[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}
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index e73acc2..6dd44b6 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -13,6 +13,7 @@
#include <algorithm> // std::min
#include <memory>
+#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
@@ -64,12 +65,14 @@
const SdpAudioFormat& format,
const std::map<int, int> cng_payload_types = {}) {
// Create the speech encoder.
- AudioCodecInfo info = encoder_factory_->QueryAudioEncoder(format).value();
+ absl::optional<AudioCodecInfo> info =
+ encoder_factory_->QueryAudioEncoder(format);
+ RTC_CHECK(info.has_value());
std::unique_ptr<AudioEncoder> enc =
encoder_factory_->MakeAudioEncoder(payload_type, format, absl::nullopt);
// If we have a compatible CN specification, stack a CNG on top.
- auto it = cng_payload_types.find(info.sample_rate_hz);
+ auto it = cng_payload_types.find(info->sample_rate_hz);
if (it != cng_payload_types.end()) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(enc);
@@ -81,7 +84,7 @@
// Actually start using the new encoder.
acm_->SetEncoder(std::move(enc));
- return info;
+ return *info;
}
int InsertOnePacketOfSilence(const AudioCodecInfo& info) {
@@ -148,8 +151,7 @@
#define MAYBE_SampleRate SampleRate
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
- const std::map<int, SdpAudioFormat> codecs = {{0, {"ISAC", 16000, 1}},
- {1, {"ISAC", 32000, 1}}};
+ const std::map<int, SdpAudioFormat> codecs = {{0, {"OPUS", 48000, 2}}};
receiver_->SetCodecs(codecs);
constexpr int kOutSampleRateHz = 8000; // Different than codec sample rate.
@@ -233,15 +235,6 @@
}
#if defined(WEBRTC_ANDROID)
-#define MAYBE_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC
-#else
-#define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC
-#endif
-TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) {
- RunVerifyAudioFrame({"ISAC", 16000, 1});
-}
-
-#if defined(WEBRTC_ANDROID)
#define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
#else
#define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
@@ -310,12 +303,10 @@
#else
#define MAYBE_LastAudioCodec LastAudioCodec
#endif
-#if defined(WEBRTC_CODEC_ISAC)
+#if defined(WEBRTC_CODEC_OPUS)
TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
- const std::map<int, SdpAudioFormat> codecs = {{0, {"ISAC", 16000, 1}},
- {1, {"PCMA", 8000, 1}},
- {2, {"ISAC", 32000, 1}},
- {3, {"L16", 32000, 1}}};
+ const std::map<int, SdpAudioFormat> codecs = {
+ {0, {"PCMU", 8000, 1}}, {1, {"PCMA", 8000, 1}}, {2, {"L16", 32000, 1}}};
const std::map<int, int> cng_payload_types = {
{8000, 100}, {16000, 101}, {32000, 102}};
{
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 7e4b764..f1eb81c 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -30,7 +30,6 @@
#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"
@@ -302,44 +301,6 @@
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
@@ -420,8 +381,7 @@
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.
+// A multi-threaded test for ACM that uses the PCM16b 16 kHz codec.
class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
protected:
static const int kNumPackets = 500;
@@ -560,272 +520,6 @@
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)
@@ -1025,38 +719,6 @@
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",