1 files changed, 81 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
new file mode 100644
index 0000000000..c3e160cb66
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2018 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/pcm16b/audio_encoder_pcm16b.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 = 48;
+static const int kOutputSampleRateKhz = 48;
+}  // namespace
+
+class NetEqPcm16bQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcm16bQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("l16", 48000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
+              absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
+              (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+  }
+
+  void SetUp() override {
+    AudioEncoderPcm16B::Config config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    config.sample_rate_hz = 48000;
+    config.num_channels = channels_;
+    encoder_.reset(new AudioEncoderPcm16B(config));
+    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 = 480;  // 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<AudioEncoderPcm16B> encoder_;
+};
+
+TEST_F(NetEqPcm16bQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc