bool AUD_OpenALDevice::AUD_OpenALHandle::seek(float position)
{
if(!m_status)
return false;
m_device->lock();
if(m_isBuffered)
alSourcef(m_source, AL_SEC_OFFSET, position);
else
{
m_reader->seek((int)(position * m_reader->getSpecs().rate));
m_eos = false;
ALint info;
alGetSourcei(m_source, AL_SOURCE_STATE, &info);
if(info != AL_PLAYING)
{
if(info == AL_PAUSED)
alSourceStop(m_source);
alSourcei(m_source, AL_BUFFER, 0);
m_current = 0;
ALenum err;
if((err = alGetError()) == AL_NO_ERROR)
{
int length;
AUD_DeviceSpecs specs = m_device->m_specs;
specs.specs = m_reader->getSpecs();
m_device->m_buffer.assureSize(m_device->m_buffersize * AUD_DEVICE_SAMPLE_SIZE(specs));
for(int i = 0; i < CYCLE_BUFFERS; i++)
{
length = m_device->m_buffersize;
m_reader->read(length, m_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)
break;
}
if(m_loopcount != 0)
m_eos = false;
alSourceQueueBuffers(m_source, CYCLE_BUFFERS, m_buffers);
}
alSourceRewind(m_source);
}
}
m_device->unlock();
return true;
}
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);
}
bool AUD_ReadDevice::read(data_t* buffer, int length)
{
if(m_playing)
mix(buffer, length);
else
if(m_specs.format == AUD_FORMAT_U8)
memset(buffer, 0x80, length * AUD_DEVICE_SAMPLE_SIZE(m_specs));
else
memset(buffer, 0, length * AUD_DEVICE_SAMPLE_SIZE(m_specs));
return m_playing;
}
AUD_FFMPEGWriter::~AUD_FFMPEGWriter()
{
// writte missing data
if(m_input_samples)
{
sample_t* buf = m_input_buffer.getBuffer();
memset(buf + m_specs.channels * m_input_samples, 0,
(m_input_size - m_input_samples) * AUD_DEVICE_SAMPLE_SIZE(m_specs));
encode(buf);
}
av_write_trailer(m_formatCtx);
avcodec_close(m_codecCtx);
av_freep(&m_formatCtx->streams[0]->codec);
av_freep(&m_formatCtx->streams[0]);
#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
av_frame_free(&m_frame);
#endif
avio_close(m_formatCtx->pb);
av_free(m_formatCtx);
}
OpenALDevice::OpenALHandle::OpenALHandle(OpenALDevice* device, ALenum format, std::shared_ptr<IReader> reader, bool keep) :
m_isBuffered(false), m_reader(reader), m_keep(keep), m_format(format),
m_eos(false), m_loopcount(0), m_stop(nullptr), m_stop_data(nullptr), m_status(STATUS_PLAYING),
m_device(device)
{
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(DeviceException, "Buffer generation failed while staring playback with OpenAL.");
try
{
m_device->m_buffer.assureSize(m_device->m_buffersize * AUD_DEVICE_SAMPLE_SIZE(specs));
int length;
bool eos;
for(m_current = 0; m_current < CYCLE_BUFFERS; m_current++)
{
length = m_device->m_buffersize;
reader->read(length, eos, m_device->m_buffer.getBuffer());
if(length == 0)
break;
alBufferData(m_buffers[m_current], m_format, m_device->m_buffer.getBuffer(), length * AUD_DEVICE_SAMPLE_SIZE(specs), specs.rate);
if(alGetError() != AL_NO_ERROR)
AUD_THROW(DeviceException, "Filling the buffer with data failed while starting playback with OpenAL.");
}
alGenSources(1, &m_source);
if(alGetError() != AL_NO_ERROR)
AUD_THROW(DeviceException, "Source generation failed while starting playback with OpenAL.");
try
{
alSourceQueueBuffers(m_source, m_current, m_buffers);
if(alGetError() != AL_NO_ERROR)
AUD_THROW(DeviceException, "Buffer queuing failed while starting playback with OpenAL.");
}
catch(Exception&)
{
alDeleteSources(1, &m_source);
throw;
}
}
catch(Exception&)
{
alDeleteBuffers(CYCLE_BUFFERS, m_buffers);
throw;
}
alSourcei(m_source, AL_SOURCE_RELATIVE, 1);
}
void AUD_FFMPEGReader::read(int& length, bool& eos, sample_t* buffer)
{
// read packages and decode them
AVPacket packet;
int data_size = 0;
int pkgbuf_pos;
int left = length;
int sample_size = AUD_DEVICE_SAMPLE_SIZE(m_specs);
sample_t* buf = buffer;
pkgbuf_pos = m_pkgbuf_left;
m_pkgbuf_left = 0;
// there may still be data in the buffer from the last call
if(pkgbuf_pos > 0)
{
data_size = AUD_MIN(pkgbuf_pos, left * sample_size);
m_convert((data_t*) buf, (data_t*) m_pkgbuf.getBuffer(),
data_size / AUD_FORMAT_SIZE(m_specs.format));
buf += data_size / AUD_FORMAT_SIZE(m_specs.format);
left -= data_size/sample_size;
}
// for each frame read as long as there isn't enough data already
while((left > 0) && (av_read_frame(m_formatCtx, &packet) >= 0))
{
// is it a frame from the audio stream?
if(packet.stream_index == m_stream)
{
// decode the package
pkgbuf_pos = decode(packet, m_pkgbuf);
// copy to output buffer
data_size = AUD_MIN(pkgbuf_pos, left * sample_size);
m_convert((data_t*) buf, (data_t*) m_pkgbuf.getBuffer(),
data_size / AUD_FORMAT_SIZE(m_specs.format));
buf += data_size / AUD_FORMAT_SIZE(m_specs.format);
left -= data_size/sample_size;
}
av_free_packet(&packet);
}
// read more data than necessary?
if(pkgbuf_pos > data_size)
{
m_pkgbuf_left = pkgbuf_pos-data_size;
memmove(m_pkgbuf.getBuffer(),
((data_t*)m_pkgbuf.getBuffer())+data_size,
pkgbuf_pos-data_size);
}
if((eos = (left > 0)))
length -= left;
m_position += length;
}
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;
}
}
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;
}
}
void AUD_OpenALDevice::updateStreams()
{
AUD_Reference<AUD_OpenALHandle> sound;
int length;
ALint info;
AUD_DeviceSpecs specs = m_specs;
ALCenum cerr;
std::list<AUD_Reference<AUD_OpenALHandle> > stopSounds;
std::list<AUD_Reference<AUD_OpenALHandle> > pauseSounds;
AUD_HandleIterator it;
while(1)
{
lock();
alcSuspendContext(m_context);
cerr = alcGetError(m_device);
if(cerr == ALC_NO_ERROR)
{
// for all sounds
for(it = m_playingSounds.begin(); it != m_playingSounds.end(); it++)
{
sound = *it;
// is it a streamed sound?
if(!sound->m_isBuffered)
{
// check for buffer refilling
alGetSourcei(sound->m_source, AL_BUFFERS_PROCESSED, &info);
if(info)
{
specs.specs = sound->m_reader->getSpecs();
m_buffer.assureSize(m_buffersize * AUD_DEVICE_SAMPLE_SIZE(specs));
// for all empty buffers
while(info--)
{
// if there's still data to play back
if(!sound->m_eos)
{
// read data
length = m_buffersize;
sound->m_reader->read(length, sound->m_eos, m_buffer.getBuffer());
// looping necessary?
if(length == 0 && sound->m_loopcount)
{
if(sound->m_loopcount > 0)
sound->m_loopcount--;
sound->m_reader->seek(0);
length = m_buffersize;
sound->m_reader->read(length, sound->m_eos, m_buffer.getBuffer());
}
if(sound->m_loopcount != 0)
sound->m_eos = false;
// read nothing?
if(length == 0)
{
break;
}
// unqueue buffer
alSourceUnqueueBuffers(sound->m_source, 1,
&sound->m_buffers[sound->m_current]);
ALenum err;
if((err = alGetError()) != AL_NO_ERROR)
{
sound->m_eos = true;
break;
}
// fill with new data
alBufferData(sound->m_buffers[sound->m_current],
sound->m_format,
m_buffer.getBuffer(), length *
AUD_DEVICE_SAMPLE_SIZE(specs),
specs.rate);
if((err = alGetError()) != AL_NO_ERROR)
{
sound->m_eos = true;
break;
}
// and queue again
alSourceQueueBuffers(sound->m_source, 1,
&sound->m_buffers[sound->m_current]);
if(alGetError() != AL_NO_ERROR)
{
sound->m_eos = true;
break;
}
sound->m_current = (sound->m_current+1) %
AUD_OpenALHandle::CYCLE_BUFFERS;
}
else
break;
}
}
}
// check if the sound has been stopped
alGetSourcei(sound->m_source, AL_SOURCE_STATE, &info);
if(info != AL_PLAYING)
{
// if it really stopped
if(sound->m_eos)
{
if(sound->m_stop)
sound->m_stop(sound->m_stop_data);
// pause or
if(sound->m_keep)
pauseSounds.push_back(sound);
// stop
else
stopSounds.push_back(sound);
}
// continue playing
else
alSourcePlay(sound->m_source);
}
}
for(it = pauseSounds.begin(); it != pauseSounds.end(); it++)
(*it)->pause();
for(it = stopSounds.begin(); it != stopSounds.end(); it++)
(*it)->stop();
pauseSounds.clear();
stopSounds.clear();
alcProcessContext(m_context);
}
// stop thread
if(m_playingSounds.empty() || (cerr != ALC_NO_ERROR))
{
m_playing = false;
unlock();
pthread_exit(NULL);
}
unlock();
#ifdef WIN32
Sleep(20);
#else
usleep(20000);
#endif
}
}
void AUD_SDLDevice::SDL_mix(void *data, Uint8* buffer, int length)
{
AUD_SDLDevice* device = (AUD_SDLDevice*)data;
device->mix((data_t*)buffer,length/AUD_DEVICE_SAMPLE_SIZE(device->m_specs));
}
void OpenALDevice::updateStreams()
{
int length;
ALint info;
DeviceSpecs specs = m_specs;
ALCenum cerr;
std::list<std::shared_ptr<OpenALHandle> > stopSounds;
std::list<std::shared_ptr<OpenALHandle> > pauseSounds;
auto sleepDuration = std::chrono::milliseconds(20);
for(;;)
{
lock();
alcSuspendContext(m_context);
cerr = alcGetError(m_device);
if(cerr == ALC_NO_ERROR)
{
// for all sounds
for(auto& sound : m_playingSounds)
{
// is it a streamed sound?
if(!sound->m_isBuffered)
{
// check for buffer refilling
alGetSourcei(sound->m_source, AL_BUFFERS_PROCESSED, &info);
info += (OpenALHandle::CYCLE_BUFFERS - sound->m_current);
if(info)
{
specs.specs = sound->m_reader->getSpecs();
m_buffer.assureSize(m_buffersize * AUD_DEVICE_SAMPLE_SIZE(specs));
// for all empty buffers
while(info--)
{
// if there's still data to play back
if(!sound->m_eos)
{
// read data
length = m_buffersize;
try
{
sound->m_reader->read(length, sound->m_eos, m_buffer.getBuffer());
// looping necessary?
if(length == 0 && sound->m_loopcount)
{
if(sound->m_loopcount > 0)
sound->m_loopcount--;
sound->m_reader->seek(0);
length = m_buffersize;
sound->m_reader->read(length, sound->m_eos, m_buffer.getBuffer());
}
}
catch(Exception& e)
{
length = 0;
std::cerr << "Caught exception while reading sound data during playback with OpenAL: " << e.getMessage() << std::endl;
}
if(sound->m_loopcount != 0)
sound->m_eos = false;
// read nothing?
if(length == 0)
{
break;
}
ALuint buffer;
if(sound->m_current < OpenALHandle::CYCLE_BUFFERS)
buffer = sound->m_buffers[sound->m_current++];
else
alSourceUnqueueBuffers(sound->m_source, 1, &buffer);
ALenum err;
if((err = alGetError()) != AL_NO_ERROR)
{
sound->m_eos = true;
break;
}
// fill with new data
alBufferData(buffer, sound->m_format, m_buffer.getBuffer(), length * AUD_DEVICE_SAMPLE_SIZE(specs), specs.rate);
if((err = alGetError()) != AL_NO_ERROR)
{
sound->m_eos = true;
break;
}
// and queue again
alSourceQueueBuffers(sound->m_source, 1,&buffer);
if(alGetError() != AL_NO_ERROR)
{
sound->m_eos = true;
break;
}
}
else
break;
}
}
}
// check if the sound has been stopped
alGetSourcei(sound->m_source, AL_SOURCE_STATE, &info);
if(info != AL_PLAYING)
{
// if it really stopped
if(sound->m_eos && info != AL_INITIAL)
{
if(sound->m_stop)
sound->m_stop(sound->m_stop_data);
// pause or
if(sound->m_keep)
pauseSounds.push_back(sound);
// stop
else
stopSounds.push_back(sound);
}
// continue playing
else
alSourcePlay(sound->m_source);
}
}
for(auto& sound : pauseSounds)
sound->pause(true);
for(auto& sound : stopSounds)
sound->stop();
pauseSounds.clear();
stopSounds.clear();
alcProcessContext(m_context);
}
// stop thread
if(m_playingSounds.empty() || (cerr != ALC_NO_ERROR))
{
m_playing = false;
unlock();
return;
}
unlock();
std::this_thread::sleep_for(sleepDuration);
}
}
bool OpenALDevice::OpenALHandle::seek(float position)
{
if(!m_status)
return false;
std::lock_guard<ILockable> lock(*m_device);
if(!m_status)
return false;
if(m_isBuffered)
alSourcef(m_source, AL_SEC_OFFSET, position);
else
{
m_reader->seek((int)(position * m_reader->getSpecs().rate));
m_eos = false;
ALint info;
alGetSourcei(m_source, AL_SOURCE_STATE, &info);
// we need to stop playing sounds as well to clear the buffers
// this might cause clicks, but fixes a bug regarding position determination
if(info == AL_PAUSED || info == AL_PLAYING)
alSourceStop(m_source);
alSourcei(m_source, AL_BUFFER, 0);
ALenum err;
if((err = alGetError()) == AL_NO_ERROR)
{
int length;
DeviceSpecs specs = m_device->m_specs;
specs.specs = m_reader->getSpecs();
m_device->m_buffer.assureSize(m_device->m_buffersize * AUD_DEVICE_SAMPLE_SIZE(specs));
for(m_current = 0; m_current < CYCLE_BUFFERS; m_current++)
{
length = m_device->m_buffersize;
m_reader->read(length, m_eos, m_device->m_buffer.getBuffer());
if(length == 0)
break;
alBufferData(m_buffers[m_current], m_format, m_device->m_buffer.getBuffer(), length * AUD_DEVICE_SAMPLE_SIZE(specs), specs.rate);
if(alGetError() != AL_NO_ERROR)
break;
}
if(m_loopcount != 0)
m_eos = false;
alSourceQueueBuffers(m_source, m_current, m_buffers);
}
alSourceRewind(m_source);
}
if(m_status == STATUS_STOPPED)
m_status = STATUS_PAUSED;
return true;
}
AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs, AUD_Container format, AUD_Codec codec, unsigned int bitrate) :
m_position(0),
m_specs(specs),
m_input_samples(0)
{
static const char* formats[] = { NULL, "ac3", "flac", "matroska", "mp2", "mp3", "ogg", "wav" };
m_formatCtx = avformat_alloc_context();
if (!m_formatCtx) AUD_THROW(AUD_ERROR_FFMPEG, context_error);
strcpy(m_formatCtx->filename, filename.c_str());
m_outputFmt = m_formatCtx->oformat = av_guess_format(formats[format], filename.c_str(), NULL);
if (!m_outputFmt) {
avformat_free_context(m_formatCtx);
AUD_THROW(AUD_ERROR_FFMPEG, context_error);
}
switch(codec)
{
case AUD_CODEC_AAC:
m_outputFmt->audio_codec = AV_CODEC_ID_AAC;
break;
case AUD_CODEC_AC3:
m_outputFmt->audio_codec = AV_CODEC_ID_AC3;
break;
case AUD_CODEC_FLAC:
m_outputFmt->audio_codec = AV_CODEC_ID_FLAC;
break;
case AUD_CODEC_MP2:
m_outputFmt->audio_codec = AV_CODEC_ID_MP2;
break;
case AUD_CODEC_MP3:
m_outputFmt->audio_codec = AV_CODEC_ID_MP3;
break;
case AUD_CODEC_PCM:
switch(specs.format)
{
case AUD_FORMAT_U8:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_U8;
break;
case AUD_FORMAT_S16:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S16LE;
break;
case AUD_FORMAT_S24:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S24LE;
break;
case AUD_FORMAT_S32:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_S32LE;
break;
case AUD_FORMAT_FLOAT32:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F32LE;
break;
case AUD_FORMAT_FLOAT64:
m_outputFmt->audio_codec = AV_CODEC_ID_PCM_F64LE;
break;
default:
m_outputFmt->audio_codec = AV_CODEC_ID_NONE;
break;
}
break;
case AUD_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(AUD_ERROR_SPECS, codec_error);
m_stream = avformat_new_stream(m_formatCtx, NULL);
if(!m_stream)
AUD_THROW(AUD_ERROR_FFMPEG, stream_error);
m_codecCtx = m_stream->codec;
m_codecCtx->codec_id = m_outputFmt->audio_codec;
m_codecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
m_codecCtx->bit_rate = bitrate;
m_codecCtx->sample_rate = int(m_specs.rate);
m_codecCtx->channels = m_specs.channels;
m_codecCtx->time_base.num = 1;
m_codecCtx->time_base.den = m_codecCtx->sample_rate;
switch(m_specs.format)
{
case AUD_FORMAT_U8:
m_convert = AUD_convert_float_u8;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_U8;
break;
case AUD_FORMAT_S16:
m_convert = AUD_convert_float_s16;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
break;
case AUD_FORMAT_S32:
m_convert = AUD_convert_float_s32;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_S32;
break;
case AUD_FORMAT_FLOAT32:
m_convert = AUD_convert_copy<float>;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_FLT;
break;
case AUD_FORMAT_FLOAT64:
m_convert = AUD_convert_float_double;
m_codecCtx->sample_fmt = AV_SAMPLE_FMT_DBL;
break;
default:
AUD_THROW(AUD_ERROR_FFMPEG, format_error);
}
try
{
if(m_formatCtx->oformat->flags & AVFMT_GLOBALHEADER)
m_codecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
AVCodec* codec = avcodec_find_encoder(m_codecCtx->codec_id);
if(!codec)
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
if(codec->sample_fmts) {
// Check if the prefered sample format for this codec is supported.
const enum AVSampleFormat *p = codec->sample_fmts;
for(; *p != -1; p++) {
if(*p == m_stream->codec->sample_fmt)
break;
}
if(*p == -1) {
// Sample format incompatible with codec. Defaulting to a format known to work.
m_stream->codec->sample_fmt = codec->sample_fmts[0];
}
}
if(avcodec_open2(m_codecCtx, codec, NULL))
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
m_output_buffer.resize(FF_MIN_BUFFER_SIZE);
int samplesize = AUD_MAX(AUD_SAMPLE_SIZE(m_specs), AUD_DEVICE_SAMPLE_SIZE(m_specs));
if(m_codecCtx->frame_size <= 1) {
m_input_size = FF_MIN_BUFFER_SIZE * 8 / m_codecCtx->bits_per_coded_sample / m_codecCtx->channels;
m_input_buffer.resize(m_input_size * samplesize);
}
else
{
m_input_buffer.resize(m_codecCtx->frame_size * samplesize);
m_input_size = m_codecCtx->frame_size;
}
#ifdef FFMPEG_HAVE_ENCODE_AUDIO2
m_frame = av_frame_alloc();
if (!m_frame)
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
avcodec_get_frame_defaults(m_frame);
m_frame->linesize[0] = m_input_size * samplesize;
m_frame->format = m_codecCtx->sample_fmt;
m_frame->nb_samples = m_input_size;
# ifdef FFMPEG_HAVE_AVFRAME_SAMPLE_RATE
m_frame->sample_rate = m_codecCtx->sample_rate;
# endif
# ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
m_frame->channel_layout = m_codecCtx->channel_layout;
# endif
m_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
m_frame_pts = 0;
m_deinterleave = av_sample_fmt_is_planar(m_codecCtx->sample_fmt);
if(m_deinterleave)
m_deinterleave_buffer.resize(m_input_size * m_codecCtx->channels * m_sample_size);
#endif
try
{
if(avio_open(&m_formatCtx->pb, filename.c_str(), AVIO_FLAG_WRITE))
AUD_THROW(AUD_ERROR_FILE, file_error);
avformat_write_header(m_formatCtx, NULL);
}
catch(AUD_Exception&)
{
avcodec_close(m_codecCtx);
av_freep(&m_formatCtx->streams[0]->codec);
throw;
}
}
catch(AUD_Exception&)
{
av_freep(&m_formatCtx->streams[0]);
throw;
}
}
catch(AUD_Exception&)
{
av_free(m_formatCtx);
throw;
}
}