blob: 39c14a880cab9b02b91188f848d5f844ae593bc1 [file] [log] [blame]
/*
* 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/base/platform_thread.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.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/constant_pcm_packet_source.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/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
using ::testing::AtLeast;
using ::testing::Invoke;
using ::testing::_;
namespace webrtc {
namespace {
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;
} // namespace
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 PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
public:
PacketizationCallbackStubOldApi()
: num_calls_(0),
last_frame_type_(kEmptyFrame),
last_payload_type_(-1),
last_timestamp_(0),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
int32_t SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) override {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;
last_frame_type_ = frame_type;
last_payload_type_ = payload_type;
last_timestamp_ = timestamp;
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();
}
FrameType last_frame_type() const {
CriticalSectionScoped lock(crit_sect_.get());
return last_frame_type_;
}
int last_payload_type() const {
CriticalSectionScoped lock(crit_sect_.get());
return last_payload_type_;
}
uint32_t last_timestamp() const {
CriticalSectionScoped lock(crit_sect_.get());
return last_timestamp_;
}
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_);
FrameType last_frame_type_ GUARDED_BY(crit_sect_);
int last_payload_type_ GUARDED_BY(crit_sect_);
uint32_t last_timestamp_ GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
const rtc::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_));
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.
static_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_));
SetUpL16Codec();
}
// Set up L16 codec.
virtual void SetUpL16Codec() {
ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1));
codec_.pltype = kPayloadType;
}
virtual void RegisterCodec() {
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_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kNumSamples10ms;
}
virtual void VerifyEncoding() {
int last_length = packet_cb_.last_payload_len_bytes();
EXPECT_TRUE(last_length == 2 * codec_.pacsize || last_length == 0)
<< "Last encoded packet was " << last_length << " bytes.";
}
virtual void InsertAudioAndVerifyEncoding() {
InsertAudio();
VerifyEncoding();
}
const int id_;
rtc::scoped_ptr<RtpUtility> rtp_utility_;
rtc::scoped_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi 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)) {
RegisterCodec();
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);
}
// 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)) {
RegisterCodec();
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(static_cast<size_t>(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));
}
// Checks that the transport callback is invoked once for each speech packet.
// Also checks that the frame type is kAudioFrameSpeech.
TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
const int k10MsBlocksPerPacket = 3;
codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
RegisterCodec();
const int kLoops = 10;
for (int i = 0; i < kLoops; ++i) {
EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls());
if (packet_cb_.num_calls() > 0)
EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
InsertAudioAndVerifyEncoding();
}
EXPECT_EQ(kLoops / k10MsBlocksPerPacket, packet_cb_.num_calls());
EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define IF_ISAC(x) x
#else
#define IF_ISAC(x) DISABLED_##x
#endif
// Verifies that the RTP timestamp series is not reset when the codec is
// changed.
TEST_F(AudioCodingModuleTestOldApi,
IF_ISAC(TimestampSeriesContinuesWhenCodecChanges)) {
RegisterCodec(); // This registers the default codec.
uint32_t expected_ts = input_frame_.timestamp_;
int blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
// Encode 5 packets of the first codec type.
const int kNumPackets1 = 5;
for (int j = 0; j < kNumPackets1; ++j) {
for (int i = 0; i < blocks_per_packet; ++i) {
EXPECT_EQ(j, packet_cb_.num_calls());
InsertAudio();
}
EXPECT_EQ(j + 1, packet_cb_.num_calls());
EXPECT_EQ(expected_ts, packet_cb_.last_timestamp());
expected_ts += codec_.pacsize;
}
// Change codec.
ASSERT_EQ(0, AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1));
RegisterCodec();
blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
// Encode another 5 packets.
const int kNumPackets2 = 5;
for (int j = 0; j < kNumPackets2; ++j) {
for (int i = 0; i < blocks_per_packet; ++i) {
EXPECT_EQ(kNumPackets1 + j, packet_cb_.num_calls());
InsertAudio();
}
EXPECT_EQ(kNumPackets1 + j + 1, packet_cb_.num_calls());
EXPECT_EQ(expected_ts, packet_cb_.last_timestamp());
expected_ts += codec_.pacsize;
}
}
// Introduce this class to set different expectations on the number of encoded
// bytes. This class expects all encoded packets to be 9 bytes (matching one
// CNG SID frame) or 0 bytes. This test depends on |input_frame_| containing
// (near-)zero values. It also introduces a way to register comfort noise with
// a custom payload type.
class AudioCodingModuleTestWithComfortNoiseOldApi
: public AudioCodingModuleTestOldApi {
protected:
void RegisterCngCodec(int rtp_payload_type) {
CodecInst codec;
AudioCodingModule::Codec("CN", &codec, kSampleRateHz, 1);
codec.pltype = rtp_payload_type;
ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec));
ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
}
void VerifyEncoding() override {
int last_length = packet_cb_.last_payload_len_bytes();
EXPECT_TRUE(last_length == 9 || last_length == 0)
<< "Last encoded packet was " << last_length << " bytes.";
}
void DoTest(int blocks_per_packet, int cng_pt) {
const int kLoops = 40;
// This array defines the expected frame types, and when they should arrive.
// We expect a frame to arrive each time the speech encoder would have
// produced a packet, and once every 100 ms the frame should be non-empty,
// that is contain comfort noise.
const struct {
int ix;
FrameType type;
} expectation[] = {{2, kAudioFrameCN},
{5, kEmptyFrame},
{8, kEmptyFrame},
{11, kAudioFrameCN},
{14, kEmptyFrame},
{17, kEmptyFrame},
{20, kAudioFrameCN},
{23, kEmptyFrame},
{26, kEmptyFrame},
{29, kEmptyFrame},
{32, kAudioFrameCN},
{35, kEmptyFrame},
{38, kEmptyFrame}};
for (int i = 0; i < kLoops; ++i) {
int num_calls_before = packet_cb_.num_calls();
EXPECT_EQ(i / blocks_per_packet, num_calls_before);
InsertAudioAndVerifyEncoding();
int num_calls = packet_cb_.num_calls();
if (num_calls == num_calls_before + 1) {
EXPECT_EQ(expectation[num_calls - 1].ix, i);
EXPECT_EQ(expectation[num_calls - 1].type, packet_cb_.last_frame_type())
<< "Wrong frame type for lap " << i;
EXPECT_EQ(cng_pt, packet_cb_.last_payload_type());
} else {
EXPECT_EQ(num_calls, num_calls_before);
}
}
}
};
// Checks that the transport callback is invoked once per frame period of the
// underlying speech encoder, even when comfort noise is produced.
// Also checks that the frame type is kAudioFrameCN or kEmptyFrame.
// This test and the next check the same thing, but differ in the order of
// speech codec and CNG registration.
TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
TransportCallbackTestForComfortNoiseRegisterCngLast) {
const int k10MsBlocksPerPacket = 3;
codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
RegisterCodec();
const int kCngPayloadType = 105;
RegisterCngCodec(kCngPayloadType);
ASSERT_EQ(0, acm_->SetVAD(true, true));
DoTest(k10MsBlocksPerPacket, kCngPayloadType);
}
TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
TransportCallbackTestForComfortNoiseRegisterCngFirst) {
const int k10MsBlocksPerPacket = 3;
codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
const int kCngPayloadType = 105;
RegisterCngCodec(kCngPayloadType);
RegisterCodec();
ASSERT_EQ(0, acm_->SetVAD(true, true));
DoTest(k10MsBlocksPerPacket, kCngPayloadType);
}
// 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_(PlatformThread::CreateThread(CbSendThread, this, "send")),
insert_packet_thread_(PlatformThread::CreateThread(CbInsertPacketThread,
this,
"insert_packet")),
pull_audio_thread_(PlatformThread::CreateThread(CbPullAudioThread,
this,
"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();
RegisterCodec(); // Must be called before the threads start below.
StartThreads();
}
void StartThreads() {
ASSERT_TRUE(send_thread_->Start());
send_thread_->SetPriority(kRealtimePriority);
ASSERT_TRUE(insert_packet_thread_->Start());
insert_packet_thread_->SetPriority(kRealtimePriority);
ASSERT_TRUE(pull_audio_thread_->Start());
pull_audio_thread_->SetPriority(kRealtimePriority);
}
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_;
InsertAudioAndVerifyEncoding();
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;
}
rtc::scoped_ptr<PlatformThread> send_thread_;
rtc::scoped_ptr<PlatformThread> insert_packet_thread_;
rtc::scoped_ptr<PlatformThread> pull_audio_thread_;
const rtc::scoped_ptr<EventWrapper> test_complete_;
int send_count_;
int insert_packet_count_;
int pull_audio_count_ GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<SimulatedClock> fake_clock_;
};
TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_ON_IOS(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();
RegisterCodec(); // Must be called before the threads start below.
// 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();
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();
}
void RegisterCodec() override {
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
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() {
// TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
// this call confuses the number of samples with the number of bytes, and
// ends up copying only half of what it should.
memcpy(input_frame_.data_, audio_loop_.GetNextBlock().data(),
kNumSamples10ms);
AudioCodingModuleTestOldApi::InsertAudio();
}
// Override the verification function with no-op, since iSAC produces variable
// payload sizes.
void VerifyEncoding() override {}
// 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, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
EXPECT_EQ(kEventSignaled, RunTest());
}
class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
protected:
static const int kRegisterAfterNumPackets = 5;
static const int kNumPackets = 10;
static const int kPacketSizeMs = 30;
static const int kPacketSizeSamples = kPacketSizeMs * 16;
AcmReRegisterIsacMtTestOldApi()
: AudioCodingModuleTestOldApi(),
receive_thread_(
PlatformThread::CreateThread(CbReceiveThread, this, "receive")),
codec_registration_thread_(
PlatformThread::CreateThread(CbCodecRegistrationThread,
this,
"codec_registration")),
test_complete_(EventWrapper::Create()),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
codec_registered_(false),
receive_packet_count_(0),
next_insert_packet_time_ms_(0),
fake_clock_(new SimulatedClock(0)) {
AudioEncoderIsac::Config config;
config.payload_type = kPayloadType;
isac_encoder_.reset(new AudioEncoderIsac(config));
clock_ = fake_clock_.get();
}
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);
RegisterCodec(); // Must be called before the threads start below.
StartThreads();
}
void RegisterCodec() override {
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
// Only register the decoder for now. The encoder is registered later.
ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
}
void StartThreads() {
ASSERT_TRUE(receive_thread_->Start());
receive_thread_->SetPriority(kRealtimePriority);
ASSERT_TRUE(codec_registration_thread_->Start());
codec_registration_thread_->SetPriority(kRealtimePriority);
}
void TearDown() {
AudioCodingModuleTestOldApi::TearDown();
receive_thread_->Stop();
codec_registration_thread_->Stop();
}
EventTypeWrapper RunTest() {
return test_complete_->Wait(10 * 60 * 1000); // 10 minutes' timeout.
}
static bool CbReceiveThread(void* context) {
return reinterpret_cast<AcmReRegisterIsacMtTestOldApi*>(context)
->CbReceiveImpl();
}
bool CbReceiveImpl() {
SleepMs(1);
const size_t max_encoded_bytes = isac_encoder_->MaxEncodedBytes();
rtc::scoped_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);
AudioEncoder::EncodedInfo info;
{
CriticalSectionScoped lock(crit_sect_.get());
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return true;
}
next_insert_packet_time_ms_ += kPacketSizeMs;
++receive_packet_count_;
// Encode new frame.
uint32_t input_timestamp = rtp_header_.header.timestamp;
while (info.encoded_bytes == 0) {
info =
isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),
max_encoded_bytes, encoded.get());
input_timestamp += 160; // 10 ms at 16 kHz.
}
EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
input_timestamp);
EXPECT_EQ(rtp_header_.header.timestamp, info.encoded_timestamp);
EXPECT_EQ(rtp_header_.header.payloadType, info.payload_type);
}
// Now we're not holding the crit sect when calling ACM.
// Insert into ACM.
EXPECT_EQ(0, acm_->IncomingPacket(encoded.get(), info.encoded_bytes,
rtp_header_));
// Pull audio.
for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) {
AudioFrame audio_frame;
EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */,
&audio_frame));
fake_clock_->AdvanceTimeMilliseconds(10);
}
rtp_utility_->Forward(&rtp_header_);
return true;
}
static bool CbCodecRegistrationThread(void* context) {
return reinterpret_cast<AcmReRegisterIsacMtTestOldApi*>(context)
->CbCodecRegistrationImpl();
}
bool CbCodecRegistrationImpl() {
SleepMs(1);
if (HasFatalFailure()) {
// End the test early if a fatal failure (ASSERT_*) has occurred.
test_complete_->Set();
}
CriticalSectionScoped lock(crit_sect_.get());
if (!codec_registered_ &&
receive_packet_count_ > kRegisterAfterNumPackets) {
// Register the iSAC encoder.
EXPECT_EQ(0, acm_->RegisterSendCodec(codec_));
codec_registered_ = true;
}
if (codec_registered_ && receive_packet_count_ > kNumPackets) {
test_complete_->Set();
}
return true;
}
rtc::scoped_ptr<PlatformThread> receive_thread_;
rtc::scoped_ptr<PlatformThread> codec_registration_thread_;
const rtc::scoped_ptr<EventWrapper> test_complete_;
const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
bool codec_registered_ GUARDED_BY(crit_sect_);
int receive_packet_count_ GUARDED_BY(crit_sect_);
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<AudioEncoderIsac> isac_encoder_;
rtc::scoped_ptr<SimulatedClock> fake_clock_;
test::AudioLoop audio_loop_;
};
TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
EXPECT_EQ(kEventSignaled, RunTest());
}
// Disabling all of these tests on iOS until file support has been added.
// See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
#if !defined(WEBRTC_IOS)
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:
struct ExternalDecoder {
int rtp_payload_type;
AudioDecoder* external_decoder;
int sample_rate_hz;
int num_channels;
};
void Run(int output_freq_hz,
const std::string& checksum_ref,
const std::vector<ExternalDecoder>& external_decoders) {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
rtc::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());
for (const auto& ed : external_decoders) {
ASSERT_EQ(0, test.RegisterExternalReceiveCodec(
ed.rtp_payload_type, ed.external_decoder,
ed.sample_rate_hz, ed.num_channels));
}
test.Run();
std::string checksum_string = checksum.Finish();
EXPECT_EQ(checksum_ref, checksum_string);
}
};
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISAC)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
#define IF_ALL_CODECS(x) x
#else
#define IF_ALL_CODECS(x) DISABLED_##x
#endif
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_8kHzOutput DISABLED_8kHzOutput
#else
#define MAYBE_8kHzOutput 8kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_8kHzOutput)) {
Run(8000, PlatformChecksum("dcee98c623b147ebe1b40dd30efa896e",
"adc92e173f908f93b96ba5844209815a",
"908002dc01fc4eb1d2be24eb1d3f354b"),
std::vector<ExternalDecoder>());
}
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_16kHzOutput DISABLED_16kHzOutput
#else
#define MAYBE_16kHzOutput 16kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_16kHzOutput)) {
Run(16000, PlatformChecksum("f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
"8cffa6abcb3e18e33b9d857666dff66a",
"a909560b5ca49fa472b17b7b277195e9"),
std::vector<ExternalDecoder>());
}
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_32kHzOutput DISABLED_32kHzOutput
#else
#define MAYBE_32kHzOutput 32kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_32kHzOutput)) {
Run(32000, PlatformChecksum("306e0d990ee6e92de3fbecc0123ece37",
"3e126fe894720c3f85edadcc91964ba5",
"441aab4b347fb3db4e9244337aca8d8e"),
std::vector<ExternalDecoder>());
}
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_48kHzOutput DISABLED_48kHzOutput
#else
#define MAYBE_48kHzOutput 48kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_48kHzOutput)) {
Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
"0155665e93067c4e89256b944dd11999",
"4ee2730fa1daae755e8a8fd3abd779ec"),
std::vector<ExternalDecoder>());
}
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
#define MAYBE_48kHzOutputExternalDecoder DISABLED_48kHzOutputExternalDecoder
#else
#define MAYBE_48kHzOutputExternalDecoder 48kHzOutputExternalDecoder
#endif
TEST_F(AcmReceiverBitExactnessOldApi,
IF_ALL_CODECS(MAYBE_48kHzOutputExternalDecoder)) {
AudioDecoderPcmU decoder(1);
MockAudioDecoder mock_decoder;
// Set expectations on the mock decoder and also delegate the calls to the
// real decoder.
EXPECT_CALL(mock_decoder, IncomingPacket(_, _, _, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::IncomingPacket));
EXPECT_CALL(mock_decoder, Channels())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Channels));
EXPECT_CALL(mock_decoder, Decode(_, _, _, _, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Decode));
EXPECT_CALL(mock_decoder, HasDecodePlc())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc));
EXPECT_CALL(mock_decoder, PacketDuration(_, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::PacketDuration));
ExternalDecoder ed;
ed.rtp_payload_type = 0;
ed.external_decoder = &mock_decoder;
ed.sample_rate_hz = 8000;
ed.num_channels = 1;
std::vector<ExternalDecoder> external_decoders;
external_decoders.push_back(ed);
Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
"0155665e93067c4e89256b944dd11999",
"4ee2730fa1daae755e8a8fd3abd779ec"),
external_decoders);
EXPECT_CALL(mock_decoder, Die());
}
// 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);
}
bool RegisterExternalSendCodec(AudioEncoder* external_speech_encoder,
int payload_type) {
payload_type_ = payload_type;
frame_size_rtp_timestamps_ =
external_speech_encoder->Num10MsFramesInNextPacket() *
external_speech_encoder->RtpTimestampRateHz() / 100;
return send_test_->RegisterExternalCodec(external_speech_encoder);
}
// 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));
}
void SetUpTestExternalEncoder(AudioEncoder* external_speech_encoder,
int payload_type) {
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(
RegisterExternalSendCodec(external_speech_encoder, payload_type));
}
rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
rtc::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_;
};
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_IsacWb30ms DISABLED_IsacWb30ms
#else
#define MAYBE_IsacWb30ms IsacWb30ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(MAYBE_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);
}
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_IsacWb60ms DISABLED_IsacWb60ms
#else
#define MAYBE_IsacWb60ms IsacWb60ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(MAYBE_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);
}
#ifdef WEBRTC_CODEC_ISAC
#define IF_ISAC_FLOAT(x) x
#else
#define IF_ISAC_FLOAT(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactnessOldApi,
DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"2b3c387d06f00b7b7aad4c9be56fb83d",
"",
"5683b58da0fbf2063c7adc2e6bfb3fb8"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"bcc2041e7744c7ebd9f701866856849c",
"",
"ce86106a93419aefb063097108ec94ab"),
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);
}
#ifdef WEBRTC_CODEC_ILBC
#define IF_ILBC(x) x
#else
#define IF_ILBC(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(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);
}
#ifdef WEBRTC_CODEC_G722
#define IF_G722(x) x
#else
#define IF_G722(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(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(IF_G722(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);
}
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
#else
#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_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);
}
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_Opus_stereo_20ms_voip DISABLED_Opus_stereo_20ms_voip
#else
#define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
// If not set, default will be kAudio in case of stereo.
EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"9b9e12bc3cc793740966e11cbfa8b35b",
"57412a4b5771d19ff03ec35deffe7067",
"9b9e12bc3cc793740966e11cbfa8b35b"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"c7340b1189652ab6b5e80dade7390cb4",
"cdfe85939c411d12b61701c566e22d26",
"c7340b1189652ab6b5e80dade7390cb4"),
50,
test::AcmReceiveTestOldApi::kStereoOutput);
}
// This test is for verifying the SetBitRate function. The bitrate is changed at
// the beginning, and the number of generated bytes are checked.
class AcmSetBitRateOldApi : public ::testing::Test {
protected:
static const int kTestDurationMs = 1000;
// 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();
}
// Registers a send codec in the test::AcmSendTest object. Returns true on
// success, false on failure.
virtual bool RegisterSendCodec(const char* payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples,
int 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(int target_bitrate_bps, int expected_total_bits) {
ASSERT_TRUE(send_test_->acm());
send_test_->acm()->SetBitRate(target_bitrate_bps);
int nr_bytes = 0;
while (test::Packet* next_packet = send_test_->NextPacket()) {
nr_bytes += next_packet->payload_length_bytes();
delete next_packet;
}
EXPECT_EQ(expected_total_bits, nr_bytes * 8);
}
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));
}
rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
rtc::scoped_ptr<test::InputAudioFile> audio_source_;
};
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 9328);
#else
Run(10000, 9072);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 47952);
#else
Run(50000, 49600);
#endif // WEBRTC_ANDROID
}
// The result on the Android platforms is inconsistent for this test case.
// On android_rel the result is different from android and android arm64 rel.
TEST_F(AcmSetBitRateOldApi, DISABLED_ON_ANDROID(Opus_48khz_20ms_100kbps)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
Run(100000, 100888);
}
// These next 2 tests ensure that the SetBitRate function has no effect on PCM
TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
Run(8000, 128000);
}
TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
Run(32000, 128000);
}
// This test is for verifying the SetBitRate function. The bitrate is changed
// in the middle, and the number of generated bytes are before and after the
// change are checked.
class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
protected:
AcmChangeBitRateOldApi() : sampling_freq_hz_(0), frame_size_samples_(0) {}
// 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) override {
frame_size_samples_ = frame_size_samples;
sampling_freq_hz_ = sampling_freq_hz;
return AcmSetBitRateOldApi::RegisterSendCodec(
payload_name, sampling_freq_hz, channels, payload_type,
frame_size_samples, frame_size_rtp_timestamps);
}
// Runs the test. SetUpSender() and RegisterSendCodec() must have been called
// before calling this method.
void Run(int target_bitrate_bps,
int expected_before_switch_bits,
int expected_after_switch_bits) {
ASSERT_TRUE(send_test_->acm());
int nr_packets =
sampling_freq_hz_ * kTestDurationMs / (frame_size_samples_ * 1000);
int nr_bytes_before = 0, nr_bytes_after = 0;
int packet_counter = 0;
while (test::Packet* next_packet = send_test_->NextPacket()) {
if (packet_counter == nr_packets / 2)
send_test_->acm()->SetBitRate(target_bitrate_bps);
if (packet_counter < nr_packets / 2)
nr_bytes_before += next_packet->payload_length_bytes();
else
nr_bytes_after += next_packet->payload_length_bytes();
packet_counter++;
delete next_packet;
}
EXPECT_EQ(expected_before_switch_bits, nr_bytes_before * 8);
EXPECT_EQ(expected_after_switch_bits, nr_bytes_after * 8);
}
uint32_t sampling_freq_hz_;
uint32_t frame_size_samples_;
};
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 32200, 5496);
#else
Run(10000, 32200, 5432);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 32200, 24912);
#else
Run(50000, 32200, 24792);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(100000, 32200, 51480);
#else
Run(100000, 32200, 50584);
#endif // WEBRTC_ANDROID
}
// These next 2 tests ensure that the SetBitRate function has no effect on PCM
TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
Run(8000, 64000, 64000);
}
TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
Run(32000, 64000, 64000);
}
TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
CodecInst codec_inst;
codec_inst.channels = 1;
codec_inst.pacsize = 160;
codec_inst.pltype = 0;
AudioEncoderPcmU encoder(codec_inst);
MockAudioEncoder mock_encoder;
// Set expectations on the mock encoder and also delegate the calls to the
// real encoder.
EXPECT_CALL(mock_encoder, MaxEncodedBytes())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::MaxEncodedBytes));
EXPECT_CALL(mock_encoder, SampleRateHz())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));
EXPECT_CALL(mock_encoder, NumChannels())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::NumChannels));
EXPECT_CALL(mock_encoder, RtpTimestampRateHz())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz));
EXPECT_CALL(mock_encoder, Num10MsFramesInNextPacket())
.Times(AtLeast(1))
.WillRepeatedly(
Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket));
EXPECT_CALL(mock_encoder, Max10MsFramesInAPacket())
.Times(AtLeast(1))
.WillRepeatedly(
Invoke(&encoder, &AudioEncoderPcmU::Max10MsFramesInAPacket));
EXPECT_CALL(mock_encoder, GetTargetBitrate())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::EncodeInternal));
EXPECT_CALL(mock_encoder, SetFec(_))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SetFec));
ASSERT_NO_FATAL_FAILURE(
SetUpTestExternalEncoder(&mock_encoder, codec_inst.pltype));
Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
// This test fixture is implemented to run ACM and change the desired output
// frequency during the call. The input packets are simply PCM16b-wb encoded
// payloads with a constant value of |kSampleValue|. The test fixture itself
// acts as PacketSource in between the receive test class and the constant-
// payload packet source class. The output is both written to file, and analyzed
// in this test fixture.
class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test,
public test::PacketSource,
public test::AudioSink {
protected:
static const size_t kTestNumPackets = 50;
static const int kEncodedSampleRateHz = 16000;
static const size_t kPayloadLenSamples = 30 * kEncodedSampleRateHz / 1000;
static const int kPayloadType = 108; // Default payload type for PCM16b-wb.
AcmSwitchingOutputFrequencyOldApi()
: first_output_(true),
num_packets_(0),
packet_source_(kPayloadLenSamples,
kSampleValue,
kEncodedSampleRateHz,
kPayloadType),
output_freq_2_(0),
has_toggled_(false) {}
void Run(int output_freq_1, int output_freq_2, int toggle_period_ms) {
// Set up the receiver used to decode the packets and verify the decoded
// output.
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 WriteArray method in
// this class.
test::AudioSinkFork output(this, &output_file);
test::AcmReceiveTestToggleOutputFreqOldApi receive_test(
this,
&output,
output_freq_1,
output_freq_2,
toggle_period_ms,
test::AcmReceiveTestOldApi::kMonoOutput);
ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
output_freq_2_ = output_freq_2;
// This is where the actual test is executed.
receive_test.Run();
}
// Inherited from test::PacketSource.
test::Packet* NextPacket() override {
// Check if it is time to terminate the test. The packet source is of type
// ConstantPcmPacketSource, which is infinite, so we must end the test
// "manually".
if (num_packets_++ > kTestNumPackets) {
EXPECT_TRUE(has_toggled_);
return NULL; // Test ended.
}
// Get the next packet from the source.
return packet_source_.NextPacket();
}
// Inherited from test::AudioSink.
bool WriteArray(const int16_t* audio, size_t num_samples) {
// Skip checking the first output frame, since it has a number of zeros
// due to how NetEq is initialized.
if (first_output_) {
first_output_ = false;
return true;
}
for (size_t i = 0; i < num_samples; ++i) {
EXPECT_EQ(kSampleValue, audio[i]);
}
if (num_samples ==
static_cast<size_t>(output_freq_2_ / 100)) // Size of 10 ms frame.
has_toggled_ = true;
// The return value does not say if the values match the expectation, just
// that the method could process the samples.
return true;
}
const int16_t kSampleValue = 1000;
bool first_output_;
size_t num_packets_;
test::ConstantPcmPacketSource packet_source_;
int output_freq_2_;
bool has_toggled_;
};
TEST_F(AcmSwitchingOutputFrequencyOldApi, TestWithoutToggling) {
Run(16000, 16000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo32Khz) {
Run(16000, 32000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle32KhzTo16Khz) {
Run(32000, 16000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo8Khz) {
Run(16000, 8000, 1000);
}
TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
Run(8000, 16000, 1000);
}
#endif
} // namespace webrtc