status_t
SoundPlayNode::AllocateBuffers()
{
CALLED();
// allocate enough buffers to span our downstream latency, plus one
size_t size = fOutput.format.u.raw_audio.buffer_size;
int32 count = int32(fLatency / BufferDuration() + 1 + 1);
TRACE("SoundPlayNode::AllocateBuffers: latency = %Ld, buffer duration "
"= %Ld, count %ld\n", fLatency, BufferDuration(), count);
if (count < 3)
count = 3;
TRACE("SoundPlayNode::AllocateBuffers: creating group of %ld buffers, "
"size = %lu\n", count, size);
fBufferGroup = new BBufferGroup(size, count);
if (fBufferGroup->InitCheck() != B_OK) {
ERROR("SoundPlayNode::AllocateBuffers: BufferGroup::InitCheck() "
"failed\n");
}
return fBufferGroup->InitCheck();
}
void
ToneProducer::AllocateBuffers()
{
FPRINTF(stderr, "ToneProducer::AllocateBuffers\n");
// allocate enough buffers to span our downstream latency, plus one
size_t size = mOutput.format.u.raw_audio.buffer_size;
int32 count = int32(mLatency / BufferDuration() + 1 + 1);
FPRINTF(stderr, "\tlatency = %Ld, buffer duration = %Ld\n", mLatency, BufferDuration());
FPRINTF(stderr, "\tcreating group of %ld buffers, size = %lx\n", count, size);
mBufferGroup = new BBufferGroup(size, count);
}
status_t
ToneProducer::SetBufferGroup(const media_source& for_source, BBufferGroup* newGroup)
{
FPRINTF(stderr, "ToneProducer::SetBufferGroup\n");
// verify that we didn't get bogus arguments before we proceed
if (for_source != mOutput.source) return B_MEDIA_BAD_SOURCE;
// Are we being passed the buffer group we're already using?
if (newGroup == mBufferGroup) return B_OK;
// Ahh, someone wants us to use a different buffer group. At this point we delete
// the one we are using and use the specified one instead. If the specified group is
// NULL, we need to recreate one ourselves, and use *that*. Note that if we're
// caching a BBuffer that we requested earlier, we have to Recycle() that buffer
// *before* deleting the buffer group, otherwise we'll deadlock waiting for that
// buffer to be recycled!
delete mBufferGroup; // waits for all buffers to recycle
if (newGroup != NULL)
{
// we were given a valid group; just use that one from now on
mBufferGroup = newGroup;
}
else
{
// we were passed a NULL group pointer; that means we construct
// our own buffer group to use from now on
size_t size = mOutput.format.u.raw_audio.buffer_size;
int32 count = int32(mLatency / BufferDuration() + 1 + 1);
mBufferGroup = new BBufferGroup(size, count);
}
return B_OK;
}
status_t
AudioProducer::SetBufferGroup(const media_source& forSource,
BBufferGroup* newGroup)
{
TRACE("%p->AudioProducer::SetBufferGroup()\n", this);
if (forSource != fOutput.source)
return B_MEDIA_BAD_SOURCE;
if (newGroup == fBufferGroup)
return B_OK;
if (fUsingOurBuffers && fBufferGroup)
delete fBufferGroup; // waits for all buffers to recycle
if (newGroup != NULL) {
// we were given a valid group; just use that one from now on
fBufferGroup = newGroup;
fUsingOurBuffers = false;
} else {
// we were passed a NULL group pointer; that means we construct
// our own buffer group to use from now on
size_t size = fOutput.format.u.raw_audio.buffer_size;
int32 count = int32(fLatency / BufferDuration() + 1 + 1);
fBufferGroup = new BBufferGroup(size, count);
fUsingOurBuffers = true;
}
return B_OK;
}
BBuffer*
SoundPlayNode::FillNextBuffer(bigtime_t eventTime)
{
CALLED();
// get a buffer from our buffer group
BBuffer* buffer = fBufferGroup->RequestBuffer(
fOutput.format.u.raw_audio.buffer_size, BufferDuration() / 2);
// If we fail to get a buffer (for example, if the request times out), we
// skip this buffer and go on to the next, to avoid locking up the control
// thread
if (buffer == NULL) {
ERROR("SoundPlayNode::FillNextBuffer: RequestBuffer failed\n");
return NULL;
}
if (fPlayer->HasData()) {
fPlayer->PlayBuffer(buffer->Data(),
fOutput.format.u.raw_audio.buffer_size, fOutput.format.u.raw_audio);
} else
memset(buffer->Data(), 0, fOutput.format.u.raw_audio.buffer_size);
// fill in the buffer header
media_header* header = buffer->Header();
header->type = B_MEDIA_RAW_AUDIO;
header->size_used = fOutput.format.u.raw_audio.buffer_size;
header->time_source = TimeSource()->ID();
header->start_time = eventTime;
return buffer;
}
void
GameProducer::Connect(status_t error, const media_source& source, const media_destination& destination, const media_format& format, char* io_name)
{
// If something earlier failed, Connect() might still be called, but with a non-zero
// error code. When that happens we simply unreserve the connection and do
// nothing else.
if (error) {
fOutput.destination = media_destination::null;
fOutput.format = fPreferredFormat;
return;
}
// Okay, the connection has been confirmed. Record the destination and format
// that we agreed on, and report our connection name again.
fOutput.destination = destination;
fOutput.format = format;
strlcpy(io_name, fOutput.name, B_MEDIA_NAME_LENGTH);
// Now that we're connected, we can determine our downstream latency.
// Do so, then make sure we get our events early enough.
media_node_id id;
FindLatencyFor(fOutput.destination, &fLatency, &id);
if (!fBufferGroup)
fBufferSize = fOutput.format.u.raw_audio.buffer_size;
// Have to set it before latency calculating
// Use a dry run to see how long it takes me to fill a buffer of data
// The first step to setup the buffer
bigtime_t start, produceLatency;
int32 frames = int32(fBufferSize / fFrameSize);
float* data = new float[frames * 2];
// Second, fill the buffer
start = ::system_time();
for (int32 i = 0; i < frames; i++) {
data[i*2] = 0.8 * float(i/frames);
data[i*2+1] = 0.8 * float(i/frames);
}
produceLatency = ::system_time();
// Third, calculate the latency
fInternalLatency = produceLatency - start;
SetEventLatency(fLatency + fInternalLatency);
// Finaily, clean up
delete [] data;
// reset our buffer duration, etc. to avoid later calculations
bigtime_t duration = bigtime_t(1000000) * frames / bigtime_t(fOutput.format.u.raw_audio.frame_rate);
SetBufferDuration(duration);
// Set up the buffer group for our connection, as long as nobody handed us a
// buffer group (via SetBufferGroup()) prior to this.
if (!fBufferGroup) {
int32 count = int32(fLatency / BufferDuration() + 2);
fBufferGroup = new BBufferGroup(fBufferSize, count);
}
}
void AudioFilterNode::processBuffer(
BBuffer* inputBuffer,
BBuffer* outputBuffer) {
ASSERT(inputBuffer);
ASSERT(outputBuffer);
ASSERT(m_op);
// create wrapper objects
AudioBuffer input(m_input.format.u.raw_audio, inputBuffer);
AudioBuffer output(m_output.format.u.raw_audio, outputBuffer);
double sourceOffset = 0.0;
uint32 destinationOffset = 0L;
// when is the first frame due to be consumed?
bigtime_t startTime = outputBuffer->Header()->start_time;
// when is the next frame to be produced going to be consumed?
bigtime_t targetTime = startTime;
// when will the first frame of the next buffer be consumed?
bigtime_t endTime = startTime + BufferDuration();
uint32 framesRemaining = input.frames();
while(framesRemaining) {
// handle all events occurring before targetTime
// +++++
bigtime_t nextEventTime = endTime;
// look for next event occurring before endTime
// +++++
// process up to found event, if any, or to end of buffer
int64 toProcess = frames_for_duration(output.format(), nextEventTime - targetTime);
ASSERT(toProcess > 0);
uint32 processed = m_op->process(
input, output, sourceOffset, destinationOffset, (uint32)toProcess, targetTime);
if(processed < toProcess) {
// +++++ in offline mode this will have to request additional buffer(s), right?
PRINT((
"*** AudioFilterNode::processBuffer(): insufficient frames filled\n"));
}
if(toProcess > framesRemaining)
framesRemaining = 0;
else
framesRemaining -= toProcess;
// advance target time
targetTime = nextEventTime; // +++++ might this drift from the real frame offset?
}
outputBuffer->Header()->size_used = input.frames() * bytes_per_frame(m_output.format.u.raw_audio);
// PRINT(("### output size: %ld\n", outputBuffer->Header()->size_used));
}
void
ESDSinkNode::Connect(status_t error, const media_source& source, const media_destination& destination, const media_format& format, char* io_name)
{
CALLED();
node_output *channel = FindOutput(source);
// is this our output?
if (channel == NULL)
{
fprintf(stderr, "ESDSinkNode::Connect returning (cause : B_MEDIA_BAD_SOURCE)\n");
return;
}
// If something earlier failed, Connect() might still be called, but with a non-zero
// error code. When that happens we simply unreserve the connection and do
// nothing else.
if (error)
{
channel->fOutput.destination = media_destination::null;
channel->fOutput.format = channel->fPreferredFormat;
return;
}
// Okay, the connection has been confirmed. Record the destination and format
// that we agreed on, and report our connection name again.
channel->fOutput.destination = destination;
channel->fOutput.format = format;
strncpy(io_name, channel->fOutput.name, B_MEDIA_NAME_LENGTH);
// reset our buffer duration, etc. to avoid later calculations
bigtime_t duration = channel->fOutput.format.u.raw_audio.buffer_size * 10000
/ ( (channel->fOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK)
* channel->fOutput.format.u.raw_audio.channel_count)
/ ((int32)(channel->fOutput.format.u.raw_audio.frame_rate / 100));
SetBufferDuration(duration);
// Now that we're connected, we can determine our downstream latency.
// Do so, then make sure we get our events early enough.
media_node_id id;
FindLatencyFor(channel->fOutput.destination, &fLatency, &id);
PRINT(("\tdownstream latency = %Ld\n", fLatency));
fInternalLatency = BufferDuration();
PRINT(("\tbuffer-filling took %Ld usec on this machine\n", fInternalLatency));
//SetEventLatency(fLatency + fInternalLatency);
// Set up the buffer group for our connection, as long as nobody handed us a
// buffer group (via SetBufferGroup()) prior to this. That can happen, for example,
// if the consumer calls SetOutputBuffersFor() on us from within its Connected()
// method.
if (!channel->fBufferGroup)
AllocateBuffers(*channel);
// we are sure the thread is started
StartThread();
}
void
SoundPlayNode::Connect(status_t error, const media_source& source,
const media_destination& destination, const media_format& format,
char* name)
{
CALLED();
// is this our output?
if (source != fOutput.source) {
TRACE("SoundPlayNode::Connect returning\n");
return;
}
// If something earlier failed, Connect() might still be called, but with
// a non-zero error code. When that happens we simply unreserve the
// connection and do nothing else.
if (error) {
fOutput.destination = media_destination::null;
fOutput.format.type = B_MEDIA_RAW_AUDIO;
fOutput.format.u.raw_audio = media_multi_audio_format::wildcard;
return;
}
// Okay, the connection has been confirmed. Record the destination and
// format that we agreed on, and report our connection name again.
fOutput.destination = destination;
fOutput.format = format;
strcpy(name, Name());
// Now that we're connected, we can determine our downstream latency.
// Do so, then make sure we get our events early enough.
media_node_id id;
FindLatencyFor(fOutput.destination, &fLatency, &id);
TRACE("SoundPlayNode::Connect: downstream latency = %Ld\n", fLatency);
// reset our buffer duration, etc. to avoid later calculations
bigtime_t duration = ((fOutput.format.u.raw_audio.buffer_size * 1000000LL)
/ ((fOutput.format.u.raw_audio.format
& media_raw_audio_format::B_AUDIO_SIZE_MASK)
* fOutput.format.u.raw_audio.channel_count))
/ (int32)fOutput.format.u.raw_audio.frame_rate;
SetBufferDuration(duration);
TRACE("SoundPlayNode::Connect: buffer duration is %Ld\n", duration);
fInternalLatency = (3 * BufferDuration()) / 4;
TRACE("SoundPlayNode::Connect: using %Ld as internal latency\n",
fInternalLatency);
SetEventLatency(fLatency + fInternalLatency);
// Set up the buffer group for our connection, as long as nobody handed us
// a buffer group (via SetBufferGroup()) prior to this.
// That can happen, for example, if the consumer calls SetOutputBuffersFor()
// on us from within its Connected() method.
if (!fBufferGroup)
AllocateBuffers();
}
BBuffer*
GameProducer::FillNextBuffer(bigtime_t event_time)
{
// get a buffer from our buffer group
BBuffer* buf = fBufferGroup->RequestBuffer(fBufferSize, BufferDuration());
// if we fail to get a buffer (for example, if the request times out), we
// skip this buffer and go on to the next, to avoid locking up the control
// thread.
if (!buf)
return NULL;
// we need to discribe the buffer
int64 frames = int64(fBufferSize / fFrameSize);
memset(buf->Data(), 0, fBufferSize);
// now fill the buffer with data, continuing where the last buffer left off
fObject->Play(buf->Data(), frames);
// fill in the buffer header
media_header* hdr = buf->Header();
hdr->type = B_MEDIA_RAW_AUDIO;
hdr->size_used = fBufferSize;
hdr->time_source = TimeSource()->ID();
bigtime_t stamp;
if (RunMode() == B_RECORDING) {
// In B_RECORDING mode, we stamp with the capture time. We're not
// really a hardware capture node, but we simulate it by using the
// (precalculated) time at which this buffer "should" have been created.
stamp = event_time;
} else {
// okay, we're in one of the "live" performance run modes. in these
// modes, we stamp the buffer with the time at which the buffer should
// be rendered to the output, not with the capture time. fStartTime is
// the cached value of the first buffer's performance time; we calculate
// this buffer's performance time as an offset from that time, based on
// the amount of media we've created so far.
// Recalculating every buffer like this avoids accumulation of error.
stamp = fStartTime + bigtime_t(double(fFramesSent)
/ double(fOutput.format.u.raw_audio.frame_rate) * 1000000.0);
}
hdr->start_time = stamp;
return buf;
}
status_t
AudioProducer::_AllocateBuffers(const media_format& format)
{
TRACE("%p->AudioProducer::_AllocateBuffers()\n", this);
if (fBufferGroup && fUsingOurBuffers) {
delete fBufferGroup;
fBufferGroup = NULL;
}
size_t size = format.u.raw_audio.buffer_size;
int32 bufferDuration = BufferDuration();
int32 count = 0;
if (bufferDuration > 0) {
count = (int32)((fLatency + fInternalLatency)
/ bufferDuration + 2);
}
fBufferGroup = new BBufferGroup(size, count);
fUsingOurBuffers = true;
return fBufferGroup->InitCheck();
}
status_t
ESDSinkNode::SetBufferGroup(const media_source& for_source, BBufferGroup* newGroup)
{
CALLED();
node_output *channel = FindOutput(for_source);
// is this our output?
if (channel == NULL)
{
fprintf(stderr, "ESDSinkNode::SetBufferGroup returning B_MEDIA_BAD_SOURCE\n");
return B_MEDIA_BAD_SOURCE;
}
// Are we being passed the buffer group we're already using?
if (newGroup == channel->fBufferGroup) return B_OK;
// Ahh, someone wants us to use a different buffer group. At this point we delete
// the one we are using and use the specified one instead. If the specified group is
// NULL, we need to recreate one ourselves, and use *that*. Note that if we're
// caching a BBuffer that we requested earlier, we have to Recycle() that buffer
// *before* deleting the buffer group, otherwise we'll deadlock waiting for that
// buffer to be recycled!
delete channel->fBufferGroup; // waits for all buffers to recycle
if (newGroup != NULL)
{
// we were given a valid group; just use that one from now on
channel->fBufferGroup = newGroup;
}
else
{
// we were passed a NULL group pointer; that means we construct
// our own buffer group to use from now on
size_t size = channel->fOutput.format.u.raw_audio.buffer_size;
int32 count = int32(fLatency / BufferDuration() + 1 + 1);
channel->fBufferGroup = new BBufferGroup(size, count);
}
return B_OK;
}
status_t
GameProducer::SetBufferGroup(const media_source& forSource,
BBufferGroup* newGroup)
{
// verify that we didn't get bogus arguments before we proceed
if (forSource != fOutput.source)
return B_MEDIA_BAD_SOURCE;
// Are we being passed the buffer group we're already using?
if (newGroup == fBufferGroup)
return B_OK;
// Ahh, someone wants us to use a different buffer group. At this point we
// delete the one we are using and use the specified one instead. If the
// specified group is NULL, we need to recreate one ourselves, and use
// *that*. Note that if we're caching a BBuffer that we requested earlier,
// we have to Recycle() that buffer *before* deleting the buffer group,
// otherwise we'll deadlock waiting for that buffer to be recycled!
delete fBufferGroup; // waits for all buffers to recycle
if (newGroup != NULL) {
// we were given a valid group; just use that one from now on
fBufferGroup = newGroup;
// get buffer length from the first buffer
BBuffer* buffers[1];
if (newGroup->GetBufferList(1, buffers) != B_OK)
return B_BAD_VALUE;
fBufferSize = buffers[0]->SizeAvailable();
} else {
// we were passed a NULL group pointer; that means we construct
// our own buffer group to use from now on
fBufferSize = fOutput.format.u.raw_audio.buffer_size;
int32 count = int32(fLatency / BufferDuration() + 2);
fBufferGroup = new BBufferGroup(fBufferSize, count);
}
return B_OK;
}
BBuffer*
AudioProducer::_FillNextBuffer(bigtime_t eventTime)
{
BBuffer* buffer = fBufferGroup->RequestBuffer(
fOutput.format.u.raw_audio.buffer_size, BufferDuration());
if (!buffer) {
ERROR("AudioProducer::_FillNextBuffer() - no buffer\n");
return NULL;
}
size_t sampleSize = fOutput.format.u.raw_audio.format
& media_raw_audio_format::B_AUDIO_SIZE_MASK;
size_t numSamples = fOutput.format.u.raw_audio.buffer_size / sampleSize;
// number of sample in the buffer
// fill in the buffer header
media_header* header = buffer->Header();
header->type = B_MEDIA_RAW_AUDIO;
header->time_source = TimeSource()->ID();
buffer->SetSizeUsed(fOutput.format.u.raw_audio.buffer_size);
bigtime_t performanceTime = bigtime_t(double(fFramesSent)
* 1000000.0 / double(fOutput.format.u.raw_audio.frame_rate));
// fill in data from audio supplier
int64 frameCount = numSamples / fOutput.format.u.raw_audio.channel_count;
bigtime_t startTime = performanceTime;
bigtime_t endTime = bigtime_t(double(fFramesSent + frameCount)
* 1000000.0 / fOutput.format.u.raw_audio.frame_rate);
if (!fSupplier || fSupplier->InitCheck() != B_OK
|| fSupplier->GetFrames(buffer->Data(), frameCount, startTime,
endTime) != B_OK) {
ERROR("AudioProducer::_FillNextBuffer() - supplier error -> silence\n");
memset(buffer->Data(), 0, buffer->SizeUsed());
}
// stamp buffer
if (RunMode() == B_RECORDING) {
header->start_time = eventTime;
} else {
header->start_time = fStartTime + performanceTime;
}
#if DEBUG_TO_FILE
BMediaTrack* track;
if (BMediaFile* file = init_media_file(fOutput.format, &track)) {
track->WriteFrames(buffer->Data(), frameCount);
}
#endif // DEBUG_TO_FILE
if (fPeakListener
&& fOutput.format.u.raw_audio.format
== media_raw_audio_format::B_AUDIO_FLOAT) {
// TODO: extend the peak notifier for other sample formats
int32 channels = fOutput.format.u.raw_audio.channel_count;
float max[channels];
float min[channels];
for (int32 i = 0; i < channels; i++) {
max[i] = -1.0;
min[i] = 1.0;
}
float* sample = (float*)buffer->Data();
for (uint32 i = 0; i < frameCount; i++) {
for (int32 k = 0; k < channels; k++) {
if (*sample < min[k])
min[k] = *sample;
if (*sample > max[k])
max[k] = *sample;
sample++;
}
}
BMessage message(MSG_PEAK_NOTIFICATION);
for (int32 i = 0; i < channels; i++) {
float maxAbs = max_c(fabs(min[i]), fabs(max[i]));
message.AddFloat("max", maxAbs);
}
bigtime_t realTime = TimeSource()->RealTimeFor(
fStartTime + performanceTime, 0);
MessageEvent* event = new (std::nothrow) MessageEvent(realTime,
fPeakListener, message);
if (event != NULL)
EventQueue::Default().AddEvent(event);
}
return buffer;
}
/*****************************************************************************
* OSSThread: asynchronous thread used to DMA the data to the device
*****************************************************************************/
static void* OSSThread( vlc_object_t *p_this )
{
aout_instance_t * p_aout = (aout_instance_t*)p_this;
struct aout_sys_t * p_sys = p_aout->output.p_sys;
mtime_t next_date = 0;
int canc = vlc_savecancel ();
while ( vlc_object_alive (p_aout) )
{
aout_buffer_t * p_buffer = NULL;
int i_tmp, i_size;
uint8_t * p_bytes;
if ( p_aout->output.output.i_format != VLC_CODEC_SPDIFL )
{
mtime_t buffered = BufferDuration( p_aout );
/* Next buffer will be played at mdate() + buffered */
p_buffer = aout_OutputNextBuffer( p_aout, mdate() + buffered,
false );
if( p_buffer == NULL &&
buffered > ( p_aout->output.p_sys->max_buffer_duration
/ p_aout->output.p_sys->i_fragstotal ) )
{
/* If we have at least a fragment full, then we can wait a
* little and retry to get a new audio buffer instead of
* playing a blank sample */
msleep( ( p_aout->output.p_sys->max_buffer_duration
/ p_aout->output.p_sys->i_fragstotal / 2 ) );
continue;
}
}
else
{
/* emu10k1 driver does not report Buffer Duration correctly in
* passthrough mode so we have to cheat */
if( !next_date )
{
next_date = mdate();
}
else
{
mtime_t delay = next_date - mdate();
if( delay > AOUT_PTS_TOLERANCE )
{
msleep( delay / 2 );
}
}
while( vlc_object_alive (p_aout) && ! ( p_buffer =
aout_OutputNextBuffer( p_aout, next_date, true ) ) )
{
msleep( VLC_HARD_MIN_SLEEP );
next_date = mdate();
}
}
if ( p_buffer != NULL )
{
p_bytes = p_buffer->p_buffer;
i_size = p_buffer->i_buffer;
/* This is theoretical ... we'll see next iteration whether
* we're drifting */
next_date += p_buffer->i_length;
}
else
{
i_size = FRAME_SIZE / p_aout->output.output.i_frame_length
* p_aout->output.output.i_bytes_per_frame;
p_bytes = malloc( i_size );
memset( p_bytes, 0, i_size );
next_date = 0;
}
i_tmp = write( p_sys->i_fd, p_bytes, i_size );
if( i_tmp < 0 )
{
msg_Err( p_aout, "write failed (%m)" );
}
if ( p_buffer != NULL )
{
aout_BufferFree( p_buffer );
}
else
{
free( p_bytes );
}
}
vlc_restorecancel (canc);
return NULL;
}
/*****************************************************************************
* OSSThread: asynchronous thread used to DMA the data to the device
*****************************************************************************/
static void* OSSThread( void *obj )
{
audio_output_t * p_aout = (audio_output_t*)obj;
struct aout_sys_t * p_sys = p_aout->sys;
mtime_t next_date = 0;
for( ;; )
{
aout_buffer_t * p_buffer = NULL;
int canc = vlc_savecancel ();
if ( p_aout->format.i_format != VLC_CODEC_SPDIFL )
{
mtime_t buffered = BufferDuration( p_aout );
/* Next buffer will be played at mdate() + buffered */
p_buffer = aout_PacketNext( p_aout, mdate() + buffered );
if( p_buffer == NULL &&
buffered > ( p_aout->sys->max_buffer_duration
/ p_aout->sys->i_fragstotal ) )
{
vlc_restorecancel (canc);
/* If we have at least a fragment full, then we can wait a
* little and retry to get a new audio buffer instead of
* playing a blank sample */
msleep( ( p_aout->sys->max_buffer_duration
/ p_aout->sys->i_fragstotal / 2 ) );
continue;
}
}
else
{
vlc_restorecancel (canc);
/* emu10k1 driver does not report Buffer Duration correctly in
* passthrough mode so we have to cheat */
if( !next_date )
{
next_date = mdate();
}
else
{
mtime_t delay = next_date - mdate();
if( delay > AOUT_MAX_PTS_ADVANCE )
{
msleep( delay / 2 );
}
}
for( ;; )
{
canc = vlc_savecancel ();
p_buffer = aout_PacketNext( p_aout, next_date );
if ( p_buffer )
break;
vlc_restorecancel (canc);
msleep( VLC_HARD_MIN_SLEEP );
next_date = mdate();
}
}
uint8_t * p_bytes;
int i_size;
if ( p_buffer != NULL )
{
p_bytes = p_buffer->p_buffer;
i_size = p_buffer->i_buffer;
/* This is theoretical ... we'll see next iteration whether
* we're drifting */
next_date += p_buffer->i_length;
}
else
{
i_size = FRAME_SIZE / p_aout->format.i_frame_length
* p_aout->format.i_bytes_per_frame;
p_bytes = malloc( i_size );
memset( p_bytes, 0, i_size );
next_date = 0;
}
oss_thread_ctx_t ctx = {
.p_buffer = p_buffer,
.p_bytes = p_bytes,
};
vlc_cleanup_push( OSSThreadCleanup, &ctx );
vlc_restorecancel( canc );
int i_tmp = write( p_sys->i_fd, p_bytes, i_size );
if( i_tmp < 0 )
{
msg_Err( p_aout, "write failed (%m)" );
}
vlc_cleanup_run();
}
return NULL;
}
// create or discard buffer group if necessary
void AudioFilterNode::updateBufferGroup() {
status_t err;
size_t inputSize = bytes_per_frame(m_input.format.u.raw_audio);
size_t outputSize = bytes_per_frame(m_output.format.u.raw_audio);
if(m_input.source == media_source::null ||
m_output.destination == media_destination::null ||
inputSize >= outputSize) {
PRINT(("###### NO BUFFER GROUP NEEDED\n"));
// no internal buffer group needed
if(m_bufferGroup) {
// does this block? +++++
delete m_bufferGroup;
m_bufferGroup = 0;
}
return;
}
int32 bufferCount = EventLatency() / BufferDuration() + 1 + 1;
// +++++
// [e.moon 27sep99] this is a reasonable number of buffers,
// but it fails with looped file-player node in BeOS 4.5.2.
//
if(bufferCount < 5)
bufferCount = 5;
// if(bufferCount < 3)
// bufferCount = 3;
if(m_bufferGroup) {
// is the current group sufficient?
int32 curBufferCount;
err = m_bufferGroup->CountBuffers(&curBufferCount);
if(err == B_OK && curBufferCount >= bufferCount) {
BBuffer* buf = m_bufferGroup->RequestBuffer(
outputSize, -1);
if(buf) {
// yup
buf->Recycle();
return;
}
}
// nope, delete it to make way for the new one
delete m_bufferGroup;
m_bufferGroup = 0;
}
// create buffer group
PRINT((
"##### AudioFilterNode::updateBufferGroup():\n"
"##### creating %ld buffers of size %ld\n",
bufferCount, m_output.format.u.raw_audio.buffer_size));
m_bufferGroup = new BBufferGroup(
m_output.format.u.raw_audio.buffer_size,
bufferCount);
}
// how should we handle late buffers? drop them?
// notify the producer?
status_t
SoundPlayNode::SendNewBuffer(const media_timed_event* event,
bigtime_t lateness, bool realTimeEvent)
{
CALLED();
// printf("latency = %12Ld, event = %12Ld, sched = %5Ld, arrive at %12Ld, now %12Ld, current lateness %12Ld\n", EventLatency() + SchedulingLatency(), EventLatency(), SchedulingLatency(), event->event_time, TimeSource()->Now(), lateness);
// make sure we're both started *and* connected before delivering a buffer
if (RunState() != BMediaEventLooper::B_STARTED
|| fOutput.destination == media_destination::null)
return B_OK;
// The event->event_time is the time at which the buffer we are preparing
// here should arrive at it's destination. The MediaEventLooper should have
// scheduled us early enough (based on EventLatency() and the
// SchedulingLatency()) to make this possible.
// lateness is independent of EventLatency()!
if (lateness > (BufferDuration() / 3) ) {
printf("SoundPlayNode::SendNewBuffer, event scheduled much too late, "
"lateness is %Ld\n", lateness);
}
// skip buffer creation if output not enabled
if (fOutputEnabled) {
// Get the next buffer of data
BBuffer* buffer = FillNextBuffer(event->event_time);
if (buffer) {
// If we are ready way too early, decrase internal latency
/*
bigtime_t how_early = event->event_time - TimeSource()->Now() - fLatency - fInternalLatency;
if (how_early > 5000) {
printf("SoundPlayNode::SendNewBuffer, event scheduled too early, how_early is %Ld\n", how_early);
if (fTooEarlyCount++ == 5) {
fInternalLatency -= how_early;
if (fInternalLatency < 500)
fInternalLatency = 500;
printf("SoundPlayNode::SendNewBuffer setting internal latency to %Ld\n", fInternalLatency);
SetEventLatency(fLatency + fInternalLatency);
fTooEarlyCount = 0;
}
}
*/
// send the buffer downstream if and only if output is enabled
if (SendBuffer(buffer, fOutput.source, fOutput.destination)
!= B_OK) {
// we need to recycle the buffer
// if the call to SendBuffer() fails
printf("SoundPlayNode::SendNewBuffer: Buffer sending "
"failed\n");
buffer->Recycle();
}
}
}
// track how much media we've delivered so far
size_t nFrames = fOutput.format.u.raw_audio.buffer_size
/ ((fOutput.format.u.raw_audio.format
& media_raw_audio_format::B_AUDIO_SIZE_MASK)
* fOutput.format.u.raw_audio.channel_count);
fFramesSent += nFrames;
// The buffer is on its way; now schedule the next one to go
// nextEvent is the time at which the buffer should arrive at it's
// destination
bigtime_t nextEvent = fStartTime + bigtime_t((1000000LL * fFramesSent)
/ (int32)fOutput.format.u.raw_audio.frame_rate);
media_timed_event nextBufferEvent(nextEvent, SEND_NEW_BUFFER_EVENT);
EventQueue()->AddEvent(nextBufferEvent);
return B_OK;
}
BBuffer*
ToneProducer::FillNextBuffer(bigtime_t event_time)
{
// get a buffer from our buffer group
BBuffer* buf = mBufferGroup->RequestBuffer(mOutput.format.u.raw_audio.buffer_size, BufferDuration());
// if we fail to get a buffer (for example, if the request times out), we skip this
// buffer and go on to the next, to avoid locking up the control thread
if (!buf)
{
return NULL;
}
// now fill it with data, continuing where the last buffer left off
// 20sep99: multichannel support
size_t numFrames =
mOutput.format.u.raw_audio.buffer_size /
(sizeof(float)*mOutput.format.u.raw_audio.channel_count);
bool stereo = (mOutput.format.u.raw_audio.channel_count == 2);
if(!stereo) {
ASSERT(mOutput.format.u.raw_audio.channel_count == 1);
}
// PRINT(("buffer: %ld, %ld frames, %s\n", mOutput.format.u.raw_audio.buffer_size, numFrames, stereo ? "stereo" : "mono"));
float* data = (float*) buf->Data();
switch (mWaveform)
{
case SINE_WAVE:
FillSineBuffer(data, numFrames, stereo);
break;
case TRIANGLE_WAVE:
FillTriangleBuffer(data, numFrames, stereo);
break;
case SAWTOOTH_WAVE:
FillSawtoothBuffer(data, numFrames, stereo);
break;
}
// fill in the buffer header
media_header* hdr = buf->Header();
hdr->type = B_MEDIA_RAW_AUDIO;
hdr->size_used = mOutput.format.u.raw_audio.buffer_size;
hdr->time_source = TimeSource()->ID();
bigtime_t stamp;
if (RunMode() == B_RECORDING)
{
// In B_RECORDING mode, we stamp with the capture time. We're not
// really a hardware capture node, but we simulate it by using the (precalculated)
// time at which this buffer "should" have been created.
stamp = event_time;
}
else
{
// okay, we're in one of the "live" performance run modes. in these modes, we
// stamp the buffer with the time at which the buffer should be rendered to the
// output, not with the capture time. mStartTime is the cached value of the
// first buffer's performance time; we calculate this buffer's performance time as
// an offset from that time, based on the amount of media we've created so far.
// Recalculating every buffer like this avoids accumulation of error.
stamp = mStartTime + bigtime_t(double(mFramesSent) / double(mOutput.format.u.raw_audio.frame_rate) * 1000000.0);
}
hdr->start_time = stamp;
return buf;
}
