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/*
* 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
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