示例#1
0
void Mixer::clear(int length)
{
	m_buffer.assureSize(length * m_specs.channels * AUD_SAMPLE_SIZE(m_specs));

	m_length = length;

	std::memset(m_buffer.getBuffer(), 0, length * m_specs.channels * AUD_SAMPLE_SIZE(m_specs));
}
void AUD_FileWriter::writeReader(AUD_Reference<AUD_IReader> reader, AUD_Reference<AUD_IWriter> writer, unsigned int length, unsigned int buffersize)
{
	AUD_Buffer buffer(buffersize * AUD_SAMPLE_SIZE(writer->getSpecs()));
	sample_t* buf = buffer.getBuffer();

	int len;
	bool eos = false;
	int channels = writer->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 i = 0; i < len * channels; i++)
		{
			// clamping!
			if(buf[i] > 1)
				buf[i] = 1;
			else if(buf[i] < -1)
				buf[i] = -1;
		}

		writer->write(len, buf);
	}
}
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);
		}
	}
}
示例#4
0
void AUD_SuperposeReader::read(int& length, bool& eos, sample_t* buffer)
{
	AUD_Specs specs = m_reader1->getSpecs();
	AUD_Specs s2 = m_reader2->getSpecs();
	if(!AUD_COMPARE_SPECS(specs, s2))
		AUD_THROW(AUD_ERROR_SPECS, specs_error);

	int samplesize = AUD_SAMPLE_SIZE(specs);

	m_buffer.assureSize(length * samplesize);

	int len1 = length;
	m_reader1->read(len1, eos, buffer);

	if(len1 < length)
		memset(buffer + len1 * specs.channels, 0, (length - len1) * samplesize);

	int len2 = length;
	bool eos2;
	sample_t* buf = m_buffer.getBuffer();
	m_reader2->read(len2, eos2, buf);

	for(int i = 0; i < len2 * specs.channels; i++)
		buffer[i] += buf[i];

	length = AUD_MAX(len1, len2);
	eos &= eos2;
}
示例#5
0
void AUD_SuperposeReader::read(int & length, sample_t* & buffer)
{
	AUD_Specs specs = m_reader1->getSpecs();
	int samplesize = AUD_SAMPLE_SIZE(specs);

	if(m_buffer.getSize() < length * samplesize)
		m_buffer.resize(length * samplesize);
	buffer = m_buffer.getBuffer();

	int len1 = length;
	sample_t* buf;
	m_reader1->read(len1, buf);
	memcpy(buffer, buf, len1 * samplesize);

	if(len1 < length)
		memset(buffer + len1 * specs.channels, 0, (length - len1) * samplesize);

	int len2 = length;
	m_reader2->read(len2, buf);

	for(int i = 0; i < len2 * specs.channels; i++)
		buffer[i] += buf[i];

	length = AUD_MAX(len1, len2);
}
void AUD_BaseIIRFilterReader::read(int & length, sample_t* & buffer)
{
	sample_t* buf;

	int samplesize = AUD_SAMPLE_SIZE(m_reader->getSpecs());

	m_reader->read(length, buf);

	if(m_buffer.getSize() < length * samplesize)
		m_buffer.resize(length * samplesize);

	buffer = m_buffer.getBuffer();

	for(m_channel = 0; m_channel < m_channels; m_channel++)
	{
		for(int i = 0; i < length; i++)
		{
			m_x[m_xpos * CC] = buf[i * CC];
			m_y[m_ypos * CC] = buffer[i * CC] = filter();

			m_xpos = (m_xpos + 1) % m_xlen;
			m_ypos = (m_ypos + 1) % m_ylen;
		}
	}
}
示例#7
0
AUD_NAMESPACE_BEGIN

void JackDevice::updateRingBuffers()
{
	size_t size, temp;
	unsigned int samplesize = AUD_SAMPLE_SIZE(m_specs);
	unsigned int i, j;
	unsigned int channels = m_specs.channels;
	sample_t* buffer = m_buffer.getBuffer();
	float* deinterleave = m_deinterleavebuf.getBuffer();
	jack_transport_state_t state;
	jack_position_t position;

	std::unique_lock<std::mutex> lock(m_mixingLock);

	while(m_valid)
	{
		if(m_sync > 1)
		{
			if(m_syncFunc)
			{
				state = AUD_jack_transport_query(m_client, &position);
				m_syncFunc(m_syncFuncData, state != JackTransportStopped, position.frame / (float) m_specs.rate);
			}

			for(i = 0; i < channels; i++)
				AUD_jack_ringbuffer_reset(m_ringbuffers[i]);
		}

		size = AUD_jack_ringbuffer_write_space(m_ringbuffers[0]);
		for(i = 1; i < channels; i++)
			if((temp = AUD_jack_ringbuffer_write_space(m_ringbuffers[i])) < size)
				size = temp;

		while(size > samplesize)
		{
			size /= samplesize;
			mix((data_t*)buffer, size);
			for(i = 0; i < channels; i++)
			{
				for(j = 0; j < size; j++)
					deinterleave[i * size + j] = buffer[i + j * channels];
				AUD_jack_ringbuffer_write(m_ringbuffers[i], (char*)(deinterleave + i * size), size * sizeof(float));
			}

			size = AUD_jack_ringbuffer_write_space(m_ringbuffers[0]);
			for(i = 1; i < channels; i++)
				if((temp = AUD_jack_ringbuffer_write_space(m_ringbuffers[i])) < size)
					size = temp;
		}

		if(m_sync > 1)
		{
			m_sync = 3;
		}

		m_mixingCondition.wait(lock);
	}
}
示例#8
0
AUD_NAMESPACE_BEGIN

LimiterReader::LimiterReader(std::shared_ptr<IReader> reader, float start, float end) :
		EffectReader(reader),
		m_start(start),
		m_end(end)
{
	if(m_start > 0)
	{
		Specs specs = m_reader->getSpecs();
		Specs specs2;

		if(m_reader->isSeekable())
			m_reader->seek(m_start * specs.rate);
		else
		{
			// skip first m_start samples by reading them
			int length = AUD_DEFAULT_BUFFER_SIZE;
			Buffer buffer(AUD_DEFAULT_BUFFER_SIZE * AUD_SAMPLE_SIZE(specs));
			bool eos = false;
			for(int len = m_start * specs.rate;
				length > 0 && !eos;
				len -= length)
			{
				if(len < AUD_DEFAULT_BUFFER_SIZE)
					length = len;

				m_reader->read(length, eos, buffer.getBuffer());

				specs2 = m_reader->getSpecs();
				if(specs2.rate != specs.rate)
				{
					len = len * specs2.rate / specs.rate;
					specs.rate = specs2.rate;
				}

				if(specs2.channels != specs.channels)
				{
					specs = specs2;
					buffer.assureSize(AUD_DEFAULT_BUFFER_SIZE * AUD_SAMPLE_SIZE(specs));
				}
			}
		}
	}
}
long AUD_SRCResampleReader::doCallback(float** data)
{
	int length = m_buffer.getSize() / AUD_SAMPLE_SIZE(m_tspecs);
	sample_t* buffer;

	m_reader->read(length, buffer);

	*data = buffer;
	return length;
}
AUD_LinearResampleReader::AUD_LinearResampleReader(boost::shared_ptr<AUD_IReader> reader,
												   AUD_Specs specs) :
	AUD_ResampleReader(reader, specs.rate),
	m_channels(reader->getSpecs().channels),
	m_cache_pos(0),
	m_cache_ok(false)
{
	specs.channels = m_channels;
	m_cache.resize(2 * AUD_SAMPLE_SIZE(specs));
}
示例#11
0
void FFMPEGWriter::write(unsigned int length, sample_t* buffer)
{
	unsigned int samplesize = AUD_SAMPLE_SIZE(m_specs);

	if(m_input_size)
	{
		sample_t* inbuf = m_input_buffer.getBuffer();

		while(length)
		{
			unsigned int len = std::min(m_input_size - m_input_samples, length);

			std::memcpy(inbuf + m_input_samples * m_specs.channels, buffer, len * samplesize);

			buffer += len * m_specs.channels;
			m_input_samples += len;
			m_position += len;
			length -= len;

			if(m_input_samples == m_input_size)
			{
				encode();

				m_input_samples = 0;
			}
		}
	}
	else // PCM data, can write directly!
	{
		int samplesize = AUD_SAMPLE_SIZE(m_specs);
		m_input_buffer.assureSize(length * std::max(AUD_DEVICE_SAMPLE_SIZE(m_specs), samplesize));

		sample_t* buf = m_input_buffer.getBuffer();
		m_convert(reinterpret_cast<data_t*>(buf), reinterpret_cast<data_t*>(buffer), length * m_specs.channels);

		m_input_samples = length;

		m_position += length;

		encode();
	}
}
void AUD_FFMPEGWriter::write(unsigned int length, sample_t* buffer)
{
	unsigned int samplesize = AUD_SAMPLE_SIZE(m_specs);

	if(m_input_size)
	{
		sample_t* inbuf = m_input_buffer.getBuffer();

		while(length)
		{
			unsigned int len = AUD_MIN(m_input_size - m_input_samples, length);

			memcpy(inbuf + m_input_samples * m_specs.channels, buffer, len * samplesize);

			buffer += len * m_specs.channels;
			m_input_samples += len;
			m_position += len;
			length -= len;

			if(m_input_samples == m_input_size)
			{
				encode(inbuf);

				m_input_samples = 0;
			}
		}
	}
	else // PCM data, can write directly!
	{
		int samplesize = AUD_SAMPLE_SIZE(m_specs);
		if(m_output_buffer.getSize() != length * m_specs.channels * m_codecCtx->bits_per_coded_sample / 8)
			m_output_buffer.resize(length * m_specs.channels * m_codecCtx->bits_per_coded_sample / 8);
		m_input_buffer.assureSize(length * AUD_MAX(AUD_DEVICE_SAMPLE_SIZE(m_specs), samplesize));

		sample_t* buf = m_input_buffer.getBuffer();
		m_convert(reinterpret_cast<data_t*>(buf), reinterpret_cast<data_t*>(buffer), length * m_specs.channels);

		encode(buf);

		m_position += length;
	}
}
示例#13
0
void AUD_ConverterReader::read(int& length, bool& eos, sample_t* buffer)
{
	AUD_Specs specs = m_reader->getSpecs();
	int samplesize = AUD_SAMPLE_SIZE(specs);

	m_buffer.assureSize(length * samplesize);

	m_reader->read(length, eos, m_buffer.getBuffer());

	m_convert((data_t*)buffer, (data_t*)m_buffer.getBuffer(),
	          length * specs.channels);
}
AUD_NAMESPACE_BEGIN

LinearResampleReader::LinearResampleReader(std::shared_ptr<IReader> reader,
												   Specs specs) :
	ResampleReader(reader, specs.rate),
	m_channels(reader->getSpecs().channels),
	m_cache_pos(0),
	m_cache_ok(false)
{
	specs.channels = m_channels;
	m_cache.resize(2 * AUD_SAMPLE_SIZE(specs));
}
示例#15
0
long AUD_SRCResampleReader::doCallback(float** data)
{
	AUD_Specs specs;
	specs.channels = m_channels;
	specs.rate = m_rate;

	int length = m_buffer.getSize() / AUD_SAMPLE_SIZE(specs);

	*data = m_buffer.getBuffer();
	m_reader->read(length, m_eos, *data);

	return length;
}
示例#16
0
void AUD_SRCResampleReader::read(int & length, sample_t* & buffer)
{
	int size = length * AUD_SAMPLE_SIZE(m_tspecs);

	if(m_buffer.getSize() < size)
		m_buffer.resize(size);

	buffer = m_buffer.getBuffer();

	length = src_callback_read(m_src, m_factor, length, buffer);

	m_position += length;
}
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;
}
示例#18
0
void AUD_SndFileReader::read(int & length, sample_t* & buffer)
{
	int sample_size = AUD_SAMPLE_SIZE(m_specs);

	// resize output buffer if necessary
	if(m_buffer.getSize() < length*sample_size)
		m_buffer.resize(length*sample_size);

	buffer = m_buffer.getBuffer();

	length = sf_readf_float(m_sndfile, buffer, length);

	m_position += length;
}
void AUD_ChannelMapperReader::read(int & length, sample_t* & buffer)
{
	sample_t* in = buffer;

	m_reader->read(length, in);

	if(m_buffer.getSize() < length * AUD_SAMPLE_SIZE(m_specs))
		m_buffer.resize(length * AUD_SAMPLE_SIZE(m_specs));

	buffer = m_buffer.getBuffer();
	sample_t sum;

	for(int i = 0; i < length; i++)
	{
		for(int j = 0; j < m_specs.channels; j++)
		{
			sum = 0;
			for(int k = 0; k < m_rch; k++)
				sum += m_mapping[j][k] * in[i * m_rch + k];
			buffer[i * m_specs.channels + j] = sum;
		}
	}
}
示例#20
0
void AUD_LoopReader::read(int & length, sample_t* & buffer)
{
	AUD_Specs specs = m_reader->getSpecs();
	int samplesize = AUD_SAMPLE_SIZE(specs);

	int len = length;

	m_reader->read(len, buffer);

	if(len < length && m_left)
	{
		int pos = 0;

		if(m_buffer.getSize() < length * samplesize)
			m_buffer.resize(length * samplesize);

		sample_t* buf = m_buffer.getBuffer();

		memcpy(buf + pos * specs.channels, buffer, len * samplesize);

		pos += len;

		while(pos < length && m_left)
		{
			if(m_left > 0)
				m_left--;

			m_reader->seek(0);

			len = length - pos;
			m_reader->read(len, buffer);

			// prevent endless loop
			if(!len)
				break;

			memcpy(buf + pos * specs.channels, buffer, len * samplesize);

			pos += len;
		}

		length = pos;
		buffer = buf;
	}
	else
		length = len;
}
示例#21
0
void AUD_ReverseReader::read(int & length, sample_t* & buffer)
{
	// first correct the length
	if(m_position + length > m_length)
		length = m_length - m_position;

	if(length <= 0)
	{
		length = 0;
		return;
	}

	AUD_Specs specs = getSpecs();
	int samplesize = AUD_SAMPLE_SIZE(specs);

	// resize buffer if needed
	if(m_buffer.getSize() < length * samplesize)
		m_buffer.resize(length * samplesize);

	buffer = m_buffer.getBuffer();

	sample_t* buf;
	int len = length;

	// read from reader
	m_reader->seek(m_length - m_position - len);
	m_reader->read(len, buf);

	// set null if reader didn't give enough data
	if(len < length)
	{
		memset(buffer, 0, (length - len) * samplesize);
		buffer += (length - len) * specs.channels;
	}

	// copy the samples reverted
	for(int i = 0; i < len; i++)
		memcpy(buffer + i * specs.channels,
			   buf + (len - 1 - i) * specs.channels,
			   samplesize);

	m_position += length;

	buffer = m_buffer.getBuffer();
}
示例#22
0
void AUD_LoopReader::read(int & length, sample_t* & buffer)
{
	int samplesize = AUD_SAMPLE_SIZE(m_reader->getSpecs());

	int len = length;

	m_reader->read(len, buffer);

	if(len < length && m_loop != 0)
	{
		int pos = 0;

		if(m_buffer->getSize() < length*samplesize)
			m_buffer->resize(length*samplesize);

		memcpy(m_buffer->getBuffer() + pos * samplesize,
			   buffer, len * samplesize);

		pos += len;

		while(pos < length && m_loop != 0)
		{
			if(m_loop > 0)
				m_loop--;

			m_reader->seek(0);

			len = length - pos;
			m_reader->read(len, buffer);
			// prevent endless loop
			if(!len)
				break;

			memcpy(m_buffer->getBuffer() + pos * samplesize,
				   buffer, len * samplesize);

			pos += len;
		}

		length = pos;
		buffer = m_buffer->getBuffer();
	}
	else
		length = len;
}
示例#23
0
void FaderReader::read(int& length, bool& eos, sample_t* buffer)
{
	int position = m_reader->getPosition();
	Specs specs = m_reader->getSpecs();
	int samplesize = AUD_SAMPLE_SIZE(specs);

	m_reader->read(length, eos, buffer);

	if((position + length) / (float)specs.rate <= m_start)
	{
		if(m_type != FADE_OUT)
		{
			std::memset(buffer, 0, length * samplesize);
		}
	}
	else if(position / (float)specs.rate >= m_start+m_length)
	{
		if(m_type == FADE_OUT)
		{
			std::memset(buffer, 0, length * samplesize);
		}
	}
	else
	{
		float volume = 1.0f;

		for(int i = 0; i < length * specs.channels; i++)
		{
			if(i % specs.channels == 0)
			{
				volume = (((position+i)/(float)specs.rate)-m_start) / m_length;
				if(volume > 1.0f)
					volume = 1.0f;
				else if(volume < 0.0f)
					volume = 0.0f;

				if(m_type == FADE_OUT)
					volume = 1.0f - volume;
			}

			buffer[i] = buffer[i] * volume;
		}
	}
}
示例#24
0
void AUD_SRCResampleReader::read(int& length, bool& eos, sample_t* buffer)
{
	AUD_Specs specs = m_reader->getSpecs();

	double factor = double(m_rate) / double(specs.rate);

	specs.rate = m_rate;

	int size = length;

	m_buffer.assureSize(length * AUD_SAMPLE_SIZE(specs));

	if(specs.channels != m_channels)
	{
		src_delete(m_src);

		m_channels = specs.channels;

		int error;
		m_src = src_callback_new(src_callback,
								 SRC_SINC_MEDIUM_QUALITY,
								 m_channels,
								 &error,
								 this);

		if(!m_src)
		{
			// XXX printf("%s\n", src_strerror(error));
			AUD_THROW(AUD_ERROR_SRC, state_error);
		}
	}

	m_eos = false;

	length = src_callback_read(m_src, factor, length, buffer);

	m_position += length;

	eos = m_eos && (length < size);
}
示例#25
0
void ReverseReader::read(int& length, bool& eos, sample_t* buffer)
{
    // first correct the length
    if(m_position + length > m_length)
        length = m_length - m_position;

    if(length <= 0)
    {
        length = 0;
        eos = true;
        return;
    }

    const Specs specs = getSpecs();
    const int samplesize = AUD_SAMPLE_SIZE(specs);

    sample_t temp[CHANNEL_MAX];

    int len = length;

    // read from reader
    m_reader->seek(m_length - m_position - len);
    m_reader->read(len, eos, buffer);

    // set null if reader didn't give enough data
    if(len < length)
        std::memset(buffer, 0, (length - len) * samplesize);

    // copy the samples reverted
    for(int i = 0; i < length / 2; i++)
    {
        std::memcpy(temp, buffer + (len - 1 - i) * specs.channels, samplesize);
        std::memcpy(buffer + (len - 1 - i) * specs.channels, buffer + i * specs.channels, samplesize);
        std::memcpy(buffer + i * specs.channels, temp, samplesize);
    }

    m_position += length;
    eos = false;
}
示例#26
0
AUD_NAMESPACE_BEGIN

StreamBuffer::StreamBuffer(std::shared_ptr<ISound> sound) :
	m_buffer(new Buffer())
{
	std::shared_ptr<IReader> reader = sound->createReader();

	m_specs = reader->getSpecs();

	int sample_size = AUD_SAMPLE_SIZE(m_specs);
	int length;
	int index = 0;
	bool eos = false;

	// get an approximated size if possible
	int size = reader->getLength();

	if(size <= 0)
		size = BUFFER_RESIZE_BYTES / sample_size;
	else
		size += m_specs.rate;

	// as long as the end of the stream is not reached
	while(!eos)
	{
		// increase
		m_buffer->resize(size*sample_size, true);

		// read more
		length = size-index;
		reader->read(length, eos, m_buffer->getBuffer() + index * m_specs.channels);
		if(index == m_buffer->getSize() / sample_size)
			size += BUFFER_RESIZE_BYTES / sample_size;
		index += length;
	}

	m_buffer->resize(index * sample_size, true);
}
示例#27
0
void AUD_BufferReader::read(int& length, bool& eos, sample_t* buffer)
{
	eos = false;

	int sample_size = AUD_SAMPLE_SIZE(m_specs);

	sample_t* buf = m_buffer->getBuffer() + m_position * m_specs.channels;

	// in case the end of the buffer is reached
	if(m_buffer->getSize() < (m_position + length) * sample_size)
	{
		length = m_buffer->getSize() / sample_size - m_position;
		eos = true;
	}

	if(length < 0)
	{
		length = 0;
		return;
	}

	m_position += length;
	memcpy(buffer, buf, length * sample_size);
}
示例#28
0
void AUD_SoftwareDevice::mix(data_t* buffer, int length)
{
	m_buffer.assureSize(length * AUD_SAMPLE_SIZE(m_specs));

	lock();

	{
		AUD_Reference<AUD_SoftwareDevice::AUD_SoftwareHandle> sound;
		int len;
		int pos;
		bool eos;
		std::list<AUD_Reference<AUD_SoftwareDevice::AUD_SoftwareHandle> > stopSounds;
		std::list<AUD_Reference<AUD_SoftwareDevice::AUD_SoftwareHandle> > pauseSounds;
		sample_t* buf = m_buffer.getBuffer();

		m_mixer->clear(length);

		// for all sounds
		AUD_HandleIterator it = m_playingSounds.begin();
		while(it != m_playingSounds.end())
		{
			sound = *it;
			// increment the iterator to make sure it's valid,
			// in case the sound gets deleted after stopping
			++it;

			// get the buffer from the source
			pos = 0;
			len = length;

			// update 3D Info
			sound->update();

			sound->m_reader->read(len, eos, buf);

			// in case of looping
			while(pos + len < length && sound->m_loopcount && eos)
			{
				m_mixer->mix(buf, pos, len, sound->m_volume);

				pos += len;

				if(sound->m_loopcount > 0)
					sound->m_loopcount--;

				sound->m_reader->seek(0);

				len = length - pos;
				sound->m_reader->read(len, eos, buf);

				// prevent endless loop
				if(!len)
					break;
			}

			m_mixer->mix(buf, pos, len, sound->m_volume);

			// in case the end of the sound is reached
			if(eos && !sound->m_loopcount)
			{
				if(sound->m_stop)
					sound->m_stop(sound->m_stop_data);

				if(sound->m_keep)
					pauseSounds.push_back(sound);
				else
					stopSounds.push_back(sound);
			}
		}

		// superpose
		m_mixer->read(buffer, m_volume);

		// cleanup
		while(!stopSounds.empty())
		{
			sound = stopSounds.front();
			stopSounds.pop_front();
			sound->stop();
		}

		while(!pauseSounds.empty())
		{
			sound = pauseSounds.front();
			pauseSounds.pop_front();
			sound->pause();
		}
	}

	unlock();
}
示例#29
0
AUD_NAMESPACE_BEGIN

void FFMPEGWriter::encode()
{
	sample_t* data = m_input_buffer.getBuffer();

	if(m_deinterleave)
	{
		m_deinterleave_buffer.assureSize(m_input_buffer.getSize());

		sample_t* dbuf = m_deinterleave_buffer.getBuffer();
		// deinterleave
		int single_size = sizeof(sample_t);
		for(int channel = 0; channel < m_specs.channels; channel++)
		{
			for(int i = 0; i < m_input_buffer.getSize() / AUD_SAMPLE_SIZE(m_specs); i++)
			{
				std::memcpy(((data_t*)dbuf) + (m_input_samples * channel + i) * single_size,
							((data_t*)data) + ((m_specs.channels * i) + channel) * single_size, single_size);
			}
		}

		// convert first
		if(m_input_size)
			m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(dbuf), m_input_samples * m_specs.channels);
		else
			std::memcpy(data, dbuf, m_input_buffer.getSize());
	}
	else
		// convert first
		if(m_input_size)
			m_convert(reinterpret_cast<data_t*>(data), reinterpret_cast<data_t*>(data), m_input_samples * m_specs.channels);

	AVPacket packet;

	packet.data = nullptr;
	packet.size = 0;

	av_init_packet(&packet);

	AVFrame* frame = av_frame_alloc();
	av_frame_unref(frame);
	int got_packet;

	frame->nb_samples = m_input_samples;
	frame->format = m_codecCtx->sample_fmt;
	frame->channel_layout = m_codecCtx->channel_layout;

	if(avcodec_fill_audio_frame(frame, m_specs.channels, m_codecCtx->sample_fmt, reinterpret_cast<data_t*>(data), m_input_buffer.getSize(), 0) < 0)
		AUD_THROW(FileException, "File couldn't be written, filling the audio frame failed with ffmpeg.");

	AVRational sample_time = { 1, static_cast<int>(m_specs.rate) };
	frame->pts = av_rescale_q(m_position - m_input_samples, m_codecCtx->time_base, sample_time);

	if(avcodec_encode_audio2(m_codecCtx, &packet, frame, &got_packet))
	{
		av_frame_free(&frame);
		AUD_THROW(FileException, "File couldn't be written, audio encoding failed with ffmpeg.");
	}

	if(got_packet)
	{
		packet.flags |= AV_PKT_FLAG_KEY;
		packet.stream_index = m_stream->index;
		if(av_write_frame(m_formatCtx, &packet) < 0)
		{
			av_free_packet(&packet);
			av_frame_free(&frame);
			AUD_THROW(FileException, "Frame couldn't be writen to the file with ffmpeg.");
		}
		av_free_packet(&packet);
	}

	av_frame_free(&frame);
}
示例#30
0
FFMPEGWriter::FFMPEGWriter(std::string filename, DeviceSpecs specs, Container format, Codec codec, unsigned int bitrate) :
	m_position(0),
	m_specs(specs),
	m_input_samples(0),
	m_deinterleave(false)
{
	static const char* formats[] = { nullptr, "ac3", "flac", "matroska", "mp2", "mp3", "ogg", "wav" };

	if(avformat_alloc_output_context2(&m_formatCtx, nullptr, formats[format], filename.c_str()) < 0)
		AUD_THROW(FileException, "File couldn't be written, format couldn't be found with ffmpeg.");

	m_outputFmt = m_formatCtx->oformat;

	if(!m_outputFmt) {
		avformat_free_context(m_formatCtx);
		AUD_THROW(FileException, "File couldn't be written, output format couldn't be found with ffmpeg.");
	}

	m_outputFmt->audio_codec = AV_CODEC_ID_NONE;

	switch(codec)
	{
	case CODEC_AAC:
		m_outputFmt->audio_codec = AV_CODEC_ID_AAC;
		break;
	case CODEC_AC3:
		m_outputFmt->audio_codec = AV_CODEC_ID_AC3;
		break;
	case CODEC_FLAC:
		m_outputFmt->audio_codec = AV_CODEC_ID_FLAC;
		break;
	case CODEC_MP2:
		m_outputFmt->audio_codec = AV_CODEC_ID_MP2;
		break;
	case CODEC_MP3:
		m_outputFmt->audio_codec = AV_CODEC_ID_MP3;
		break;
	case CODEC_OPUS:
		m_outputFmt->audio_codec = AV_CODEC_ID_OPUS;
		break;
	case CODEC_PCM:
		switch(specs.format)
		{
		case FORMAT_U8:
			m_outputFmt->audio_codec = AV_CODEC_ID_PCM_U8;
			break;
		case FORMAT_S16:
			m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S16LE;
			break;
		case FORMAT_S24:
			m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S24LE;
			break;
		case FORMAT_S32:
			m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S32LE;
			break;
		case FORMAT_FLOAT32:
			m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F32LE;
			break;
		case FORMAT_FLOAT64:
			m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F64LE;
			break;
		default:
			m_outputFmt->audio_codec = AV_CODEC_ID_NONE;
			break;
		}
		break;
	case CODEC_VORBIS:
		m_outputFmt->audio_codec = AV_CODEC_ID_VORBIS;
		break;
	default:
		m_outputFmt->audio_codec = AV_CODEC_ID_NONE;
		break;
	}

	try
	{
		if(m_outputFmt->audio_codec == AV_CODEC_ID_NONE)
			AUD_THROW(FileException, "File couldn't be written, audio codec not found with ffmpeg.");

		AVCodec* codec = avcodec_find_encoder(m_outputFmt->audio_codec);
		if(!codec)
			AUD_THROW(FileException, "File couldn't be written, audio encoder couldn't be found with ffmpeg.");

		m_stream = avformat_new_stream(m_formatCtx, codec);
		if(!m_stream)
			AUD_THROW(FileException, "File couldn't be written, stream creation failed with ffmpeg.");

		m_stream->id = m_formatCtx->nb_streams - 1;

		m_codecCtx = m_stream->codec;

		switch(m_specs.format)
		{
		case FORMAT_U8:
			m_convert = convert_float_u8;
			m_codecCtx->sample_fmt = AV_SAMPLE_FMT_U8;
			break;
		case FORMAT_S16:
			m_convert = convert_float_s16;
			m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
			break;
		case FORMAT_S32:
			m_convert = convert_float_s32;
			m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S32;
			break;
		case FORMAT_FLOAT64:
			m_convert = convert_float_double;
			m_codecCtx->sample_fmt = AV_SAMPLE_FMT_DBL;
			break;
		default:
			m_convert = convert_copy<sample_t>;
			m_codecCtx->sample_fmt = AV_SAMPLE_FMT_FLT;
			break;
		}

		if(m_formatCtx->oformat->flags & AVFMT_GLOBALHEADER)
			m_codecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;

		bool format_supported = false;

		for(int i = 0; codec->sample_fmts[i] != -1; i++)
		{
			if(av_get_alt_sample_fmt(codec->sample_fmts[i], false) == m_codecCtx->sample_fmt)
			{
				m_deinterleave = av_sample_fmt_is_planar(codec->sample_fmts[i]);
				m_codecCtx->sample_fmt = codec->sample_fmts[i];
				format_supported = true;
			}
		}

		if(!format_supported)
		{
			int chosen_index = 0;
			auto chosen = av_get_alt_sample_fmt(codec->sample_fmts[chosen_index], false);
			for(int i = 1; codec->sample_fmts[i] != -1; i++)
			{
				auto fmt = av_get_alt_sample_fmt(codec->sample_fmts[i], false);
				if((fmt > chosen && chosen < m_codecCtx->sample_fmt) || (fmt > m_codecCtx->sample_fmt && fmt < chosen))
				{
					chosen = fmt;
					chosen_index = i;
				}
			}

			m_codecCtx->sample_fmt = codec->sample_fmts[chosen_index];
			m_deinterleave = av_sample_fmt_is_planar(m_codecCtx->sample_fmt);
			switch(av_get_alt_sample_fmt(m_codecCtx->sample_fmt, false))
			{
			case AV_SAMPLE_FMT_U8:
				specs.format = FORMAT_U8;
				m_convert = convert_float_u8;
				break;
			case AV_SAMPLE_FMT_S16:
				specs.format = FORMAT_S16;
				m_convert = convert_float_s16;
				break;
			case AV_SAMPLE_FMT_S32:
				specs.format = FORMAT_S32;
				m_convert = convert_float_s32;
				break;
			case AV_SAMPLE_FMT_FLT:
				specs.format = FORMAT_FLOAT32;
				m_convert = convert_copy<sample_t>;
				break;
			case AV_SAMPLE_FMT_DBL:
				specs.format = FORMAT_FLOAT64;
				m_convert = convert_float_double;
				break;
			default:
				AUD_THROW(FileException, "File couldn't be written, sample format not supported with ffmpeg.");
			}
		}

		m_codecCtx->sample_rate = 0;

		if(codec->supported_samplerates)
		{
			for(int i = 0; codec->supported_samplerates[i]; i++)
			{
				if(codec->supported_samplerates[i] == m_specs.rate)
				{
					m_codecCtx->sample_rate = codec->supported_samplerates[i];
					break;
				}
				else if((codec->supported_samplerates[i] > m_codecCtx->sample_rate && m_specs.rate > m_codecCtx->sample_rate) ||
						(codec->supported_samplerates[i] < m_codecCtx->sample_rate && m_specs.rate < codec->supported_samplerates[i]))
				{
					m_codecCtx->sample_rate = codec->supported_samplerates[i];
				}
			}
		}

		if(m_codecCtx->sample_rate == 0)
			m_codecCtx->sample_rate = m_specs.rate;

		m_specs.rate = m_codecCtx->sample_rate;

		m_codecCtx->codec_id = m_outputFmt->audio_codec;
		m_codecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
		m_codecCtx->bit_rate = bitrate;
		m_codecCtx->channels = m_specs.channels;
		m_stream->time_base.num = m_codecCtx->time_base.num = 1;
		m_stream->time_base.den = m_codecCtx->time_base.den = m_codecCtx->sample_rate;

		if(avcodec_open2(m_codecCtx, codec, nullptr) < 0)
			AUD_THROW(FileException, "File couldn't be written, encoder couldn't be opened with ffmpeg.");

		int samplesize = std::max(int(AUD_SAMPLE_SIZE(m_specs)), AUD_DEVICE_SAMPLE_SIZE(m_specs));

		if((m_input_size = m_codecCtx->frame_size))
			m_input_buffer.resize(m_input_size * samplesize);

		if(avio_open(&m_formatCtx->pb, filename.c_str(), AVIO_FLAG_WRITE))
			AUD_THROW(FileException, "File couldn't be written, file opening failed with ffmpeg.");

		avformat_write_header(m_formatCtx, nullptr);
	}
	catch(Exception&)
	{
		avformat_free_context(m_formatCtx);
		throw;
	}
}