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Revert "[ACM] iSAC audio codec removed"

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This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1.

Reason for revert: breaks a downstream project

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
This commit is contained in:
Alessio Bazzica
2022-11-16 19:13:25 +00:00
committed by WebRTC LUCI CQ
parent cb2b133bf0
commit fbeb76ab51
164 changed files with 39429 additions and 117 deletions
Download Patch File Download Diff File Expand all files Collapse all files

340
modules/audio_coding/acm2/audio_coding_module_unittest.cc
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@ -30,6 +30,7 @@
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.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_typedefs.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
@ -301,6 +302,44 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
EXPECT_EQ(AudioFrameType::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 = pac_size_ / (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 += pac_size_;
}
// Change codec.
audio_format_ = SdpAudioFormat("ISAC", kSampleRateHz, 1);
pac_size_ = 480;
RegisterCodec();
blocks_per_packet = pac_size_ / (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 += pac_size_;
}
}
#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
@ -381,7 +420,8 @@ TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
DoTest(k10MsBlocksPerPacket, kCngPayloadType);
}
// A multi-threaded test for ACM that uses the PCM16b 16 kHz codec.
// 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;
@ -520,6 +560,272 @@ TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
EXPECT_TRUE(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);
pac_size_ = 480;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
}
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) {
MutexLock lock(&mutex_);
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_TRUE(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(),
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 {
// 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.
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
acm_->SetReceiveCodecs({{kPayloadType, {"ISAC", kSampleRateHz, 1}}});
}
void StartThreads() {
quit_.store(false);
const auto attributes =
rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
receive_thread_ = rtc::PlatformThread::SpawnJoinable(
[this] {
while (!quit_.load() && CbReceiveImpl()) {
}
},
"receive", attributes);
codec_registration_thread_ = rtc::PlatformThread::SpawnJoinable(
[this] {
while (!quit_.load()) {
CbCodecRegistrationImpl();
}
},
"codec_registration", attributes);
}
void TearDown() override {
AudioCodingModuleTestOldApi::TearDown();
quit_.store(true);
receive_thread_.Finalize();
codec_registration_thread_.Finalize();
}
bool RunTest() { return test_complete_.Wait(TimeDelta::Minutes(10)); }
bool CbReceiveImpl() {
SleepMs(1);
rtc::Buffer encoded;
AudioEncoder::EncodedInfo info;
{
MutexLock lock(&mutex_);
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_.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_.timestamp + kPacketSizeSamples, input_timestamp);
EXPECT_EQ(rtp_header_.timestamp, info.encoded_timestamp);
EXPECT_EQ(rtp_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;
}
void CbCodecRegistrationImpl() {
SleepMs(1);
if (HasFatalFailure()) {
// End the test early if a fatal failure (ASSERT_*) has occurred.
test_complete_.Set();
}
MutexLock lock(&mutex_);
if (!codec_registered_ &&
receive_packet_count_ > kRegisterAfterNumPackets) {
// Register the iSAC encoder.
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
codec_registered_ = true;
}
if (codec_registered_ && receive_packet_count_ > kNumPackets) {
test_complete_.Set();
}
}
rtc::PlatformThread receive_thread_;
rtc::PlatformThread codec_registration_thread_;
// Used to force worker threads to stop looping.
std::atomic<bool> quit_;
rtc::Event test_complete_;
Mutex mutex_;
bool codec_registered_ RTC_GUARDED_BY(mutex_);
int receive_packet_count_ RTC_GUARDED_BY(mutex_);
int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_);
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_TRUE(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)
@ -719,6 +1025,38 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {};
// Run bit exactness tests only for release builds.
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(NDEBUG) && defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(/*audio_checksum_ref=*/"37ecdabad1698a857cf811e6d1fa91df",
/*payload_checksum_ref=*/"3c79f16f34218271f3dca4e2b1dfe1bb",
/*expected_packets=*/33,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(/*audio_checksum_ref=*/"0e9078d23454901496a88362ba0740c3",
/*payload_checksum_ref=*/"9e0a0ab743ad987b55b8e14802769c56",
/*expected_packets=*/16,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
// Run bit exactness test only for release build.
#if defined(WEBRTC_CODEC_ISAC) && defined(NDEBUG) && defined(WEBRTC_LINUX) && \
defined(WEBRTC_ARCH_X86_64)
TEST_F(AcmSenderBitExactnessOldApi, IsacSwb30ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
Run(/*audio_checksum_ref=*/"f4cf577f28a0dcbac33358b757518e0c",
/*payload_checksum_ref=*/"ce86106a93419aefb063097108ec94ab",
/*expected_packets=*/33,
/*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
}
#endif
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
Run(/*audio_checksum_ref=*/"69118ed438ac76252d023e0463819471",