaderSize = 12;
size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize;
uint8_t* packet_memory = new uint8_t[allocated_bytes];
// Populate the header bytes.
packet_memory[0] = 0x80;
packet_memory[1] = payload_type_;
packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
packet_memory[3] = (sequence_number_) & 0xFF;
packet_memory[4] = (timestamp_ >> 24) & 0xFF;
packet_memory[5] = (timestamp_ >> 16) & 0xFF;
packet_memory[6] = (timestamp_ >> 8) & 0xFF;
packet_memory[7] = timestamp_ & 0xFF;
// Set SSRC to 0x12345678.
packet_memory[8] = 0x12;
packet_memory[9] = 0x34;
packet_memory[10] = 0x56;
packet_memory[11] = 0x78;
++sequence_number_;
// Copy the payload data.
memcpy(packet_memory + kRtpHeaderSize,
&last_payload_vec_[0],
last_payload_vec_.size());
Packet* packet =
new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
assert(packet);
assert(packet->valid_header());
return packet;
}
} // namespace test
} // namespace webrtc

86
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h Normal file
View File

@ -0,0 +1,86 @@
/*
* Copyright (c) 2014 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_ACM2_ACM_SEND_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
namespace test {
class InputAudioFile;
class Packet;
class AcmSendTestOldApi : public AudioPacketizationCallback,
public PacketSource {
public:
AcmSendTestOldApi(InputAudioFile* audio_source,
int source_rate_hz,
int test_duration_ms);
virtual ~AcmSendTestOldApi() {}
// Registers the send codec. Returns true on success, false otherwise.
bool RegisterCodec(const char* payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples);
// Returns the next encoded packet. Returns NULL if the test duration was
// exceeded. Ownership of the packet is handed over to the caller.
// Inherited from PacketSource.
Packet* NextPacket();
// Inherited from AudioPacketizationCallback.
virtual int32_t SendData(
FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:
static const int kBlockSizeMs = 10;
// Creates a Packet object from the last packet produced by ACM (and received
// through the SendData method as a callback). Ownership of the new Packet
// object is transferred to the caller.
Packet* CreatePacket();
SimulatedClock clock_;
scoped_ptr<AudioCodingModule> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
const int input_block_size_samples_;
AudioFrame input_frame_;
CodecInst codec_;
bool codec_registered_;
int test_duration_ms_;
// The following member variables are set whenever SendData() is called.
FrameType frame_type_;
int payload_type_;
uint32_t timestamp_;
uint16_t sequence_number_;
std::vector<uint8_t> last_payload_vec_;
DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_

12
webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
View File

@ -119,7 +119,11 @@
{
'target_name': 'acm_receive_test',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
],
'dependencies': [
'<@(audio_coding_dependencies)',
'audio_coding_module',
'neteq_unittest_tools',
'<(DEPTH)/testing/gtest.gyp:gtest',
@ -127,12 +131,18 @@
'sources': [
'acm_receive_test.cc',
'acm_receive_test.h',
'acm_receive_test_oldapi.cc',
'acm_receive_test_oldapi.h',
],
}, # acm_receive_test
{
'target_name': 'acm_send_test',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
],
'dependencies': [
'<@(audio_coding_dependencies)',
'audio_coding_module',
'neteq_unittest_tools',
'<(DEPTH)/testing/gtest.gyp:gtest',
@ -140,6 +150,8 @@
'sources': [
'acm_send_test.cc',
'acm_send_test.h',
'acm_send_test_oldapi.cc',
'acm_send_test_oldapi.h',
],
}, # acm_send_test
{

160
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
View File

@ -12,6 +12,7 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/md5digest.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h"
@ -118,19 +119,15 @@ class PacketizationCallbackStub : public AudioPacketizationCallback {
class AudioCodingModuleTest : public ::testing::Test {
protected:
AudioCodingModuleTest()
: id_(1),
rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
clock_(Clock::GetRealTimeClock()) {}
: rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) {
config_.transport = &packet_cb_;
}
~AudioCodingModuleTest() {}
void TearDown() OVERRIDE {}
void SetUp() OVERRIDE {
acm_.reset(AudioCodingModule::Create(id_, clock_));
RegisterCodec();
rtp_utility_->Populate(&rtp_header_);
input_frame_.sample_rate_hz_ = kSampleRateHz;
@ -141,17 +138,32 @@ class AudioCodingModuleTest : public ::testing::Test {
memset(input_frame_.data_,
0,
input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0]));
}
ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_));
void CreateAcm() {
acm_.reset(AudioCoding::Create(config_));
ASSERT_TRUE(acm_.get() != NULL);
RegisterCodec();
}
virtual void RegisterCodec() {
AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
// Register L16 codec in ACM.
ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
ASSERT_EQ(0, acm_->RegisterSendCodec(codec_));
int codec_type = acm2::ACMCodecDB::kNone;
switch (kSampleRateHz) {
case 8000:
codec_type = acm2::ACMCodecDB::kPCM16B;
break;
case 16000:
codec_type = acm2::ACMCodecDB::kPCM16Bwb;
break;
case 32000:
codec_type = acm2::ACMCodecDB::kPCM16Bswb32kHz;
break;
default:
FATAL() << "Sample rate not supported in this test.";
}
ASSERT_TRUE(acm_->RegisterSendCodec(codec_type, kPayloadType));
ASSERT_TRUE(acm_->RegisterReceiveCodec(codec_type, kPayloadType));
}
virtual void InsertPacketAndPullAudio() {
@ -161,41 +173,33 @@ class AudioCodingModuleTest : public ::testing::Test {
virtual void InsertPacket() {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
ASSERT_EQ(0,
acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
ASSERT_TRUE(acm_->InsertPacket(kPayload, kPayloadSizeBytes, rtp_header_));
rtp_utility_->Forward(&rtp_header_);
}
virtual void PullAudio() {
AudioFrame audio_frame;
ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame));
ASSERT_TRUE(acm_->Get10MsAudio(&audio_frame));
}
virtual void InsertAudio() {
ASSERT_EQ(0, acm_->Add10MsData(input_frame_));
int encoded_bytes = acm_->Add10MsAudio(input_frame_);
ASSERT_GE(encoded_bytes, 0);
input_frame_.timestamp_ += kNumSamples10ms;
}
virtual void Encode() {
int32_t encoded_bytes = acm_->Process();
// Expect to get one packet with two bytes per sample, or no packet at all,
// depending on how many 10 ms blocks go into |codec_.pacsize|.
EXPECT_TRUE(encoded_bytes == 2 * codec_.pacsize || encoded_bytes == 0);
}
const int id_;
AudioCoding::Config config_;
scoped_ptr<RtpUtility> rtp_utility_;
scoped_ptr<AudioCodingModule> acm_;
scoped_ptr<AudioCoding> acm_;
PacketizationCallbackStub packet_cb_;
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
CodecInst codec_;
Clock* clock_;
};
// Check if the statistics are initialized correctly. Before any call to ACM
// all fields have to be zero.
TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) {
CreateAcm();
AudioDecodingCallStats stats;
acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(0, stats.calls_to_neteq);
@ -209,10 +213,10 @@ TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) {
// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
// should result in generating silence, check the associated field.
TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
AudioDecodingCallStats stats;
const int kInitialDelay = 100;
acm_->SetInitialPlayoutDelay(kInitialDelay);
config_.initial_playout_delay_ms = kInitialDelay;
CreateAcm();
AudioDecodingCallStats stats;
int num_calls = 0;
for (int time_ms = 0; time_ms < kInitialDelay;
@ -232,6 +236,7 @@ TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
// simulate packet loss and check if PLC and PLC-to-CNG statistics are
// correctly updated.
TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) {
CreateAcm();
AudioDecodingCallStats stats;
const int kNumNormalCalls = 10;
@ -263,21 +268,16 @@ TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) {
}
TEST_F(AudioCodingModuleTest, VerifyOutputFrame) {
CreateAcm();
AudioFrame audio_frame;
const int kSampleRateHz = 32000;
EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
EXPECT_EQ(id_, audio_frame.id_);
EXPECT_TRUE(acm_->Get10MsAudio(&audio_frame));
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0);
EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
}
TEST_F(AudioCodingModuleTest, FailOnZeroDesiredFrequency) {
AudioFrame audio_frame;
EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame));
}
// A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz
// codec, while the derive class AcmIsacMtTest is using iSAC.
class AudioCodingModuleMtTest : public AudioCodingModuleTest {
@ -306,11 +306,12 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest {
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
next_insert_packet_time_ms_(0),
fake_clock_(new SimulatedClock(0)) {
clock_ = fake_clock_.get();
config_.clock = fake_clock_.get();
}
virtual void SetUp() OVERRIDE {
AudioCodingModuleTest::SetUp();
CreateAcm();
StartThreads();
}
@ -357,7 +358,6 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest {
}
++send_count_;
InsertAudio();
Encode();
if (TestDone()) {
test_complete_->Set();
}
@ -373,7 +373,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest {
SleepMs(1);
{
CriticalSectionScoped lock(crit_sect_.get());
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
if (fake_clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return true;
}
next_insert_packet_time_ms_ += 10;
@ -394,7 +394,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest {
{
CriticalSectionScoped lock(crit_sect_.get());
// Don't let the insert thread fall behind.
if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) {
if (next_insert_packet_time_ms_ < fake_clock_->TimeInMilliseconds()) {
return true;
}
++pull_audio_count_;
@ -439,6 +439,7 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest {
virtual void SetUp() OVERRIDE {
AudioCodingModuleTest::SetUp();
CreateAcm();
// Set up input audio source to read from specified file, loop after 5
// seconds, and deliver blocks of 10 ms.
@ -450,7 +451,6 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest {
int loop_counter = 0;
while (packet_cb_.last_payload_len_bytes() == 0) {
InsertAudio();
Encode();
ASSERT_LT(loop_counter++, 10);
}
// Set |last_packet_number_| to one less that |num_calls| so that the packet
@ -462,13 +462,12 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest {
virtual void RegisterCodec() OVERRIDE {
COMPILE_ASSERT(kSampleRateHz == 16000, test_designed_for_isac_16khz);
AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTest::SetUp();
ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
ASSERT_EQ(0, acm_->RegisterSendCodec(codec_));
ASSERT_TRUE(acm_->RegisterSendCodec(acm2::ACMCodecDB::kISAC, kPayloadType));
ASSERT_TRUE(
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, kPayloadType));
}
virtual void InsertPacket() OVERRIDE {
@ -484,10 +483,8 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest {
last_packet_number_ = num_calls;
}
ASSERT_GT(last_payload_vec_.size(), 0u);
ASSERT_EQ(
0,
acm_->IncomingPacket(
&last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
ASSERT_TRUE(acm_->InsertPacket(
&last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
}
virtual void InsertAudio() OVERRIDE {
@ -495,8 +492,6 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest {
AudioCodingModuleTest::InsertAudio();
}
virtual void Encode() OVERRIDE { ASSERT_GE(acm_->Process(), 0); }
// This method is the same as AudioCodingModuleMtTest::TestDone(), but here
// it is using the constants defined in this class (i.e., shorter test run).
virtual bool TestDone() OVERRIDE {
@ -634,19 +629,15 @@ class AcmSenderBitExactness : public ::testing::Test,
// Registers a send codec in the test::AcmSendTest object. Returns true on
// success, false on failure.
bool RegisterSendCodec(const char* payload_name,
int sampling_freq_hz,
bool RegisterSendCodec(int codec_type,
int channels,
int payload_type,
int frame_size_samples,
int frame_size_rtp_timestamps) {
payload_type_ = payload_type;
frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
return send_test_->RegisterCodec(payload_name,
sampling_freq_hz,
channels,
payload_type,
frame_size_samples);
return send_test_->RegisterCodec(
codec_type, channels, payload_type, frame_size_samples);
}
// Runs the test. SetUpSender() and RegisterSendCodec() must have been called
@ -728,15 +719,13 @@ class AcmSenderBitExactness : public ::testing::Test,
payload_checksum_.Update(packet->payload(), packet->payload_length_bytes());
}
void SetUpTest(const char* codec_name,
int codec_sample_rate_hz,
void SetUpTest(int codec_type,
int channels,
int payload_type,
int codec_frame_size_samples,
int codec_frame_size_rtp_timestamps) {
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterSendCodec(codec_name,
codec_sample_rate_hz,
ASSERT_TRUE(RegisterSendCodec(codec_type,
channels,
payload_type,
codec_frame_size_samples,
@ -754,7 +743,7 @@ class AcmSenderBitExactness : public ::testing::Test,
};
TEST_F(AcmSenderBitExactness, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 480, 480));
Run(AcmReceiverBitExactness::PlatformChecksum(
"c7e5bdadfa2871df95639fcc297cf23d",
"0499ca260390769b3172136faad925b9",
@ -768,7 +757,7 @@ TEST_F(AcmSenderBitExactness, IsacWb30ms) {
}
TEST_F(AcmSenderBitExactness, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 960, 960));
Run(AcmReceiverBitExactness::PlatformChecksum(
"14d63c5f08127d280e722e3191b73bdd",
"8da003e16c5371af2dc2be79a50f9076",
@ -782,7 +771,8 @@ TEST_F(AcmSenderBitExactness, IsacWb60ms) {
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960));
Run(AcmReceiverBitExactness::PlatformChecksum(
"98d960600eb4ddb3fcbe11f5057ddfd7",
"",
@ -796,7 +786,7 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) {
}
TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCM16B, 1, 107, 80, 80));
Run("de4a98e1406f8b798d99cd0704e862e2",
"c1edd36339ce0326cc4550041ad719a0",
100,
@ -804,7 +794,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) {
}
TEST_F(AcmSenderBitExactness, Pcm16_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kPCM16Bwb, 1, 108, 160, 160));
Run("ae646d7b68384a1269cc080dd4501916",
"ad786526383178b08d80d6eee06e9bad",
100,
@ -812,7 +803,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_16000khz_10ms) {
}
TEST_F(AcmSenderBitExactness, Pcm16_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz, 1, 109, 320, 320));
Run("7fe325e8fbaf755e3c5df0b11a4774fb",
"5ef82ea885e922263606c6fdbc49f651",
100,
@ -820,7 +812,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_32000khz_10ms) {
}
TEST_F(AcmSenderBitExactness, Pcm16_stereo_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kPCM16B_2ch, 2, 111, 80, 80));
Run("fb263b74e7ac3de915474d77e4744ceb",
"62ce5adb0d4965d0a52ec98ae7f98974",
100,
@ -828,7 +821,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_stereo_8000khz_10ms) {
}
TEST_F(AcmSenderBitExactness, Pcm16_stereo_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kPCM16Bwb_2ch, 2, 112, 160, 160));
Run("d09e9239553649d7ac93e19d304281fd",
"41ca8edac4b8c71cd54fd9f25ec14870",
100,
@ -836,7 +830,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_stereo_16000khz_10ms) {
}
TEST_F(AcmSenderBitExactness, Pcm16_stereo_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 2, 113, 320, 320));
Run("5f025d4f390982cc26b3d92fe02e3044",
"50e58502fb04421bf5b857dda4c96879",
100,
@ -844,7 +839,7 @@ TEST_F(AcmSenderBitExactness, Pcm16_stereo_32000khz_10ms) {
}
TEST_F(AcmSenderBitExactness, Pcmu_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMU, 1, 0, 160, 160));
Run("81a9d4c0bb72e9becc43aef124c981e9",
"8f9b8750bd80fe26b6cbf6659b89f0f9",
50,
@ -852,7 +847,7 @@ TEST_F(AcmSenderBitExactness, Pcmu_20ms) {
}
TEST_F(AcmSenderBitExactness, Pcma_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMA, 1, 8, 160, 160));
Run("39611f798969053925a49dc06d08de29",
"6ad745e55aa48981bfc790d0eeef2dd1",
50,
@ -860,7 +855,8 @@ TEST_F(AcmSenderBitExactness, Pcma_20ms) {
}
TEST_F(AcmSenderBitExactness, Pcmu_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kPCMU_2ch, 2, 110, 160, 160));
Run("437bec032fdc5cbaa0d5175430af7b18",
"60b6f25e8d1e74cb679cfe756dd9bca5",
50,
@ -868,7 +864,8 @@ TEST_F(AcmSenderBitExactness, Pcmu_stereo_20ms) {
}
TEST_F(AcmSenderBitExactness, Pcma_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kPCMA_2ch, 2, 118, 160, 160));
Run("a5c6d83c5b7cedbeff734238220a4b0c",
"92b282c83efd20e7eeef52ba40842cf7",
50,
@ -876,7 +873,7 @@ TEST_F(AcmSenderBitExactness, Pcma_stereo_20ms) {
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(Ilbc_30ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kILBC, 1, 102, 240, 240));
Run(AcmReceiverBitExactness::PlatformChecksum(
"7b6ec10910debd9af08011d3ed5249f7",
"android_audio",
@ -890,7 +887,7 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(Ilbc_30ms)) {
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_20ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kG722, 1, 9, 320, 160));
Run(AcmReceiverBitExactness::PlatformChecksum(
"7d759436f2533582950d148b5161a36c",
"android_audio",
@ -904,7 +901,8 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_20ms)) {
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_stereo_20ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kG722_2ch, 2, 119, 320, 160));
Run(AcmReceiverBitExactness::PlatformChecksum(
"7190ee718ab3d80eca181e5f7140c210",
"android_audio",
@ -918,7 +916,7 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_stereo_20ms)) {
}
TEST_F(AcmSenderBitExactness, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kOpus, 2, 120, 960, 960));
Run(AcmReceiverBitExactness::PlatformChecksum(
"855041f2490b887302bce9d544731849",
"1e1a0fce893fef2d66886a7f09e2ebce",

938
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc Normal file
View File

@ -0,0 +1,938 @@
/*
* Copyright (c) 2014 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 <string.h>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/md5digest.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.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/neteq/tools/audio_checksum.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/compile_assert.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
const int kSampleRateHz = 16000;
const int kNumSamples10ms = kSampleRateHz / 100;
const int kFrameSizeMs = 10; // Multiple of 10.
const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
const uint8_t kPayloadType = 111;
class RtpUtility {
public:
RtpUtility(int samples_per_packet, uint8_t payload_type)
: samples_per_packet_(samples_per_packet), payload_type_(payload_type) {}
virtual ~RtpUtility() {}
void Populate(WebRtcRTPHeader* rtp_header) {
rtp_header->header.sequenceNumber = 0xABCD;
rtp_header->header.timestamp = 0xABCDEF01;
rtp_header->header.payloadType = payload_type_;
rtp_header->header.markerBit = false;
rtp_header->header.ssrc = 0x1234;
rtp_header->header.numCSRCs = 0;
rtp_header->frameType = kAudioFrameSpeech;
rtp_header->header.payload_type_frequency = kSampleRateHz;
rtp_header->type.Audio.channel = 1;
rtp_header->type.Audio.isCNG = false;
}
void Forward(WebRtcRTPHeader* rtp_header) {
++rtp_header->header.sequenceNumber;
rtp_header->header.timestamp += samples_per_packet_;
}
private:
int samples_per_packet_;
uint8_t payload_type_;
};
class PacketizationCallbackStub : public AudioPacketizationCallback {
public:
PacketizationCallbackStub()
: num_calls_(0),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
virtual int32_t SendData(
FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;
last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
return 0;
}
int num_calls() const {
CriticalSectionScoped lock(crit_sect_.get());
return num_calls_;
}
int last_payload_len_bytes() const {
CriticalSectionScoped lock(crit_sect_.get());
return last_payload_vec_.size();
}
void SwapBuffers(std::vector<uint8_t>* payload) {
CriticalSectionScoped lock(crit_sect_.get());
last_payload_vec_.swap(*payload);
}
private:
int num_calls_ GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
const scoped_ptr<CriticalSectionWrapper> crit_sect_;
};
class AudioCodingModuleTestOldApi : public ::testing::Test {
protected:
AudioCodingModuleTestOldApi()
: id_(1),
rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
clock_(Clock::GetRealTimeClock()) {}
~AudioCodingModuleTestOldApi() {}
void TearDown() {}
void SetUp() {
acm_.reset(AudioCodingModule::Create(id_, clock_));
RegisterCodec();
rtp_utility_->Populate(&rtp_header_);
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000; // 10 ms.
COMPILE_ASSERT(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples,
audio_frame_too_small);
memset(input_frame_.data_,
0,
input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0]));
ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_));
}
virtual void RegisterCodec() {
AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
// Register L16 codec in ACM.
ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
ASSERT_EQ(0, acm_->RegisterSendCodec(codec_));
}
virtual void InsertPacketAndPullAudio() {
InsertPacket();
PullAudio();
}
virtual void InsertPacket() {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
ASSERT_EQ(0,
acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
rtp_utility_->Forward(&rtp_header_);
}
virtual void PullAudio() {
AudioFrame audio_frame;
ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame));
}
virtual void InsertAudio() {
ASSERT_EQ(0, acm_->Add10MsData(input_frame_));
input_frame_.timestamp_ += kNumSamples10ms;
}
virtual void Encode() {
int32_t encoded_bytes = acm_->Process();
// Expect to get one packet with two bytes per sample, or no packet at all,
// depending on how many 10 ms blocks go into |codec_.pacsize|.
EXPECT_TRUE(encoded_bytes == 2 * codec_.pacsize || encoded_bytes == 0);
}
const int id_;
scoped_ptr<RtpUtility> rtp_utility_;
scoped_ptr<AudioCodingModule> acm_;
PacketizationCallbackStub packet_cb_;
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
CodecInst codec_;
Clock* clock_;
};
// Check if the statistics are initialized correctly. Before any call to ACM
// all fields have to be zero.
TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
AudioDecodingCallStats stats;
acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
}
// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
// should result in generating silence, check the associated field.
TEST_F(AudioCodingModuleTestOldApi,
DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
AudioDecodingCallStats stats;
const int kInitialDelay = 100;
acm_->SetInitialPlayoutDelay(kInitialDelay);
int num_calls = 0;
for (int time_ms = 0; time_ms < kInitialDelay;
time_ms += kFrameSizeMs, ++num_calls) {
InsertPacketAndPullAudio();
}
acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
}
// Insert some packets and pull audio. Check statistics are valid. Then,
// simulate packet loss and check if PLC and PLC-to-CNG statistics are
// correctly updated.
TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(NetEqCalls)) {
AudioDecodingCallStats stats;
const int kNumNormalCalls = 10;
for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
InsertPacketAndPullAudio();
}
acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
const int kNumPlc = 3;
const int kNumPlcCng = 5;
// Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG.
for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
PullAudio();
}
acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(kNumPlc, stats.decoded_plc);
EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
}
TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
AudioFrame audio_frame;
const int kSampleRateHz = 32000;
EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
EXPECT_EQ(id_, audio_frame.id_);
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0);
EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
}
TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
AudioFrame audio_frame;
EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame));
}
// A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz
// codec, while the derive class AcmIsacMtTest is using iSAC.
class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
protected:
static const int kNumPackets = 500;
static const int kNumPullCalls = 500;
AudioCodingModuleMtTestOldApi()
: AudioCodingModuleTestOldApi(),
send_thread_(ThreadWrapper::CreateThread(CbSendThread,
this,
kRealtimePriority,
"send")),
insert_packet_thread_(ThreadWrapper::CreateThread(CbInsertPacketThread,
this,
kRealtimePriority,
"insert_packet")),
pull_audio_thread_(ThreadWrapper::CreateThread(CbPullAudioThread,
this,
kRealtimePriority,
"pull_audio")),
test_complete_(EventWrapper::Create()),
send_count_(0),
insert_packet_count_(0),
pull_audio_count_(0),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
next_insert_packet_time_ms_(0),
fake_clock_(new SimulatedClock(0)) {
clock_ = fake_clock_.get();
}
void SetUp() {
AudioCodingModuleTestOldApi::SetUp();
StartThreads();
}
void StartThreads() {
unsigned int thread_id = 0;
ASSERT_TRUE(send_thread_->Start(thread_id));
ASSERT_TRUE(insert_packet_thread_->Start(thread_id));
ASSERT_TRUE(pull_audio_thread_->Start(thread_id));
}
void TearDown() {
AudioCodingModuleTestOldApi::TearDown();
pull_audio_thread_->Stop();
send_thread_->Stop();
insert_packet_thread_->Stop();
}
EventTypeWrapper RunTest() {
return test_complete_->Wait(10 * 60 * 1000); // 10 minutes' timeout.
}
virtual bool TestDone() {
if (packet_cb_.num_calls() > kNumPackets) {
CriticalSectionScoped lock(crit_sect_.get());
if (pull_audio_count_ > kNumPullCalls) {
// Both conditions for completion are met. End the test.
return true;
}
}
return false;
}
static bool CbSendThread(void* context) {
return reinterpret_cast<AudioCodingModuleMtTestOldApi*>(context)
->CbSendImpl();
}
// The send thread doesn't have to care about the current simulated time,
// since only the AcmReceiver is using the clock.
bool CbSendImpl() {
SleepMs(1);
if (HasFatalFailure()) {
// End the test early if a fatal failure (ASSERT_*) has occurred.
test_complete_->Set();
}
++send_count_;
InsertAudio();
Encode();
if (TestDone()) {
test_complete_->Set();
}
return true;
}
static bool CbInsertPacketThread(void* context) {
return reinterpret_cast<AudioCodingModuleMtTestOldApi*>(context)
->CbInsertPacketImpl();
}
bool CbInsertPacketImpl() {
SleepMs(1);
{
CriticalSectionScoped lock(crit_sect_.get());
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return true;
}
next_insert_packet_time_ms_ += 10;
}
// Now we're not holding the crit sect when calling ACM.
++insert_packet_count_;
InsertPacket();
return true;
}
static bool CbPullAudioThread(void* context) {
return reinterpret_cast<AudioCodingModuleMtTestOldApi*>(context)
->CbPullAudioImpl();
}
bool CbPullAudioImpl() {
SleepMs(1);
{
CriticalSectionScoped lock(crit_sect_.get());
// Don't let the insert thread fall behind.
if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) {
return true;
}
++pull_audio_count_;
}
// Now we're not holding the crit sect when calling ACM.
PullAudio();
fake_clock_->AdvanceTimeMilliseconds(10);
return true;
}
scoped_ptr<ThreadWrapper> send_thread_;
scoped_ptr<ThreadWrapper> insert_packet_thread_;
scoped_ptr<ThreadWrapper> pull_audio_thread_;
const scoped_ptr<EventWrapper> test_complete_;
int send_count_;
int insert_packet_count_;
int pull_audio_count_ GUARDED_BY(crit_sect_);
const scoped_ptr<CriticalSectionWrapper> crit_sect_;
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
scoped_ptr<SimulatedClock> fake_clock_;
};
TEST_F(AudioCodingModuleMtTestOldApi, DoTest) {
EXPECT_EQ(kEventSignaled, RunTest());
}
// This is a multi-threaded ACM test using iSAC. The test encodes audio
// from a PCM file. The most recent encoded frame is used as input to the
// receiving part. Depending on timing, it may happen that the same RTP packet
// is inserted into the receiver multiple times, but this is a valid use-case,
// and simplifies the test code a lot.
class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
protected:
static const int kNumPackets = 500;
static const int kNumPullCalls = 500;
AcmIsacMtTestOldApi()
: AudioCodingModuleMtTestOldApi(), last_packet_number_(0) {}
~AcmIsacMtTestOldApi() {}
void SetUp() {
AudioCodingModuleTestOldApi::SetUp();
// Set up input audio source to read from specified file, loop after 5
// seconds, and deliver blocks of 10 ms.
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
// Generate one packet to have something to insert.
int loop_counter = 0;
while (packet_cb_.last_payload_len_bytes() == 0) {
InsertAudio();
Encode();
ASSERT_LT(loop_counter++, 10);
}
// Set |last_packet_number_| to one less that |num_calls| so that the packet
// will be fetched in the next InsertPacket() call.
last_packet_number_ = packet_cb_.num_calls() - 1;
StartThreads();
}
virtual void RegisterCodec() {
COMPILE_ASSERT(kSampleRateHz == 16000, test_designed_for_isac_16khz);
AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
ASSERT_EQ(0, acm_->RegisterSendCodec(codec_));
}
void InsertPacket() {
int num_calls = packet_cb_.num_calls(); // Store locally for thread safety.
if (num_calls > last_packet_number_) {
// Get the new payload out from the callback handler.
// Note that since we swap buffers here instead of directly inserting
// a pointer to the data in |packet_cb_|, we avoid locking the callback
// for the duration of the IncomingPacket() call.
packet_cb_.SwapBuffers(&last_payload_vec_);
ASSERT_GT(last_payload_vec_.size(), 0u);
rtp_utility_->Forward(&rtp_header_);
last_packet_number_ = num_calls;
}
ASSERT_GT(last_payload_vec_.size(), 0u);
ASSERT_EQ(
0,
acm_->IncomingPacket(
&last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
}
void InsertAudio() {
memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
AudioCodingModuleTestOldApi::InsertAudio();
}
void Encode() { ASSERT_GE(acm_->Process(), 0); }
// This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but
// here it is using the constants defined in this class (i.e., shorter test
// run).
virtual bool TestDone() {
if (packet_cb_.num_calls() > kNumPackets) {
CriticalSectionScoped lock(crit_sect_.get());
if (pull_audio_count_ > kNumPullCalls) {
// Both conditions for completion are met. End the test.
return true;
}
}
return false;
}
int last_packet_number_;
std::vector<uint8_t> last_payload_vec_;
test::AudioLoop audio_loop_;
};
TEST_F(AcmIsacMtTestOldApi, DoTest) {
EXPECT_EQ(kEventSignaled, RunTest());
}
class AcmReceiverBitExactnessOldApi : public ::testing::Test {
public:
static std::string PlatformChecksum(std::string win64,
std::string android,
std::string others) {
#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
return win64;
#elif defined(WEBRTC_ANDROID)
return android;
#else
return others;
#endif
}
protected:
void Run(int output_freq_hz, const std::string& checksum_ref) {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
scoped_ptr<test::RtpFileSource> packet_source(
test::RtpFileSource::Create(input_file_name));
#ifdef WEBRTC_ANDROID
// Filter out iLBC and iSAC-swb since they are not supported on Android.
packet_source->FilterOutPayloadType(102); // iLBC.
packet_source->FilterOutPayloadType(104); // iSAC-swb.
#endif
test::AudioChecksum checksum;
const std::string output_file_name =
webrtc::test::OutputPath() +
::testing::UnitTest::GetInstance()
->current_test_info()
->test_case_name() +
"_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
"_output.pcm";
test::OutputAudioFile output_file(output_file_name);
test::AudioSinkFork output(&checksum, &output_file);
test::AcmReceiveTestOldApi test(
packet_source.get(),
&output,
output_freq_hz,
test::AcmReceiveTestOldApi::kArbitraryChannels);
ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs());
test.Run();
std::string checksum_string = checksum.Finish();
EXPECT_EQ(checksum_ref, checksum_string);
}
};
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
Run(8000,
PlatformChecksum("bd6f8d9602cd82444ea2539e674df747",
"6ac89c7145072c26bfeba602cd661afb",
"8a8440f5511eb729221b9aac25cda3a0"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
Run(16000,
PlatformChecksum("a39bc6ee0c4eb15f3ad2f43cebcc571d",
"3e888eb04f57db2c6ef952fe64f17fe6",
"7be583092c5adbcb0f6cd66eca20ea63"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
Run(32000,
PlatformChecksum("80964572aaa2dc92f9e34896dd3802b3",
"aeca37e963310f5b6552b7edea23c2f1",
"3a84188abe9fca25fedd6034760f3e22"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
Run(48000,
PlatformChecksum("8aacde91f390e0d5a9c2ed571a25fd37",
"76b9e99e0a3998aa28355e7a2bd836f7",
"89b4b19bdb4de40f1d88302ef8cb9f9b"));
}
// This test verifies bit exactness for the send-side of ACM. The test setup is
// a chain of three different test classes:
//
// test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
//
// The receiver side is driving the test by requesting new packets from
// AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
// packet from test::AcmSendTest::NextPacket, which inserts audio from the
// input file until one packet is produced. (The input file loops indefinitely.)
// Before passing the packet to the receiver, this test class verifies the
// packet header and updates a payload checksum with the new payload. The
// decoded output from the receiver is also verified with a (separate) checksum.
class AcmSenderBitExactnessOldApi : public ::testing::Test,
public test::PacketSource {
protected:
static const int kTestDurationMs = 1000;
AcmSenderBitExactnessOldApi()
: frame_size_rtp_timestamps_(0),
packet_count_(0),
payload_type_(0),
last_sequence_number_(0),
last_timestamp_(0) {}
// Sets up the test::AcmSendTest object. Returns true on success, otherwise
// false.
bool SetUpSender() {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
// Note that |audio_source_| will loop forever. The test duration is set
// explicitly by |kTestDurationMs|.
audio_source_.reset(new test::InputAudioFile(input_file_name));
static const int kSourceRateHz = 32000;
send_test_.reset(new test::AcmSendTestOldApi(
audio_source_.get(), kSourceRateHz, kTestDurationMs));
return send_test_.get() != NULL;
}
// Registers a send codec in the test::AcmSendTest object. Returns true on
// success, false on failure.
bool RegisterSendCodec(const char* payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples,
int frame_size_rtp_timestamps) {
payload_type_ = payload_type;
frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
return send_test_->RegisterCodec(payload_name,
sampling_freq_hz,
channels,
payload_type,
frame_size_samples);
}
// Runs the test. SetUpSender() and RegisterSendCodec() must have been called
// before calling this method.
void Run(const std::string& audio_checksum_ref,
const std::string& payload_checksum_ref,
int expected_packets,
test::AcmReceiveTestOldApi::NumOutputChannels expected_channels) {
// Set up the receiver used to decode the packets and verify the decoded
// output.
test::AudioChecksum audio_checksum;
const std::string output_file_name =
webrtc::test::OutputPath() +
::testing::UnitTest::GetInstance()
->current_test_info()
->test_case_name() +
"_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
"_output.pcm";
test::OutputAudioFile output_file(output_file_name);
// Have the output audio sent both to file and to the checksum calculator.
test::AudioSinkFork output(&audio_checksum, &output_file);
const int kOutputFreqHz = 8000;
test::AcmReceiveTestOldApi receive_test(
this, &output, kOutputFreqHz, expected_channels);
ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
// This is where the actual test is executed.
receive_test.Run();
// Extract and verify the audio checksum.
std::string checksum_string = audio_checksum.Finish();
EXPECT_EQ(audio_checksum_ref, checksum_string);
// Extract and verify the payload checksum.
char checksum_result[rtc::Md5Digest::kSize];
payload_checksum_.Finish(checksum_result, rtc::Md5Digest::kSize);
checksum_string = rtc::hex_encode(checksum_result, rtc::Md5Digest::kSize);
EXPECT_EQ(payload_checksum_ref, checksum_string);
// Verify number of packets produced.
EXPECT_EQ(expected_packets, packet_count_);
}
// Returns a pointer to the next packet. Returns NULL if the source is
// depleted (i.e., the test duration is exceeded), or if an error occurred.
// Inherited from test::PacketSource.
test::Packet* NextPacket() OVERRIDE {
// Get the next packet from AcmSendTest. Ownership of |packet| is
// transferred to this method.
test::Packet* packet = send_test_->NextPacket();
if (!packet)
return NULL;
VerifyPacket(packet);
// TODO(henrik.lundin) Save the packet to file as well.
// Pass it on to the caller. The caller becomes the owner of |packet|.
return packet;
}
// Verifies the packet.
void VerifyPacket(const test::Packet* packet) {
EXPECT_TRUE(packet->valid_header());
// (We can check the header fields even if valid_header() is false.)
EXPECT_EQ(payload_type_, packet->header().payloadType);
if (packet_count_ > 0) {
// This is not the first packet.
uint16_t sequence_number_diff =
packet->header().sequenceNumber - last_sequence_number_;
EXPECT_EQ(1, sequence_number_diff);
uint32_t timestamp_diff = packet->header().timestamp - last_timestamp_;
EXPECT_EQ(frame_size_rtp_timestamps_, timestamp_diff);
}
++packet_count_;
last_sequence_number_ = packet->header().sequenceNumber;
last_timestamp_ = packet->header().timestamp;
// Update the checksum.
payload_checksum_.Update(packet->payload(), packet->payload_length_bytes());
}
void SetUpTest(const char* codec_name,
int codec_sample_rate_hz,
int channels,
int payload_type,
int codec_frame_size_samples,
int codec_frame_size_rtp_timestamps) {
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterSendCodec(codec_name,
codec_sample_rate_hz,
channels,
payload_type,
codec_frame_size_samples,
codec_frame_size_rtp_timestamps));
}
scoped_ptr<test::AcmSendTestOldApi> send_test_;
scoped_ptr<test::InputAudioFile> audio_source_;
uint32_t frame_size_rtp_timestamps_;
int packet_count_;
uint8_t payload_type_;
uint16_t last_sequence_number_;
uint32_t last_timestamp_;
rtc::Md5Digest payload_checksum_;
};
TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"c7e5bdadfa2871df95639fcc297cf23d",
"0499ca260390769b3172136faad925b9",
"0b58f9eeee43d5891f5f6c75e77984a3"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"d42cb5195463da26c8129bbfe73a22e6",
"83de248aea9c3c2bd680b6952401b4ca",
"3c79f16f34218271f3dca4e2b1dfe1bb"),
33,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"14d63c5f08127d280e722e3191b73bdd",
"8da003e16c5371af2dc2be79a50f9076",
"1ad29139a04782a33daad8c2b9b35875"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"ebe04a819d3a9d83a83a17f271e1139a",
"97aeef98553b5a4b5a68f8b716e8eaf0",
"9e0a0ab743ad987b55b8e14802769c56"),
16,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IsacSwb30ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"98d960600eb4ddb3fcbe11f5057ddfd7",
"",
"2f6dfe142f735f1d96f6bd86d2526f42"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"cc9d2d86a71d6f99f97680a5c27e2762",
"",
"7b214fc3a5e33d68bf30e77969371f31"),
33,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
Run("de4a98e1406f8b798d99cd0704e862e2",
"c1edd36339ce0326cc4550041ad719a0",
100,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160));
Run("ae646d7b68384a1269cc080dd4501916",
"ad786526383178b08d80d6eee06e9bad",
100,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320));
Run("7fe325e8fbaf755e3c5df0b11a4774fb",
"5ef82ea885e922263606c6fdbc49f651",
100,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80));
Run("fb263b74e7ac3de915474d77e4744ceb",
"62ce5adb0d4965d0a52ec98ae7f98974",
100,
test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160));
Run("d09e9239553649d7ac93e19d304281fd",
"41ca8edac4b8c71cd54fd9f25ec14870",
100,
test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320));
Run("5f025d4f390982cc26b3d92fe02e3044",
"50e58502fb04421bf5b857dda4c96879",
100,
test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160));
Run("81a9d4c0bb72e9becc43aef124c981e9",
"8f9b8750bd80fe26b6cbf6659b89f0f9",
50,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160));
Run("39611f798969053925a49dc06d08de29",
"6ad745e55aa48981bfc790d0eeef2dd1",
50,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160));
Run("437bec032fdc5cbaa0d5175430af7b18",
"60b6f25e8d1e74cb679cfe756dd9bca5",
50,
test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160));
Run("a5c6d83c5b7cedbeff734238220a4b0c",
"92b282c83efd20e7eeef52ba40842cf7",
50,
test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(Ilbc_30ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7b6ec10910debd9af08011d3ed5249f7",
"android_audio",
"7b6ec10910debd9af08011d3ed5249f7"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"cfae2e9f6aba96e145f2bcdd5050ce78",
"android_payload",
"cfae2e9f6aba96e145f2bcdd5050ce78"),
33,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_20ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7d759436f2533582950d148b5161a36c",
"android_audio",
"7d759436f2533582950d148b5161a36c"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"fc68a87e1380614e658087cb35d5ca10",
"android_payload",
"fc68a87e1380614e658087cb35d5ca10"),
50,
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_stereo_20ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7190ee718ab3d80eca181e5f7140c210",
"android_audio",
"7190ee718ab3d80eca181e5f7140c210"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"66516152eeaa1e650ad94ff85f668dac",
"android_payload",
"66516152eeaa1e650ad94ff85f668dac"),
50,
test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"855041f2490b887302bce9d544731849",
"1e1a0fce893fef2d66886a7f09e2ebce",
"855041f2490b887302bce9d544731849"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"d781cce1ab986b618d0da87226cdde30",
"1a1fe04dd12e755949987c8d729fb3e0",
"d781cce1ab986b618d0da87226cdde30"),
50,
test::AcmReceiveTestOldApi::kStereoOutput);
}
} // namespace webrtc

1
webrtc/modules/modules.gyp
View File

@ -106,6 +106,7 @@
'audio_coding/main/acm2/acm_receiver_unittest.cc',
'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc',
'audio_coding/main/acm2/audio_coding_module_unittest.cc',
'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc',
'audio_coding/main/acm2/call_statistics_unittest.cc',
'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
'audio_coding/main/acm2/nack_unittest.cc',