C++ AUD_Reference::read方法代码示例

void AUD_FileWriter::writeReader(AUD_Reference<AUD_IReader> reader, std::vector<AUD_Reference<AUD_IWriter> >& writers, unsigned int length, unsigned int buffersize)
{
	AUD_Buffer buffer(buffersize * AUD_SAMPLE_SIZE(reader->getSpecs()));
	AUD_Buffer buffer2(buffersize * sizeof(sample_t));
	sample_t* buf = buffer.getBuffer();
	sample_t* buf2 = buffer2.getBuffer();

	int len;
	bool eos = false;
	int channels = reader->getSpecs().channels;

	for(unsigned int pos = 0; ((pos < length) || (length <= 0)) && !eos; pos += len)
	{
		len = buffersize;
		if((len > length - pos) && (length > 0))
			len = length - pos;
		reader->read(len, eos, buf);

		for(int channel = 0; channel < channels; channel++)
		{
			for(int i = 0; i < len; i++)
			{
				// clamping!
				if(buf[i * channels + channel] > 1)
					buf2[i] = 1;
				else if(buf[i * channels + channel] < -1)
					buf2[i] = -1;
				else
					buf2[i] = buf[i * channels + channel];
			}

			writers[channel]->write(len, buf2);
		}
	}
}

float* AUD_readSoundBuffer(const char* filename, float low, float high,
						   float attack, float release, float threshold,
						   int accumulate, int additive, int square,
						   float sthreshold, double samplerate, int* length)
{
	AUD_Buffer buffer;
	AUD_DeviceSpecs specs;
	specs.channels = AUD_CHANNELS_MONO;
	specs.rate = (AUD_SampleRate)samplerate;
	AUD_Reference<AUD_IFactory> sound;

	AUD_Reference<AUD_IFactory> file = new AUD_FileFactory(filename);

	AUD_Reference<AUD_IReader> reader = file->createReader();
	AUD_SampleRate rate = reader->getSpecs().rate;

	sound = new AUD_ChannelMapperFactory(file, specs);

	if(high < rate)
		sound = new AUD_LowpassFactory(sound, high);
	if(low > 0)
		sound = new AUD_HighpassFactory(sound, low);

	sound = new AUD_EnvelopeFactory(sound, attack, release, threshold, 0.1f);
	sound = new AUD_LinearResampleFactory(sound, specs);

	if(square)
		sound = new AUD_SquareFactory(sound, sthreshold);

	if(accumulate)
		sound = new AUD_AccumulatorFactory(sound, additive);
	else if(additive)
		sound = new AUD_SumFactory(sound);

	reader = sound->createReader();

	if(reader.isNull())
		return NULL;

	int len;
	int position = 0;
	bool eos;
	do
	{
		len = samplerate;
		buffer.resize((position + len) * sizeof(float), true);
		reader->read(len, eos, buffer.getBuffer() + position);
		position += len;
	} while(!eos);

	float* result = (float*)malloc(position * sizeof(float));
	memcpy(result, buffer.getBuffer(), position * sizeof(float));
	*length = position;
	return result;
}

AUD_OpenALDevice::AUD_OpenALHandle::AUD_OpenALHandle(AUD_OpenALDevice* device, ALenum format, AUD_Reference<AUD_IReader> reader, bool keep) :
	m_isBuffered(false), m_reader(reader), m_keep(keep), m_format(format), m_current(0),
	m_eos(false), m_loopcount(0), m_stop(NULL), m_stop_data(NULL), m_status(AUD_STATUS_PLAYING),
	m_device(device)
{
	AUD_DeviceSpecs specs = m_device->m_specs;
	specs.specs = m_reader->getSpecs();

	// OpenAL playback code
	alGenBuffers(CYCLE_BUFFERS, m_buffers);
	if(alGetError() != AL_NO_ERROR)
		AUD_THROW(AUD_ERROR_OPENAL, genbuffer_error);

	try
	{
		m_device->m_buffer.assureSize(m_device->m_buffersize * AUD_DEVICE_SAMPLE_SIZE(specs));
		int length;
		bool eos;

		for(int i = 0; i < CYCLE_BUFFERS; i++)
		{
			length = m_device->m_buffersize;
			reader->read(length, eos, m_device->m_buffer.getBuffer());
			alBufferData(m_buffers[i], m_format, m_device->m_buffer.getBuffer(),
						 length * AUD_DEVICE_SAMPLE_SIZE(specs),
						 specs.rate);
			if(alGetError() != AL_NO_ERROR)
				AUD_THROW(AUD_ERROR_OPENAL, bufferdata_error);
		}

		alGenSources(1, &m_source);
		if(alGetError() != AL_NO_ERROR)
			AUD_THROW(AUD_ERROR_OPENAL, gensource_error);

		try
		{
			alSourceQueueBuffers(m_source, CYCLE_BUFFERS,
								 m_buffers);
			if(alGetError() != AL_NO_ERROR)
				AUD_THROW(AUD_ERROR_OPENAL, queue_error);
		}
		catch(AUD_Exception&)
		{
			alDeleteSources(1, &m_source);
			throw;
		}
	}
	catch(AUD_Exception&)
	{
		alDeleteBuffers(CYCLE_BUFFERS, m_buffers);
		throw;
	}
	alSourcei(m_source, AL_SOURCE_RELATIVE, 1);
}

int AUD_readSound(AUD_Sound* sound, sample_t* buffer, int length, int samples_per_second)
{
	AUD_DeviceSpecs specs;
	sample_t* buf;
	AUD_Buffer aBuffer;

	specs.rate = AUD_RATE_INVALID;
	specs.channels = AUD_CHANNELS_MONO;
	specs.format = AUD_FORMAT_INVALID;

	AUD_Reference<AUD_IReader> reader = AUD_ChannelMapperFactory(*sound, specs).createReader();

	specs.specs = reader->getSpecs();
	int len;
	float samplejump = specs.rate / samples_per_second;
	float min, max, power;
	bool eos;

	for(int i = 0; i < length; i++)
	{
		len = floor(samplejump * (i+1)) - floor(samplejump * i);

		aBuffer.assureSize(len * AUD_SAMPLE_SIZE(specs));
		buf = aBuffer.getBuffer();

		reader->read(len, eos, buf);

		max = min = *buf;
		power = *buf * *buf;
		for(int j = 1; j < len; j++)
		{
			if(buf[j] < min)
				min = buf[j];
			if(buf[j] > max)
				max = buf[j];
			power += buf[j] * buf[j];
		}

		buffer[i * 3] = min;
		buffer[i * 3 + 1] = max;
		buffer[i * 3 + 2] = sqrt(power) / len;

		if(eos)
		{
			length = i;
			break;
		}
	}

	return length;
}