Reland "[ACM] iSAC audio codec removed"
This is a reland of commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1
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: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
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",