audio_utils/tests/spdif_tests.cpp

/*
 * Copyright (C) 2020 The Android Open Source Project
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *      http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#include <array>
#include <climits>
#include <math.h>
#include <memory>
#include <string.h>

#include <gtest/gtest.h>

#include <audio_utils/spdif/SPDIFEncoder.h>

using namespace android;

class MySPDIFEncoder : public SPDIFEncoder {
public:

    explicit MySPDIFEncoder(audio_format_t format)
            : SPDIFEncoder(format)
    {
    }
    // Defaults to AC3 format. Was in original API.
    MySPDIFEncoder() = default;

    ssize_t writeOutput( const void* /* buffer */, size_t numBytes ) override {
        mOutputSizeBytes = numBytes;
        return numBytes;
    }

    FrameScanner *getFramer() const { return mFramer; }
    size_t        getByteCursor() const { return mByteCursor; }
    size_t        getPayloadBytesPending() const { return mPayloadBytesPending; }
    size_t        getBurstBufferSizeBytes() const { return mBurstBufferSizeBytes; }

    size_t                     mOutputSizeBytes = 0;
};

// This is the beginning of the file voice1-48k-64kbps-15s.ac3
static const uint8_t sVoice1ch48k_AC3[] = {
    0x0b, 0x77, 0x44, 0xcd, 0x08, 0x40, 0x2f, 0x84, 0x29, 0xca, 0x6e, 0x44, 0xa4, 0xfd, 0xce, 0xf7,
    0xc9, 0x9f, 0x3e, 0x74, 0xfa, 0x01, 0x0a, 0xda, 0xb3, 0x3e, 0xb0, 0x95, 0xf2, 0x5a, 0xef, 0x9e
};

// This is the beginning of the file channelcheck_48k6ch.eac3
static const uint8_t sChannel6ch48k_EAC3[] = {
    0x0b, 0x77, 0x01, 0xbf, 0x3f, 0x85, 0x7f, 0xe8, 0x1e, 0x40, 0x82, 0x10, 0x00, 0x00, 0x00, 0x01,
    0x00, 0x00, 0x00, 0x03, 0xfc, 0x60, 0x80, 0x7e, 0x59, 0x00, 0xfc, 0xf3, 0xcf, 0x01, 0xf9, 0xe7
};

static const uint8_t sZeros[32] = { 0 };

static constexpr int kBytesPerOutputFrame = 2 * sizeof(int16_t); // stereo

TEST(audio_utils_spdif, SupportedFormats)
{
    ASSERT_FALSE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_PCM_FLOAT));
    ASSERT_FALSE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_PCM_16_BIT));
    ASSERT_FALSE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_MP3));

    ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_AC3));
    ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_E_AC3));
    ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_DTS));
    ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_DTS_HD));
}

TEST(audio_utils_spdif, ScanAC3)
{
    MySPDIFEncoder encoder(AUDIO_FORMAT_AC3);
    FrameScanner *scanner = encoder.getFramer();
    // It should recognize the valid AC3 header.
    int i = 0;
    while (i < 5) {
        ASSERT_FALSE(scanner->scan(sVoice1ch48k_AC3[i++]));
    }
    ASSERT_TRUE(scanner->scan(sVoice1ch48k_AC3[i++]));
    ASSERT_FALSE(scanner->scan(sVoice1ch48k_AC3[i++]));
}

TEST(audio_utils_spdif, WriteAC3)
{
    MySPDIFEncoder encoder(AUDIO_FORMAT_AC3);
    encoder.write(sVoice1ch48k_AC3, sizeof(sVoice1ch48k_AC3));
    ASSERT_EQ(48000, encoder.getFramer()->getSampleRate());
    ASSERT_EQ(kBytesPerOutputFrame, encoder.getBytesPerOutputFrame());
    ASSERT_EQ(1, encoder.getRateMultiplier());

    // Check to make sure that the pending bytes calculation did not overflow.
    size_t burstBufferSizeBytes = encoder.getBurstBufferSizeBytes(); // allocated maximum size
    size_t pendingBytes = encoder.getPayloadBytesPending();
    ASSERT_GE(burstBufferSizeBytes, pendingBytes);

    // Write some fake compressed audio to force an output data burst.
    for (int i = 0; i < 7; i++) {
        auto result = encoder.write(sZeros, sizeof(sZeros));
        ASSERT_EQ(sizeof(sZeros), result);
    }
    // This value is calculated in SPDIFEncoder::sendZeroPad()
    //    size_t burstSize = mFramer->getSampleFramesPerSyncFrame() * sizeof(uint16_t)
    //        * SPDIF_ENCODED_CHANNEL_COUNT;
    // If it changes then there is probably a regression.
    const int kExpectedBurstSize = 6144;
    ASSERT_EQ(kExpectedBurstSize, encoder.mOutputSizeBytes);
}

TEST(audio_utils_spdif, ValidEAC3)
{
    MySPDIFEncoder encoder(AUDIO_FORMAT_E_AC3);
    auto result = encoder.write(sChannel6ch48k_EAC3, sizeof(sChannel6ch48k_EAC3));
    ASSERT_EQ(sizeof(sChannel6ch48k_EAC3), result);
    ASSERT_EQ(4, encoder.getRateMultiplier()); // EAC3_RATE_MULTIPLIER
    ASSERT_EQ(48000, encoder.getFramer()->getSampleRate());
    ASSERT_EQ(kBytesPerOutputFrame, encoder.getBytesPerOutputFrame());

    // Check to make sure that the pending bytes calculation did not overflow.
    size_t bufferSize = encoder.getBurstBufferSizeBytes();
    size_t pendingBytes = encoder.getPayloadBytesPending();
    ASSERT_GE(bufferSize, pendingBytes);
}

TEST(audio_utils_spdif, InvalidLengthEAC3)
{
    MySPDIFEncoder encoder(AUDIO_FORMAT_E_AC3);
    // Mangle a valid header and try to force a numeric overflow.
    uint8_t mangled[sizeof(sChannel6ch48k_EAC3)] = {0};
    memcpy(mangled, sChannel6ch48k_EAC3, sizeof(sChannel6ch48k_EAC3));

    // force frmsiz to zero!
    mangled[2] = mangled[2] & 0xF8;
    mangled[3] = 0;
    auto result = encoder.write(mangled, sizeof(mangled));
    ASSERT_EQ(sizeof(mangled), result);

    // Check to make sure that the pending bytes calculation did not overflow.
    size_t bufferSize = encoder.getBurstBufferSizeBytes();
    size_t pendingBytes = encoder.getPayloadBytesPending();
    ASSERT_GE(bufferSize, pendingBytes);

}