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diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
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+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
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+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+} // namespace
+
+class NetEqIlbcQualityTest : public NetEqQualityTest {
+ protected:
+ NetEqIlbcQualityTest()
+ : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+ kInputSampleRateKhz,
+ kOutputSampleRateKhz,
+ SdpAudioFormat("ilbc", 8000, 1)) {
+ // Flag validation
+ RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) == 20 ||
+ absl::GetFlag(FLAGS_frame_size_ms) == 30 ||
+ absl::GetFlag(FLAGS_frame_size_ms) == 40 ||
+ absl::GetFlag(FLAGS_frame_size_ms) == 60)
+ << "Invalid frame size, should be 20, 30, 40, or 60 ms.";
+ }
+
+ void SetUp() override {
+ ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
+ AudioEncoderIlbcConfig config;
+ config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+ encoder_.reset(new AudioEncoderIlbcImpl(config, 102));
+ NetEqQualityTest::SetUp();
+ }
+
+ int EncodeBlock(int16_t* in_data,
+ size_t block_size_samples,
+ rtc::Buffer* payload,
+ size_t max_bytes) override {
+ const size_t kFrameSizeSamples = 80; // Samples per 10 ms.
+ size_t encoded_samples = 0;
+ uint32_t dummy_timestamp = 0;
+ AudioEncoder::EncodedInfo info;
+ do {
+ info = encoder_->Encode(dummy_timestamp,
+ rtc::ArrayView<const int16_t>(
+ in_data + encoded_samples, kFrameSizeSamples),
+ payload);
+ encoded_samples += kFrameSizeSamples;
+ } while (info.encoded_bytes == 0);
+ return rtc::checked_cast<int>(info.encoded_bytes);
+ }
+
+ private:
+ std::unique_ptr<AudioEncoderIlbcImpl> encoder_;
+};
+
+TEST_F(NetEqIlbcQualityTest, Test) {
+ Simulate();
+}
+
+} // namespace test
+} // namespace webrtc