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Reland "[ACM] iSAC audio codec removed"

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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}
This commit is contained in:
Alessio Bazzica
2022-11-11 16:52:46 +01:00
committed by WebRTC LUCI CQ
parent e158b77427
commit 17887eb04a
164 changed files with 117 additions and 39429 deletions
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156
modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -22,10 +22,6 @@
#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
@ -195,8 +191,8 @@ class AudioDecoderTest : public ::testing::Test {
processed_samples += frame_size_;
}
// For some codecs it doesn't make sense to check expected number of bytes,
// since the number can vary for different platforms. Opus and iSAC are
// such codecs. In this case expected_bytes is set to 0.
// since the number can vary for different platforms. Opus is such a codec.
// In this case expected_bytes is set to 0.
if (expected_bytes) {
EXPECT_EQ(expected_bytes, encoded_bytes);
}
@ -347,66 +343,6 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
}
};
class AudioDecoderIsacFloatTest : public AudioDecoderTest {
protected:
AudioDecoderIsacFloatTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 16000;
frame_size_ = 480;
data_length_ = 10 * frame_size_;
AudioEncoderIsacFloatImpl::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
AudioDecoderIsacFloatImpl::Config decoder_config;
decoder_config.sample_rate_hz = codec_input_rate_hz_;
decoder_ = new AudioDecoderIsacFloatImpl(decoder_config);
}
};
class AudioDecoderIsacSwbTest : public AudioDecoderTest {
protected:
AudioDecoderIsacSwbTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 32000;
frame_size_ = 960;
data_length_ = 10 * frame_size_;
AudioEncoderIsacFloatImpl::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
AudioDecoderIsacFloatImpl::Config decoder_config;
decoder_config.sample_rate_hz = codec_input_rate_hz_;
decoder_ = new AudioDecoderIsacFloatImpl(decoder_config);
}
};
class AudioDecoderIsacFixTest : public AudioDecoderTest {
protected:
AudioDecoderIsacFixTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 16000;
frame_size_ = 480;
data_length_ = 10 * frame_size_;
AudioEncoderIsacFixImpl::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsacFixImpl(config));
audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
AudioDecoderIsacFixImpl::Config decoder_config;
decoder_config.sample_rate_hz = codec_input_rate_hz_;
decoder_ = new AudioDecoderIsacFixImpl(decoder_config);
}
};
class AudioDecoderG722Test : public AudioDecoderTest {
protected:
AudioDecoderG722Test() : AudioDecoderTest() {
@ -533,94 +469,6 @@ TEST_F(AudioDecoderIlbcTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 13333);
}
TEST_F(AudioDecoderIsacFloatTest, EncodeDecode) {
int tolerance = 3399;
double mse = 434951.0;
int delay = 48; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderIsacFloatTest, SetTargetBitrate) {
const int overhead_rate =
8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
EXPECT_EQ(10000,
SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate));
EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(),
10000 + overhead_rate));
EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(),
23456 + overhead_rate));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
32000 + overhead_rate));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
32001 + overhead_rate));
}
TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) {
int tolerance = 19757;
double mse = 8.18e6;
int delay = 160; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderIsacSwbTest, SetTargetBitrate) {
const int overhead_rate =
8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
EXPECT_EQ(10000,
SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate));
EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(),
10000 + overhead_rate));
EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(),
23456 + overhead_rate));
EXPECT_EQ(56000, SetAndGetTargetBitrate(audio_encoder_.get(),
56000 + overhead_rate));
EXPECT_EQ(56000, SetAndGetTargetBitrate(audio_encoder_.get(),
56001 + overhead_rate));
}
// Run bit exactness test only for release builds.
#if defined(NDEBUG)
TEST_F(AudioDecoderIsacFixTest, EncodeDecode) {
int tolerance = 11034;
double mse = 3.46e6;
int delay = 54; // Delay from input to output.
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
static const int kEncodedBytes = 685;
#elif defined(WEBRTC_MAC) && defined(WEBRTC_ARCH_ARM64) // M1 Mac
static const int kEncodedBytes = 673;
#elif defined(WEBRTC_ARCH_ARM64)
static const int kEncodedBytes = 673;
#elif defined(WEBRTC_WIN) && defined(_MSC_VER) && !defined(__clang__)
static const int kEncodedBytes = 671;
#elif defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_X86_64)
static const int kEncodedBytes = 671;
#else
static const int kEncodedBytes = 671;
#endif
EncodeDecodeTest(kEncodedBytes, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
#endif
TEST_F(AudioDecoderIsacFixTest, SetTargetBitrate) {
const int overhead_rate =
8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
EXPECT_EQ(10000,
SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate));
EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(),
10000 + overhead_rate));
EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(),
23456 + overhead_rate));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
32000 + overhead_rate));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
32001 + overhead_rate));
}
TEST_F(AudioDecoderG722Test, EncodeDecode) {
int tolerance = 6176;
double mse = 238630.0;