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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc | 81 |
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 |