blob: fdbe39cce2627e3cdb295ee6018b0ce30d0a03b7 [file] [log] [blame]
/*
* 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