[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}
This commit is contained in:
Alessio Bazzica
committed by
WebRTC LUCI CQ
WebRTC LUCI CQ
parent
6aa755c201
commit
b46c4bf27b
@ -22,10 +22,6 @@
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#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
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#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
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#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
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#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
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#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
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#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
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#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
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#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
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#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
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#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
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@ -195,8 +191,8 @@ class AudioDecoderTest : public ::testing::Test {
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processed_samples += frame_size_;
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}
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// For some codecs it doesn't make sense to check expected number of bytes,
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// since the number can vary for different platforms. Opus and iSAC are
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// such codecs. In this case expected_bytes is set to 0.
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// since the number can vary for different platforms. Opus is such a codec.
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// In this case expected_bytes is set to 0.
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if (expected_bytes) {
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EXPECT_EQ(expected_bytes, encoded_bytes);
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}
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@ -347,66 +343,6 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
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}
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};
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class AudioDecoderIsacFloatTest : public AudioDecoderTest {
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protected:
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AudioDecoderIsacFloatTest() : AudioDecoderTest() {
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codec_input_rate_hz_ = 16000;
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frame_size_ = 480;
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data_length_ = 10 * frame_size_;
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AudioEncoderIsacFloatImpl::Config config;
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config.payload_type = payload_type_;
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config.sample_rate_hz = codec_input_rate_hz_;
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config.frame_size_ms =
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1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
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audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
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audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
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AudioDecoderIsacFloatImpl::Config decoder_config;
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decoder_config.sample_rate_hz = codec_input_rate_hz_;
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decoder_ = new AudioDecoderIsacFloatImpl(decoder_config);
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}
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};
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class AudioDecoderIsacSwbTest : public AudioDecoderTest {
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protected:
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AudioDecoderIsacSwbTest() : AudioDecoderTest() {
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codec_input_rate_hz_ = 32000;
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frame_size_ = 960;
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data_length_ = 10 * frame_size_;
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AudioEncoderIsacFloatImpl::Config config;
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config.payload_type = payload_type_;
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config.sample_rate_hz = codec_input_rate_hz_;
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config.frame_size_ms =
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1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
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audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
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audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
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AudioDecoderIsacFloatImpl::Config decoder_config;
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decoder_config.sample_rate_hz = codec_input_rate_hz_;
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decoder_ = new AudioDecoderIsacFloatImpl(decoder_config);
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}
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};
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class AudioDecoderIsacFixTest : public AudioDecoderTest {
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protected:
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AudioDecoderIsacFixTest() : AudioDecoderTest() {
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codec_input_rate_hz_ = 16000;
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frame_size_ = 480;
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data_length_ = 10 * frame_size_;
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AudioEncoderIsacFixImpl::Config config;
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config.payload_type = payload_type_;
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config.sample_rate_hz = codec_input_rate_hz_;
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config.frame_size_ms =
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1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
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audio_encoder_.reset(new AudioEncoderIsacFixImpl(config));
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audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
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AudioDecoderIsacFixImpl::Config decoder_config;
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decoder_config.sample_rate_hz = codec_input_rate_hz_;
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decoder_ = new AudioDecoderIsacFixImpl(decoder_config);
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}
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};
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class AudioDecoderG722Test : public AudioDecoderTest {
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protected:
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AudioDecoderG722Test() : AudioDecoderTest() {
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@ -533,94 +469,6 @@ TEST_F(AudioDecoderIlbcTest, SetTargetBitrate) {
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TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 13333);
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}
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TEST_F(AudioDecoderIsacFloatTest, EncodeDecode) {
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int tolerance = 3399;
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double mse = 434951.0;
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int delay = 48; // Delay from input to output.
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EncodeDecodeTest(0, tolerance, mse, delay);
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ReInitTest();
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EXPECT_FALSE(decoder_->HasDecodePlc());
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}
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TEST_F(AudioDecoderIsacFloatTest, SetTargetBitrate) {
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const int overhead_rate =
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8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
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EXPECT_EQ(10000,
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SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate));
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EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(),
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10000 + overhead_rate));
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EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(),
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23456 + overhead_rate));
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EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
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32000 + overhead_rate));
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EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
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32001 + overhead_rate));
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}
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TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) {
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int tolerance = 19757;
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double mse = 8.18e6;
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int delay = 160; // Delay from input to output.
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EncodeDecodeTest(0, tolerance, mse, delay);
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ReInitTest();
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EXPECT_FALSE(decoder_->HasDecodePlc());
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}
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TEST_F(AudioDecoderIsacSwbTest, SetTargetBitrate) {
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const int overhead_rate =
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8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
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EXPECT_EQ(10000,
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SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate));
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EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(),
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10000 + overhead_rate));
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EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(),
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23456 + overhead_rate));
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EXPECT_EQ(56000, SetAndGetTargetBitrate(audio_encoder_.get(),
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56000 + overhead_rate));
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EXPECT_EQ(56000, SetAndGetTargetBitrate(audio_encoder_.get(),
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56001 + overhead_rate));
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}
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// Run bit exactness test only for release builds.
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#if defined(NDEBUG)
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TEST_F(AudioDecoderIsacFixTest, EncodeDecode) {
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int tolerance = 11034;
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double mse = 3.46e6;
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int delay = 54; // Delay from input to output.
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#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
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static const int kEncodedBytes = 685;
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#elif defined(WEBRTC_MAC) && defined(WEBRTC_ARCH_ARM64) // M1 Mac
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static const int kEncodedBytes = 673;
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#elif defined(WEBRTC_ARCH_ARM64)
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static const int kEncodedBytes = 673;
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#elif defined(WEBRTC_WIN) && defined(_MSC_VER) && !defined(__clang__)
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static const int kEncodedBytes = 671;
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#elif defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_X86_64)
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static const int kEncodedBytes = 671;
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#else
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static const int kEncodedBytes = 671;
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#endif
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EncodeDecodeTest(kEncodedBytes, tolerance, mse, delay);
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ReInitTest();
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EXPECT_FALSE(decoder_->HasDecodePlc());
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}
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#endif
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TEST_F(AudioDecoderIsacFixTest, SetTargetBitrate) {
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const int overhead_rate =
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8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
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EXPECT_EQ(10000,
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SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate));
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EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(),
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10000 + overhead_rate));
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EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(),
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23456 + overhead_rate));
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EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
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32000 + overhead_rate));
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EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
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32001 + overhead_rate));
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}
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TEST_F(AudioDecoderG722Test, EncodeDecode) {
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int tolerance = 6176;
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double mse = 238630.0;
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@ -1,102 +0,0 @@
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "absl/flags/flag.h"
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#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
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#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
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ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
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using ::testing::InitGoogleTest;
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namespace webrtc {
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namespace test {
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namespace {
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static const int kIsacBlockDurationMs = 30;
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static const int kIsacInputSamplingKhz = 16;
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static const int kIsacOutputSamplingKhz = 16;
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} // namespace
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class NetEqIsacQualityTest : public NetEqQualityTest {
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protected:
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NetEqIsacQualityTest();
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void SetUp() override;
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void TearDown() override;
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int EncodeBlock(int16_t* in_data,
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size_t block_size_samples,
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rtc::Buffer* payload,
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size_t max_bytes) override;
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private:
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ISACFIX_MainStruct* isac_encoder_;
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int bit_rate_kbps_;
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};
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NetEqIsacQualityTest::NetEqIsacQualityTest()
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: NetEqQualityTest(kIsacBlockDurationMs,
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kIsacInputSamplingKhz,
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kIsacOutputSamplingKhz,
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SdpAudioFormat("isac", 16000, 1)),
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isac_encoder_(NULL),
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bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)) {
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// Flag validation
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RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 10 &&
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absl::GetFlag(FLAGS_bit_rate_kbps) <= 32)
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<< "Invalid bit rate, should be between 10 and 32 kbps.";
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}
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void NetEqIsacQualityTest::SetUp() {
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ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio.";
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// Create encoder memory.
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WebRtcIsacfix_Create(&isac_encoder_);
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ASSERT_TRUE(isac_encoder_ != NULL);
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EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(isac_encoder_, 1));
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// Set bitrate and block length.
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EXPECT_EQ(0, WebRtcIsacfix_Control(isac_encoder_, bit_rate_kbps_ * 1000,
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kIsacBlockDurationMs));
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NetEqQualityTest::SetUp();
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}
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void NetEqIsacQualityTest::TearDown() {
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// Free memory.
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EXPECT_EQ(0, WebRtcIsacfix_Free(isac_encoder_));
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NetEqQualityTest::TearDown();
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}
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int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
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size_t block_size_samples,
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rtc::Buffer* payload,
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size_t max_bytes) {
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// ISAC takes 10 ms for every call.
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const int subblocks = kIsacBlockDurationMs / 10;
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const int subblock_length = 10 * kIsacInputSamplingKhz;
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int value = 0;
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int pointer = 0;
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for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
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// The Isac encoder does not perform encoding (and returns 0) until it
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// receives a sequence of sub-blocks that amount to the frame duration.
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EXPECT_EQ(0, value);
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payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
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value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer],
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payload.data());
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return (value >= 0) ? static_cast<size_t>(value) : 0;
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});
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}
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EXPECT_GT(value, 0);
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return value;
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}
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TEST_F(NetEqIsacQualityTest, Test) {
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Simulate();
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}
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} // namespace test
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} // namespace webrtc
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@ -30,7 +30,6 @@
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#include "api/audio_codecs/g711/audio_encoder_g711.h"
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#include "api/audio_codecs/g722/audio_encoder_g722.h"
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#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
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#include "api/audio_codecs/isac/audio_encoder_isac.h"
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#include "api/audio_codecs/opus/audio_encoder_opus.h"
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#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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@ -71,7 +70,6 @@ enum class CodecType {
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kPcm16b32,
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kPcm16b48,
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kIlbc,
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kIsac
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};
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struct CodecTypeAndInfo {
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@ -94,8 +92,7 @@ const std::map<std::string, CodecTypeAndInfo>& CodecList() {
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{"pcm16b_16", {CodecType::kPcm16b16, 94, false}},
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{"pcm16b_32", {CodecType::kPcm16b32, 95, false}},
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{"pcm16b_48", {CodecType::kPcm16b48, 96, false}},
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{"ilbc", {CodecType::kIlbc, 102, false}},
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{"isac", {CodecType::kIsac, 103, false}}};
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{"ilbc", {CodecType::kIlbc, 102, false}}};
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return *codec_list;
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}
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@ -236,11 +233,6 @@ std::unique_ptr<AudioEncoder> CreateEncoder(CodecType codec_type,
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return AudioEncoderIlbc::MakeAudioEncoder(
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GetCodecConfig<AudioEncoderIlbc>(), payload_type);
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}
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case CodecType::kIsac: {
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return AudioEncoderIsac::MakeAudioEncoder(
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GetCodecConfig<AudioEncoderIsac>(), payload_type);
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}
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}
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RTC_DCHECK_NOTREACHED();
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return nullptr;
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