| /* |
| * 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 "webrtc/base/checks.h" |
| #include "webrtc/base/safe_conversions.h" |
| #include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| |
| using google::RegisterFlagValidator; |
| using google::ParseCommandLineFlags; |
| using std::string; |
| using testing::InitGoogleTest; |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| static const int kInputSampleRateKhz = 8; |
| static const int kOutputSampleRateKhz = 8; |
| |
| // Define switch for frame size. |
| static bool ValidateFrameSize(const char* flagname, int32_t value) { |
| if (value == 20 || value == 30 || value == 40 || value == 60) |
| return true; |
| printf("Invalid frame size, should be 20, 30, 40, or 60 ms."); |
| return false; |
| } |
| |
| DEFINE_int32(frame_size_ms, 20, "Codec frame size (milliseconds)."); |
| |
| static const bool frame_size_dummy = |
| RegisterFlagValidator(&FLAGS_frame_size_ms, &ValidateFrameSize); |
| |
| } // namespace |
| |
| class NetEqIlbcQualityTest : public NetEqQualityTest { |
| protected: |
| NetEqIlbcQualityTest() |
| : NetEqQualityTest(FLAGS_frame_size_ms, |
| kInputSampleRateKhz, |
| kOutputSampleRateKhz, |
| NetEqDecoder::kDecoderILBC) {} |
| |
| void SetUp() override { |
| ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio."; |
| AudioEncoderIlbc::Config config; |
| config.frame_size_ms = FLAGS_frame_size_ms; |
| encoder_.reset(new AudioEncoderIlbc(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<AudioEncoderIlbc> encoder_; |
| }; |
| |
| TEST_F(NetEqIlbcQualityTest, Test) { |
| Simulate(); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |