text: /*
   *  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/g711/audio_encoder_pcm.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 NetEqPcmuQualityTest : public NetEqQualityTest {
   protected:
    NetEqPcmuQualityTest()
        : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
                           kInputSampleRateKhz,
                           kOutputSampleRateKhz,
                           SdpAudioFormat("pcmu", 8000, 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 {
      ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
      AudioEncoderPcmU::Config config;
      config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
      encoder_.reset(new AudioEncoderPcmU(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 = 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<AudioEncoderPcmU> encoder_;
  };
  
  TEST_F(NetEqPcmuQualityTest, Test) {
    Simulate();
  }
  
  }  // namespace test
  }  // namespace webrtc
  