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webrtc / src.git / aaa882cea5dc42532e44d8ed9b1eaeb064a249a1 / . / modules / audio_coding / acm2 / audio_coding_module_unittest.cc
blob: aaa4230d1ed395e8a24433ca4b41a4eed0f44c04 [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 <stdio.h>
#include <string.h>
#include <memory>
#include <vector>
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/acm2/acm_receive_test.h"
#include "modules/audio_coding/acm2/acm_send_test.h"
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/messagedigest.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/refcountedobject.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/event_wrapper.h"
#include "system_wrappers/include/sleep.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder.h"
#include "test/mock_audio_encoder.h"
#include "test/testsupport/fileutils.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) {}
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 {
rtc::CritScope lock(&crit_sect_);
++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 {
rtc::CritScope lock(&crit_sect_);
return num_calls_;
}
int last_payload_len_bytes() const {
rtc::CritScope lock(&crit_sect_);
return rtc::checked_cast<int>(last_payload_vec_.size());
}
FrameType last_frame_type() const {
rtc::CritScope lock(&crit_sect_);
return last_frame_type_;
}
int last_payload_type() const {
rtc::CritScope lock(&crit_sect_);
return last_payload_type_;
}
uint32_t last_timestamp() const {
rtc::CritScope lock(&crit_sect_);
return last_timestamp_;
}
void SwapBuffers(std::vector<uint8_t>* payload) {
rtc::CritScope lock(&crit_sect_);
last_payload_vec_.swap(*payload);
}
private:
int num_calls_ RTC_GUARDED_BY(crit_sect_);
FrameType last_frame_type_ RTC_GUARDED_BY(crit_sect_);
int last_payload_type_ RTC_GUARDED_BY(crit_sect_);
uint32_t last_timestamp_ RTC_GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(crit_sect_);
rtc::CriticalSection crit_sect_;
};
class AudioCodingModuleTestOldApi : public ::testing::Test {
protected:
AudioCodingModuleTestOldApi()
: rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
clock_(Clock::GetRealTimeClock()) {}
~AudioCodingModuleTestOldApi() {}
void TearDown() {}
void SetUp() {
acm_.reset(AudioCodingModule::Create(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");
input_frame_.Mute();
ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_));
SetUpL16Codec();
}
// Set up L16 codec.
virtual void SetUpL16Codec() {
audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1));
codec_.pltype = kPayloadType;
}
virtual void RegisterCodec() {
EXPECT_EQ(true, acm_->RegisterReceiveCodec(kPayloadType, *audio_format_));
EXPECT_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;
bool muted;
ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted));
ASSERT_FALSE(muted);
}
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();
}
std::unique_ptr<RtpUtility> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
// These two have to be kept in sync for now. In the future, we'll be able to
// eliminate the CodecInst and keep only the SdpAudioFormat.
rtc::Optional<SdpAudioFormat> audio_format_;
CodecInst codec_;
Clock* clock_;
};
// Check if the statistics are initialized correctly. Before any call to ACM
// all fields have to be zero.
#if defined(WEBRTC_ANDROID)
#define MAYBE_InitializedToZero DISABLED_InitializedToZero
#else
#define MAYBE_InitializedToZero InitializedToZero
#endif
TEST_F(AudioCodingModuleTestOldApi, MAYBE_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);
EXPECT_EQ(0, stats.decoded_muted_output);
}
// 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.
#if defined(WEBRTC_ANDROID)
#define MAYBE_NetEqCalls DISABLED_NetEqCalls
#else
#define MAYBE_NetEqCalls NetEqCalls
#endif
TEST_F(AudioCodingModuleTestOldApi, MAYBE_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);
EXPECT_EQ(0, stats.decoded_muted_output);
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);
EXPECT_EQ(0, stats.decoded_muted_output);
// TODO(henrik.lundin) Add a test with muted state enabled.
}
TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
AudioFrame audio_frame;
const int kSampleRateHz = 32000;
bool muted;
EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
ASSERT_FALSE(muted);
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0u);
EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
audio_frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
}
// The below test is temporarily disabled on Windows due to problems
// with clang debug builds.
// TODO(tommi): Re-enable when we've figured out what the problem is.
// http://crbug.com/615050
#if !defined(WEBRTC_WIN) && defined(__clang__) && RTC_DCHECK_IS_ON && \
GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
AudioFrame audio_frame;
bool muted;
EXPECT_DEATH(acm_->PlayoutData10Ms(0, &audio_frame, &muted),
"dst_sample_rate_hz");
}
#endif
// 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)
// Verifies that the RTP timestamp series is not reset when the codec is
// changed.
TEST_F(AudioCodingModuleTestOldApi, 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;
}
}
#endif
// 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) {
EXPECT_EQ(true,
acm_->RegisterReceiveCodec(
rtp_payload_type, SdpAudioFormat("cn", kSampleRateHz, 1)));
CodecInst codec;
EXPECT_EQ(0, AudioCodingModule::Codec("CN", &codec, kSampleRateHz, 1));
codec.pltype = rtp_payload_type;
EXPECT_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_(CbSendThread, this, "send"),
insert_packet_thread_(CbInsertPacketThread, this, "insert_packet"),
pull_audio_thread_(CbPullAudioThread, this, "pull_audio"),
test_complete_(EventWrapper::Create()),
send_count_(0),
insert_packet_count_(0),
pull_audio_count_(0),
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() {
send_thread_.Start();
send_thread_.SetPriority(rtc::kRealtimePriority);
insert_packet_thread_.Start();
insert_packet_thread_.SetPriority(rtc::kRealtimePriority);
pull_audio_thread_.Start();
pull_audio_thread_.SetPriority(rtc::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) {
rtc::CritScope lock(&crit_sect_);
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);
{
rtc::CritScope lock(&crit_sect_);
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);
{
rtc::CritScope lock(&crit_sect_);
// 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::PlatformThread send_thread_;
rtc::PlatformThread insert_packet_thread_;
rtc::PlatformThread pull_audio_thread_;
const std::unique_ptr<EventWrapper> test_complete_;
int send_count_;
int insert_packet_count_;
int pull_audio_count_ RTC_GUARDED_BY(crit_sect_);
rtc::CriticalSection crit_sect_;
int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<SimulatedClock> fake_clock_;
};
#if defined(WEBRTC_IOS)
#define MAYBE_DoTest DISABLED_DoTest
#else
#define MAYBE_DoTest DoTest
#endif
TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_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() override {
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");
audio_format_ = SdpAudioFormat("isac", kSampleRateHz, 1);
AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
EXPECT_EQ(true, acm_->RegisterReceiveCodec(kPayloadType, *audio_format_));
EXPECT_EQ(0, acm_->RegisterSendCodec(codec_));
}
void InsertPacket() override {
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() override {
// 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_.mutable_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).
bool TestDone() override {
if (packet_cb_.num_calls() > kNumPackets) {
rtc::CritScope lock(&crit_sect_);
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_;
};
#if defined(WEBRTC_IOS)
#define MAYBE_DoTest DISABLED_DoTest
#else
#define MAYBE_DoTest DoTest
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) {
EXPECT_EQ(kEventSignaled, RunTest());
}
#endif
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_(CbReceiveThread, this, "receive"),
codec_registration_thread_(CbCodecRegistrationThread,
this,
"codec_registration"),
test_complete_(EventWrapper::Create()),
codec_registered_(false),
receive_packet_count_(0),
next_insert_packet_time_ms_(0),
fake_clock_(new SimulatedClock(0)) {
AudioEncoderIsacFloatImpl::Config config;
config.payload_type = kPayloadType;
isac_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
clock_ = fake_clock_.get();
}
void SetUp() override {
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(true, acm_->RegisterReceiveCodec(codec_.pltype,
CodecInstToSdp(codec_)));
}
void StartThreads() {
receive_thread_.Start();
receive_thread_.SetPriority(rtc::kRealtimePriority);
codec_registration_thread_.Start();
codec_registration_thread_.SetPriority(rtc::kRealtimePriority);
}
void TearDown() override {
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);
rtc::Buffer encoded;
AudioEncoder::EncodedInfo info;
{
rtc::CritScope lock(&crit_sect_);
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(),
&encoded);
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.data(), info.encoded_bytes,
rtp_header_));
// Pull audio.
for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) {
AudioFrame audio_frame;
bool muted;
EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */,
&audio_frame, &muted));
if (muted) {
ADD_FAILURE();
return false;
}
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();
}
rtc::CritScope lock(&crit_sect_);
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::PlatformThread receive_thread_;
rtc::PlatformThread codec_registration_thread_;
const std::unique_ptr<EventWrapper> test_complete_;
rtc::CriticalSection crit_sect_;
bool codec_registered_ RTC_GUARDED_BY(crit_sect_);
int receive_packet_count_ RTC_GUARDED_BY(crit_sect_);
int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<AudioEncoderIsacFloatImpl> isac_encoder_;
std::unique_ptr<SimulatedClock> fake_clock_;
test::AudioLoop audio_loop_;
};
#if defined(WEBRTC_IOS)
#define MAYBE_DoTest DISABLED_DoTest
#else
#define MAYBE_DoTest DoTest
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) {
EXPECT_EQ(kEventSignaled, RunTest());
}
#endif
// 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 others,
std::string win64,
std::string android_arm32,
std::string android_arm64,
std::string android_arm64_clang) {
#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
return win64;
#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
return android_arm32;
#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#if defined(__clang__)
// Android ARM64 with Clang compiler
return android_arm64_clang;
#else
// Android ARM64 with non-Clang compiler
return android_arm64;
#endif // __clang__
#else
return others;
#endif
}
protected:
struct ExternalDecoder {
int rtp_payload_type;
AudioDecoder* external_decoder;
int sample_rate_hz;
int num_channels;
std::string name;
};
void Run(int output_freq_hz, const std::string& checksum_ref) {
Run(output_freq_hz, checksum_ref, CreateBuiltinAudioDecoderFactory(),
[](AudioCodingModule*) {});
}
void Run(int output_freq_hz,
const std::string& checksum_ref,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
rtc::FunctionView<void(AudioCodingModule*)> decoder_reg) {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
std::unique_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.wav";
test::OutputWavFile output_file(output_file_name, output_freq_hz);
test::AudioSinkFork output(&checksum, &output_file);
test::AcmReceiveTestOldApi test(
packet_source.get(), &output, output_freq_hz,
test::AcmReceiveTestOldApi::kArbitraryChannels,
std::move(decoder_factory));
ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs());
decoder_reg(test.get_acm());
test.Run();
std::string checksum_string = checksum.Finish();
EXPECT_EQ(checksum_ref, checksum_string);
// Delete the output file.
remove(output_file_name.c_str());
}
};
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC)
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
Run(8000, PlatformChecksum("2adede965c6f87de7142c51552111d08",
"028c0fc414b1c9ab7e582dccdf381e98",
"36c95170c1393d4b765d1c17b61ef977",
"4598140b5e4f7ee66c5adad609e65a3e",
"bac5db6dff44323be401060f1279a532"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
Run(16000, PlatformChecksum("c2550a3db7632de409e8db0093df1c12",
"edd31f4b6665cd5b9041fb93f2316594",
"22128bca51650cb61c80bed63b595603",
"f2aad418af974a3b1694d5ae5cc2c3c7",
"61c3cb9386b9503feebcb829c9be54bd"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
Run(32000, PlatformChecksum("85e28d7950132d56f90b099c90f82153",
"7b903f5c89997f271b405e63c245ef45",
"8b8fc6c6fd1dcdcfb3dd90e1ce597f10",
"100869c8dcde51346c2073e52a272d98",
"fdec5301dc649a47d407382b587e14da"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
"d8609bc9b495d81f29779344c68bcc47",
"ec5ebb90cda0ea5bb89e79d698af65de",
"bd44bf97e7899186532f91235cef444d",
"0baae2972cca142027d4af44f95f0bd5"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
class ADFactory : public AudioDecoderFactory {
public:
ADFactory()
: mock_decoder_(new MockAudioDecoder()),
pcmu_decoder_(1),
decode_forwarder_(&pcmu_decoder_),
fact_(CreateBuiltinAudioDecoderFactory()) {
// 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(&pcmu_decoder_, &AudioDecoderPcmU::IncomingPacket));
EXPECT_CALL(*mock_decoder_, SampleRateHz())
.Times(AtLeast(1))
.WillRepeatedly(
Invoke(&pcmu_decoder_, &AudioDecoderPcmU::SampleRateHz));
EXPECT_CALL(*mock_decoder_, Channels())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&pcmu_decoder_, &AudioDecoderPcmU::Channels));
EXPECT_CALL(*mock_decoder_, DecodeInternal(_, _, _, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&decode_forwarder_, &DecodeForwarder::Decode));
EXPECT_CALL(*mock_decoder_, HasDecodePlc())
.Times(AtLeast(1))
.WillRepeatedly(
Invoke(&pcmu_decoder_, &AudioDecoderPcmU::HasDecodePlc));
EXPECT_CALL(*mock_decoder_, PacketDuration(_, _))
.Times(AtLeast(1))
.WillRepeatedly(
Invoke(&pcmu_decoder_, &AudioDecoderPcmU::PacketDuration));
EXPECT_CALL(*mock_decoder_, Die());
}
std::vector<AudioCodecSpec> GetSupportedDecoders() override {
return fact_->GetSupportedDecoders();
}
bool IsSupportedDecoder(const SdpAudioFormat& format) override {
return format.name == "MockPCMu" ? true
: fact_->IsSupportedDecoder(format);
}
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
rtc::Optional<AudioCodecPairId> codec_pair_id) override {
return format.name == "MockPCMu"
? std::move(mock_decoder_)
: fact_->MakeAudioDecoder(format, codec_pair_id);
}
private:
// Class intended to forward a call from a mock DecodeInternal to Decode on
// the real decoder's Decode. DecodeInternal for the real decoder isn't
// public.
class DecodeForwarder {
public:
explicit DecodeForwarder(AudioDecoder* decoder) : decoder_(decoder) {}
int Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
AudioDecoder::SpeechType* speech_type) {
return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
decoder_->PacketDuration(encoded, encoded_len) *
decoder_->Channels() * sizeof(int16_t),
decoded, speech_type);
}
private:
AudioDecoder* const decoder_;
};
std::unique_ptr<MockAudioDecoder> mock_decoder_;
AudioDecoderPcmU pcmu_decoder_;
DecodeForwarder decode_forwarder_;
rtc::scoped_refptr<AudioDecoderFactory> fact_; // Fallback factory.
};
rtc::scoped_refptr<rtc::RefCountedObject<ADFactory>> factory(
new rtc::RefCountedObject<ADFactory>);
Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
"d8609bc9b495d81f29779344c68bcc47",
"ec5ebb90cda0ea5bb89e79d698af65de",
"bd44bf97e7899186532f91235cef444d",
"0baae2972cca142027d4af44f95f0bd5"),
factory, [](AudioCodingModule* acm) {
acm->RegisterReceiveCodec(0, {"MockPCMu", 8000, 1});
});
}
#endif
// 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),
payload_checksum_(rtc::MessageDigestFactory::Create(rtc::DIGEST_MD5)) {}
// 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_ = rtc::checked_cast<uint32_t>(
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.wav";
const int kOutputFreqHz = 8000;
test::OutputWavFile output_file(output_file_name, kOutputFreqHz);
// Have the output audio sent both to file and to the checksum calculator.
test::AudioSinkFork output(&audio_checksum, &output_file);
test::AcmReceiveTestOldApi receive_test(this, &output, kOutputFreqHz,
expected_channels,
CreateBuiltinAudioDecoderFactory());
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.
rtc::Buffer checksum_result(payload_checksum_->Size());
payload_checksum_->Finish(checksum_result.data(), checksum_result.size());
checksum_string =
rtc::hex_encode(checksum_result.data<char>(), checksum_result.size());
EXPECT_EQ(payload_checksum_ref, checksum_string);
// Verify number of packets produced.
EXPECT_EQ(expected_packets, packet_count_);
// Delete the output file.
remove(output_file_name.c_str());
}
// Inherited from test::PacketSource.
std::unique_ptr<test::Packet> NextPacket() override {
auto packet = send_test_->NextPacket();
if (!packet)
return NULL;
VerifyPacket(packet.get());
// 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));
}
std::unique_ptr<test::AcmSendTestOldApi> send_test_;
std::unique_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_;
std::unique_ptr<rtc::MessageDigest> payload_checksum_;
};
class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {};
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"2c9cb15d4ed55b5a0cadd04883bc73b0",
"9336a9b993cbd8a751f0e8958e66c89c",
"bd4682225f7c4ad5f2049f6769713ac2",
"343f1f42be0607c61e6516aece424609",
"2c9cb15d4ed55b5a0cadd04883bc73b0"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"3c79f16f34218271f3dca4e2b1dfe1bb",
"d42cb5195463da26c8129bbfe73a22e6",
"83de248aea9c3c2bd680b6952401b4ca",
"3c79f16f34218271f3dca4e2b1dfe1bb",
"3c79f16f34218271f3dca4e2b1dfe1bb"),
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"1ad29139a04782a33daad8c2b9b35875",
"14d63c5f08127d280e722e3191b73bdd",
"edcf26694c289e3d9691faf79b74f09f",
"ef75e900e6f375e3061163c53fd09a63",
"1ad29139a04782a33daad8c2b9b35875"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"9e0a0ab743ad987b55b8e14802769c56",
"ebe04a819d3a9d83a83a17f271e1139a",
"97aeef98553b5a4b5a68f8b716e8eaf0",
"9e0a0ab743ad987b55b8e14802769c56",
"9e0a0ab743ad987b55b8e14802769c56"),
16, test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
#if defined(WEBRTC_ANDROID)
#define MAYBE_IsacSwb30ms DISABLED_IsacSwb30ms
#else
#define MAYBE_IsacSwb30ms IsacSwb30ms
#endif
#if defined(WEBRTC_CODEC_ISAC)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacSwb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"5683b58da0fbf2063c7adc2e6bfb3fb8",
"2b3c387d06f00b7b7aad4c9be56fb83d", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"ce86106a93419aefb063097108ec94ab",
"bcc2041e7744c7ebd9f701866856849c", "android_arm32_payload",
"android_arm64_payload", "android_arm64_clang_payload"),
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
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);
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_Ilbc_30ms DISABLED_Ilbc_30ms
#else
#define MAYBE_Ilbc_30ms Ilbc_30ms
#endif
#if defined(WEBRTC_CODEC_ILBC)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7b6ec10910debd9af08011d3ed5249f7",
"7b6ec10910debd9af08011d3ed5249f7", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"cfae2e9f6aba96e145f2bcdd5050ce78",
"cfae2e9f6aba96e145f2bcdd5050ce78", "android_arm32_payload",
"android_arm64_payload", "android_arm64_clang_payload"),
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
#if defined(WEBRTC_ANDROID)
#define MAYBE_G722_20ms DISABLED_G722_20ms
#else
#define MAYBE_G722_20ms G722_20ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"e99c89be49a46325d03c0d990c292d68",
"e99c89be49a46325d03c0d990c292d68", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"fc68a87e1380614e658087cb35d5ca10",
"fc68a87e1380614e658087cb35d5ca10", "android_arm32_payload",
"android_arm64_payload", "android_arm64_clang_payload"),
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms
#else
#define MAYBE_G722_stereo_20ms G722_stereo_20ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"e280aed283e499d37091b481ca094807",
"e280aed283e499d37091b481ca094807", "android_arm32_audio",
"android_arm64_audio", "android_arm64_clang_audio"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"66516152eeaa1e650ad94ff85f668dac",
"66516152eeaa1e650ad94ff85f668dac", "android_arm32_payload",
"android_arm64_payload", "android_arm64_clang_payload"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"3e285b74510e62062fbd8142dacd16e9",
"3e285b74510e62062fbd8142dacd16e9",
"439e97ad1932c49923b5da029c17dd5e",
"038ec90f5f3fc2320f3090f8ecef6bb7",
"038ec90f5f3fc2320f3090f8ecef6bb7"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"78cf8f03157358acdc69f6835caa0d9b",
"78cf8f03157358acdc69f6835caa0d9b",
"ab88b1a049c36bdfeb7e8b057ef6982a",
"27fef7b799393347ec3b5694369a1c36",
"27fef7b799393347ec3b5694369a1c36"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 120);
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(encoder.get(), 120));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"3e285b74510e62062fbd8142dacd16e9",
"3e285b74510e62062fbd8142dacd16e9",
"439e97ad1932c49923b5da029c17dd5e",
"038ec90f5f3fc2320f3090f8ecef6bb7",
"038ec90f5f3fc2320f3090f8ecef6bb7"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"78cf8f03157358acdc69f6835caa0d9b",
"78cf8f03157358acdc69f6835caa0d9b",
"ab88b1a049c36bdfeb7e8b057ef6982a",
"27fef7b799393347ec3b5694369a1c36",
"27fef7b799393347ec3b5694369a1c36"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, 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(
"b0325df4e8104f04e03af23c0b75800e",
"b0325df4e8104f04e03af23c0b75800e",
"1c81121f5d9286a5a865d01dbab22ce8",
"11d547f89142e9ef03f37d7ca7f32379",
"11d547f89142e9ef03f37d7ca7f32379"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"4eab2259b6fe24c22dd242a113e0b3d9",
"4eab2259b6fe24c22dd242a113e0b3d9",
"839ea60399447268ee0f0262a50b75fd",
"1815fd5589cad0c6f6cf946c76b81aeb",
"1815fd5589cad0c6f6cf946c76b81aeb"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 120);
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(encoder.get(), 120));
// If not set, default will be kAudio in case of stereo.
EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"b0325df4e8104f04e03af23c0b75800e",
"b0325df4e8104f04e03af23c0b75800e",
"1c81121f5d9286a5a865d01dbab22ce8",
"11d547f89142e9ef03f37d7ca7f32379",
"11d547f89142e9ef03f37d7ca7f32379"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"4eab2259b6fe24c22dd242a113e0b3d9",
"4eab2259b6fe24c22dd242a113e0b3d9",
"839ea60399447268ee0f0262a50b75fd",
"1815fd5589cad0c6f6cf946c76b81aeb",
"1815fd5589cad0c6f6cf946c76b81aeb"),
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 AcmSetBitRateTest : 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);
}
bool RegisterExternalSendCodec(AudioEncoder* external_speech_encoder,
int payload_type) {
return send_test_->RegisterExternalCodec(external_speech_encoder);
}
void RunInner(int expected_total_bits) {
int nr_bytes = 0;
while (std::unique_ptr<test::Packet> next_packet =
send_test_->NextPacket()) {
nr_bytes += rtc::checked_cast<int>(next_packet->payload_length_bytes());
}
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));
}
std::unique_ptr<test::AcmSendTestOldApi> send_test_;
std::unique_ptr<test::InputAudioFile> audio_source_;
};
class AcmSetBitRateOldApi : public AcmSetBitRateTest {
protected:
// Runs the test. SetUpSender() must have been called and a codec must be set
// up 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);
RunInner(expected_total_bits);
}
};
class AcmSetBitRateNewApi : public AcmSetBitRateTest {
protected:
// Runs the test. SetUpSender() must have been called and a codec must be set
// up before calling this method.
void Run(int expected_total_bits) { RunInner(expected_total_bits); }
};
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 8640);
#else
Run(10000, 8680);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
#if defined(WEBRTC_ANDROID)
RunInner(8640);
#else
RunInner(8680);
#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, 45792);
#else
Run(50000, 45520);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
#if defined(WEBRTC_ANDROID)
RunInner(45792);
#else
RunInner(45520);
#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.
#if defined(WEBRTC_ANDROID)
#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
DISABLED_OpusFromFormat_48khz_20ms_100kbps
#else
#define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
OpusFromFormat_48khz_20ms_100kbps
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
Run(100000, 100832);
}
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}}));
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
ASSERT_TRUE(SetUpSender());
ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
RunInner(100832);
}
// 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 (std::unique_ptr<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 += rtc::checked_cast<int>(
next_packet->payload_length_bytes());
else
nr_bytes_after += rtc::checked_cast<int>(
next_packet->payload_length_bytes());
packet_counter++;
}
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_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 29512, 4800);
#else
Run(10000, 32200, 5368);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 29512, 23304);
#else
Run(50000, 32200, 23920);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
#if defined(WEBRTC_ARCH_ARM64)
Run(100000, 29512, 50440);
#else
Run(100000, 29512, 50496);
#endif // WEBRTC_ARCH_ARM64
#else
Run(100000, 32200, 50448);
#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, 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, GetTargetBitrate())
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
EXPECT_CALL(mock_encoder, EncodeImpl(_, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke(&encoder,
static_cast<
AudioEncoder::EncodedInfo(AudioEncoder::*)(
uint32_t,
rtc::ArrayView<const int16_t>,
rtc::Buffer*)>(&AudioEncoderPcmU::Encode)));
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();
// Delete output file.
remove(output_file_name.c_str());
}
// Inherited from test::PacketSource.
std::unique_ptr<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) override {
// 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