void
MediaAudioTest::testCreation() {
const int32_t numSamples = 1024;
const int32_t sampleRate = 22050;
const int32_t channels = 8;
const AudioChannel::Layout layout = AudioChannel::CH_LAYOUT_7POINT1;
const AudioFormat::Type format = AudioFormat::SAMPLE_FMT_S16P;
// now let's test invalid methods.
RefPointer<MediaAudio> audio;
{
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_ERROR, false);
TS_ASSERT_THROWS(MediaAudio::make(-1, sampleRate, channels, layout, format),
HumbleInvalidArgument);
}
{
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_ERROR, false);
TS_ASSERT_THROWS(MediaAudio::make(numSamples, -1, channels, layout, format),
HumbleInvalidArgument);
}
{
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_ERROR, false);
TS_ASSERT_THROWS(MediaAudio::make(numSamples, sampleRate, -1, layout, format),
HumbleInvalidArgument);
}
{
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_ERROR, false);
TS_ASSERT_THROWS(MediaAudio::make(numSamples, sampleRate, channels + 1, layout,
format), HumbleInvalidArgument);
}
{
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_ERROR, false);
TS_ASSERT_THROWS(MediaAudio::make(numSamples, sampleRate, channels, layout,
AudioFormat::SAMPLE_FMT_NONE), HumbleInvalidArgument);
}
// And this should be valid
audio = MediaAudio::make(numSamples, sampleRate, channels + 1,
AudioChannel::CH_LAYOUT_UNKNOWN, format);
TS_ASSERT(audio);
audio = MediaAudio::make(numSamples, sampleRate, channels, layout, format);
TS_ASSERT(audio);
// now let's try getting the data
RefPointer<Buffer> buf;
TS_ASSERT_EQUALS(numSamples, audio->getMaxNumSamples());
TS_ASSERT_EQUALS(numSamples, audio->getNumSamples());
TS_ASSERT(!audio->isComplete());
TS_ASSERT_EQUALS(channels, audio->getChannels());
TS_ASSERT_EQUALS(channels, audio->getNumDataPlanes());
TS_ASSERT_EQUALS(layout, audio->getChannelLayout());
TS_ASSERT_EQUALS(sampleRate, audio->getSampleRate());
TS_ASSERT_EQUALS(format, audio->getFormat());
{
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_ERROR, false);
TS_ASSERT_THROWS(
audio->setNumSamples(audio->getMaxNumSamples()+1),
HumbleInvalidArgument);
TS_ASSERT_THROWS(audio->setNumSamples(-1), HumbleInvalidArgument);
TS_ASSERT_THROWS(audio->setNumSamples(0), HumbleInvalidArgument);
}
audio->setNumSamples(1);
TS_ASSERT_EQUALS(1, audio->getNumSamples());
audio->setComplete(true);
TS_ASSERT(audio->isComplete());
for (int i = 0; i < channels; i++) {
buf = audio->getData(i);
TS_ASSERT(buf);
TS_ASSERT_EQUALS(audio->getDataPlaneSize(i),
audio->getNumSamples()*AudioFormat::getBytesPerSample(audio->getFormat())*(audio->isPlanar()?1:audio->getChannels()));
}
// now let's try packed audio
audio = MediaAudio::make(numSamples, sampleRate, channels, layout,
AudioFormat::getPackedSampleFormat(format));
TS_ASSERT(audio);
TS_ASSERT_EQUALS(AudioFormat::getPackedSampleFormat(format),
audio->getFormat());
TS_ASSERT_EQUALS(channels, audio->getChannels());
TS_ASSERT_EQUALS(1, audio->getNumDataPlanes());
buf = audio->getData(0);
TS_ASSERT(buf);
for (int i = 1; i < channels; i++) {
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_ERROR, false);
TS_ASSERT_THROWS(audio->getData(i), HumbleInvalidArgument);
}
}
void
EncoderTest::testRegression36Internal (const Codec::ID codecId,
const int32_t numSamples,
const int32_t sampleRate,
const int32_t channels,
const AudioChannel::Layout channelLayout,
const AudioFormat::Type audioFormat,
const int64_t bitRate,
const char* testOutputName)
{
VS_LOG_DEBUG("Output filename: %s", testOutputName);
RefPointer<Codec> codec = Codec::findEncodingCodec (codecId);
RefPointer<Encoder> encoder = Encoder::make (codec.value ());
RefPointer<FilterGraph> graph = FilterGraph::make ();
RefPointer<MediaAudio> audio = MediaAudio::make (numSamples, sampleRate,
channels, channelLayout,
audioFormat);
// set the encoder properties we need
encoder->setSampleRate (audio->getSampleRate ());
encoder->setSampleFormat (audio->getFormat ());
encoder->setChannelLayout (audio->getChannelLayout ());
encoder->setChannels (audio->getChannels ());
encoder->setProperty ("b", (int64_t) (bitRate));
RefPointer<Rational> tb = Rational::make (1, sampleRate);
encoder->setTimeBase (tb.value ());
// create an output muxer
RefPointer<Muxer> muxer = Muxer::make (testOutputName, 0, 0);
RefPointer<MuxerFormat> format = muxer->getFormat ();
if (format->getFlag (MuxerFormat::GLOBAL_HEADER))
encoder->setFlag (Encoder::FLAG_GLOBAL_HEADER, true);
// open the encoder
encoder->open (0, 0);
RefPointer<FilterAudioSink> fsink = graph->addAudioSink (
"out", audio->getSampleRate (), audio->getChannelLayout (),
audio->getFormat ());
// Generate a 220 Hz sine wave with a 880 Hz beep each second, for 10 seconds.
graph->open ("sine=frequency=220:beep_factor=4:duration=11[out]");
fsink->setFrameSize (numSamples);
// add a stream for the encoded packets
{
RefPointer<MuxerStream> stream = muxer->addNewStream (encoder.value ());
} // and open the muxer
muxer->open (0, 0);
// now we're (in theory) ready to start writing data.
int32_t numCompletePackets = 0;
RefPointer<MediaPacket> packet;
// Get one audio packet that is larger than the frame-size.
fsink->getAudio (audio.value ());
TS_ASSERT(audio->isComplete ());
TS_ASSERT_EQUALS(audio->getNumSamples (), sampleRate);
audio->setTimeStamp (0);
// let's encode
packet = MediaPacket::make ();
encoder->encodeAudio (packet.value (), audio.value ());
if (packet->isComplete ())
{
muxer->write (packet.value (), false);
}
// now flush the encoder
do
{
packet = MediaPacket::make ();
encoder->encodeAudio (packet.value (), 0);
if (packet->isComplete ())
{
muxer->write (packet.value (), false);
++numCompletePackets;
}
}
while (packet->isComplete ());
muxer->close ();
if (!(codec->getCapabilities() & Codec::CAP_VARIABLE_FRAME_SIZE)) {
const int32_t numExpectedPackets = audio->getNumSamples()
/ encoder->getFrameSize();
VS_LOG_DEBUG("%ld vs %ld; framesize: %ld", numCompletePackets, numExpectedPackets, encoder->getFrameSize());
TS_ASSERT(numCompletePackets > 10);
}
}
void
MediaAudioTest::testCopy() {
const int32_t numSamples = 155; // I choose an odd number because HV will align up to 32
const int32_t sampleRate = 22050;
const int32_t channels = 15; // choose a large # of channels to make sure we expand the Frame
const AudioChannel::Layout layout = AudioChannel::CH_LAYOUT_UNKNOWN;
const AudioFormat::Type format = AudioFormat::SAMPLE_FMT_DBLP;
int32_t bufSize = AudioFormat::getBufferSizeNeeded(numSamples, channels,
format);
// test that there is rounding up
int32_t minSize = AudioFormat::getBytesPerSample(format) * numSamples
* channels;
TS_ASSERT_LESS_THAN(minSize, bufSize);
RefPointer<Buffer> src = Buffer::make(0, bufSize);
double* srcData = (double*) src->getBytes(0, bufSize);
// now, let's go nuts!
for (size_t i = 0; i < bufSize / sizeof(double); i++) {
srcData[i] = i;
}
RefPointer<MediaAudio> audio;
audio = MediaAudio::make(src.value(), numSamples, sampleRate, channels,
layout, format);
TS_ASSERT(audio);
TS_ASSERT_EQUALS(channels, audio->getNumDataPlanes());
bool tests[] =
{ true, false };
for (size_t i = 0; i < sizeof(tests) / sizeof(*tests); i++) {
// now let's make a copy
RefPointer<MediaAudio> copy = MediaAudio::make(audio.value(), tests[i]);
TS_ASSERT_EQUALS(copy->getMaxNumSamples(), audio->getMaxNumSamples());
TS_ASSERT_EQUALS(copy->getNumSamples(), audio->getNumSamples());
TS_ASSERT_EQUALS(copy->getChannels(), audio->getChannels());
TS_ASSERT_EQUALS(copy->getNumDataPlanes(), audio->getNumDataPlanes());
TS_ASSERT_EQUALS(copy->getChannelLayout(), audio->getChannelLayout());
TS_ASSERT_EQUALS(copy->getSampleRate(), audio->getSampleRate());
TS_ASSERT_EQUALS(copy->getFormat(), audio->getFormat());
for (int32_t j = 0; j < audio->getNumDataPlanes(); j++) {
RefPointer<Buffer> srcBuf = audio->getData(j);
RefPointer<Buffer> dstBuf = copy->getData(j);
int32_t planeSize = srcBuf->getBufferSize();
TS_ASSERT_EQUALS(planeSize, dstBuf->getBufferSize());
uint8_t* srcBytes = (uint8_t*) srcBuf->getBytes(0, planeSize);
uint8_t* dstBytes = (uint8_t*) dstBuf->getBytes(0, planeSize);
if (tests[i]) {
for (int32_t k = 0; k < planeSize; k++) {
// should be byte-by-byte the same
TS_ASSERT_EQUALS(srcBytes[k], dstBytes[k]);
}
} else {
TS_ASSERT_EQUALS(srcBytes, dstBytes);
}
}
}
}
void
EncoderTest::testEncodeAudio() {
Logger::setGlobalIsLogging(Logger::LEVEL_TRACE, false);
LoggerStack stack;
stack.setGlobalLevel(Logger::LEVEL_INFO, false);
const bool isMemCheck = getenv("VS_TEST_MEMCHECK") ? true : false;
const int32_t sampleRate = 44100;
const int32_t maxSamples = isMemCheck ? sampleRate*0.5 : sampleRate*10;
const int32_t numSamples = 1024;
const AudioChannel::Layout channelLayout = AudioChannel::CH_LAYOUT_STEREO;
const int32_t channels = AudioChannel::getNumChannelsInLayout(channelLayout);
const AudioFormat::Type audioFormat = AudioFormat::SAMPLE_FMT_S16;
RefPointer<Codec> codec = Codec::findEncodingCodec(Codec::CODEC_ID_AAC);
RefPointer<Encoder> encoder = Encoder::make(codec.value());
RefPointer<FilterGraph> graph = FilterGraph::make();
RefPointer<MediaAudio> audio = MediaAudio::make(numSamples, sampleRate, channels, channelLayout,
audioFormat);
// set the encoder properties we need
encoder->setSampleRate(audio->getSampleRate());
encoder->setSampleFormat(audio->getFormat());
encoder->setChannelLayout(audio->getChannelLayout());
encoder->setChannels(audio->getChannels());
encoder->setProperty("b", (int64_t)64000); // bitrate
RefPointer<Rational> tb = Rational::make(1,25);
encoder->setTimeBase(tb.value());
// create an output muxer
RefPointer<Muxer> muxer = Muxer::make("EncoderTest_encodeAudio.mp4", 0, 0);
RefPointer<MuxerFormat> format = muxer->getFormat();
if (format->getFlag(MuxerFormat::GLOBAL_HEADER))
encoder->setFlag(Encoder::FLAG_GLOBAL_HEADER, true);
// open the encoder
encoder->open(0, 0);
RefPointer<FilterAudioSink> fsink = graph->addAudioSink("out", audio->getSampleRate(), audio->getChannelLayout(), audio->getFormat());
// Generate a 220 Hz sine wave with a 880 Hz beep each second, for 10 seconds.
graph->open("sine=frequency=660:beep_factor=4:duration=11[out]");
// Generate an amplitude modulated signal
//graph->open("aevalsrc=sin(10*2*PI*t)*sin(880*2*PI*t)[out]");
// add a stream for the encoded packets
{
RefPointer<MuxerStream> stream = muxer->addNewStream(encoder.value());
}
// and open the muxer
muxer->open(0, 0);
// now we're (in theory) ready to start writing data.
int32_t numFrames = 0;
RefPointer<MediaPacket> packet;
while(fsink->getAudio(audio.value()) >= 0 && audio->isComplete() && numFrames*audio->getNumSamples() < maxSamples) {
audio->setTimeStamp(numFrames*audio->getNumSamples());
// let's encode
packet = MediaPacket::make();
encoder->encodeAudio(packet.value(), audio.value());
if (packet->isComplete()) {
muxer->write(packet.value(), false);
}
++numFrames;
}
// now flush the encoder
do {
packet = MediaPacket::make();
encoder->encodeAudio(packet.value(), 0);
if (packet->isComplete()) {
muxer->write(packet.value(), false);
}
} while (packet->isComplete());
muxer->close();
}
