blob: d2833d8926a2eed731bac8da0af9eecc45501053 [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 "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.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 >= 10 && value <= 60 && (value % 10) == 0)
return true;
printf("Invalid frame size, should be 10, 20, ..., 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 NetEqPcmuQualityTest : public NetEqQualityTest {
protected:
NetEqPcmuQualityTest()
: NetEqQualityTest(FLAGS_frame_size_ms,
kInputSampleRateKhz,
kOutputSampleRateKhz,
kDecoderPCMu,
1) {
AudioEncoderPcmU::Config config;
config.frame_size_ms = FLAGS_frame_size_ms;
encoder_.reset(new AudioEncoderPcmU(config));
}
int EncodeBlock(int16_t* in_data,
int block_size_samples,
uint8_t* payload,
int max_bytes) override {
const int kFrameSizeSamples = 80; // Samples per 10 ms.
int encoded_samples = 0;
uint32_t dummy_timestamp = 0;
AudioEncoder::EncodedInfo info;
do {
info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
kFrameSizeSamples, max_bytes, payload);
encoded_samples += kFrameSizeSamples;
} while (info.encoded_bytes == 0);
return rtc::checked_cast<int>(info.encoded_bytes);
}
private:
rtc::scoped_ptr<AudioEncoderPcmU> encoder_;
};
TEST_F(NetEqPcmuQualityTest, Test) {
Simulate();
}
} // namespace test
} // namespace webrtc