AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
- UInt32 inNumberFrames,
+ UInt32 inNumberSamples,
AudioBufferList* ioData)
{
if (userData) {
- return ((AUPlugin*)userData)->render_callback (ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, ioData);
+ return ((AUPlugin*)userData)->render_callback (ioActionFlags, inTimeStamp, inBusNumber, inNumberSamples, ioData);
}
return paramErr;
}
, cb_offsets (0)
, input_buffers (0)
, input_map (0)
- , frames_processed (0)
+ , samples_processed (0)
, audio_input_cnt (0)
, _parameter_listener (0)
, _parameter_listener_arg (0)
- , transport_frame (0)
+ , transport_sample (0)
, transport_speed (0)
, last_transport_speed (0.0)
{
, cb_offsets (0)
, input_buffers (0)
, input_map (0)
- , frames_processed (0)
+ , samples_processed (0)
, _parameter_listener (0)
, _parameter_listener_arg (0)
- , transport_frame (0)
+ , transport_sample (0)
, transport_speed (0)
, last_transport_speed (0.0)
DEBUG_TRACE (DEBUG::AudioUnits, "count output elements\n");
unit->GetElementCount (kAudioUnitScope_Output, output_elements);
- cb_offsets = (framecnt_t*) calloc (input_elements, sizeof(framecnt_t));
+ cb_offsets = (samplecnt_t*) calloc (input_elements, sizeof(samplecnt_t));
bus_inputs = (uint32_t*) calloc (input_elements, sizeof(uint32_t));
bus_outputs = (uint32_t*) calloc (output_elements, sizeof(uint32_t));
kAudioUnitParameterUnit_Boolean = 2
kAudioUnitParameterUnit_Percent = 3
kAudioUnitParameterUnit_Seconds = 4
- kAudioUnitParameterUnit_SampleFrames = 5
+ kAudioUnitParameterUnit_SampleSamples = 5
kAudioUnitParameterUnit_Phase = 6
kAudioUnitParameterUnit_Rate = 7
kAudioUnitParameterUnit_Hertz = 8
d.integer_step = (info.unit == kAudioUnitParameterUnit_Indexed);
d.toggled = (info.unit == kAudioUnitParameterUnit_Boolean) ||
(d.integer_step && ((d.upper - d.lower) == 1.0));
- d.sr_dependent = (info.unit == kAudioUnitParameterUnit_SampleFrames);
+ d.sr_dependent = (info.unit == kAudioUnitParameterUnit_SampleSamples);
d.automatable = /* !d.toggled && -- ardour can automate toggles, can AU ? */
!(info.flags & kAudioUnitParameterFlag_NonRealTime) &&
(info.flags & kAudioUnitParameterFlag_IsWritable);
return 0;
}
-framecnt_t
+samplecnt_t
AUPlugin::signal_latency () const
{
guint lat = g_atomic_int_get (&_current_latency);;
if (lat == UINT_MAX) {
- lat = unit->Latency() * _session.frame_rate();
+ lat = unit->Latency() * _session.sample_rate();
g_atomic_int_set (&_current_latency, lat);
}
return lat;
if ((err = unit->Initialize()) != noErr) {
error << string_compose (_("AUPlugin: %1 cannot initialize plugin (err = %2)"), name(), err) << endmsg;
} else {
- frames_processed = 0;
+ samples_processed = 0;
initialized = true;
}
}
AUPlugin::set_block_size (pframes_t nframes)
{
bool was_initialized = initialized;
- UInt32 numFrames = nframes;
+ UInt32 numSamples = nframes;
OSErr err;
if (initialized) {
deactivate ();
}
- DEBUG_TRACE (DEBUG::AudioUnits, string_compose ("set MaximumFramesPerSlice in global scope to %1\n", numFrames));
- if ((err = unit->SetProperty (kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global,
- 0, &numFrames, sizeof (numFrames))) != noErr) {
- error << string_compose (_("AU: cannot set max frames (err = %1)"), err) << endmsg;
+ DEBUG_TRACE (DEBUG::AudioUnits, string_compose ("set MaximumSamplesPerSlice in global scope to %1\n", numSamples));
+ if ((err = unit->SetProperty (kAudioUnitProperty_MaximumSamplesPerSlice, kAudioUnitScope_Global,
+ 0, &numSamples, sizeof (numSamples))) != noErr) {
+ error << string_compose (_("AU: cannot set max samples (err = %1)"), err) << endmsg;
return -1;
}
}
}
- streamFormat.mSampleRate = _session.frame_rate();
+ streamFormat.mSampleRate = _session.sample_rate();
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked|kAudioFormatFlagIsNonInterleaved;
#endif
streamFormat.mBitsPerChannel = 32;
- streamFormat.mFramesPerPacket = 1;
+ streamFormat.mSamplesPerPacket = 1;
/* apple says that for non-interleaved data, these
* values always refer to a single channel.
AUPlugin::render_callback(AudioUnitRenderActionFlags*,
const AudioTimeStamp*,
UInt32 bus,
- UInt32 inNumberFrames,
+ UInt32 inNumberSamples,
AudioBufferList* ioData)
{
/* not much to do with audio - the data is already in the buffers given to us in connect_and_run() */
- // DEBUG_TRACE (DEBUG::AudioUnits, string_compose ("%1: render callback, frames %2 bus %3 bufs %4\n",
- // name(), inNumberFrames, bus, ioData->mNumberBuffers));
+ // DEBUG_TRACE (DEBUG::AudioUnits, string_compose ("%1: render callback, samples %2 bus %3 bufs %4\n",
+ // name(), inNumberSamples, bus, ioData->mNumberBuffers));
if (input_maxbuf == 0) {
DEBUG_TRACE (DEBUG::AudioUnits, "AUPlugin: render callback called illegally!");
for (uint32_t i = 0; i < limit; ++i) {
ioData->mBuffers[i].mNumberChannels = 1;
- ioData->mBuffers[i].mDataByteSize = sizeof (Sample) * inNumberFrames;
+ ioData->mBuffers[i].mDataByteSize = sizeof (Sample) * inNumberSamples;
bool valid = false;
uint32_t idx = input_map->get (DataType::AUDIO, i + busoff, &valid);
ioData->mBuffers[i].mData = silent_bufs.get_audio(0).data (cb_offsets[bus] + input_offset);
}
}
- cb_offsets[bus] += inNumberFrames;
+ cb_offsets[bus] += inNumberSamples;
return noErr;
}
int
AUPlugin::connect_and_run (BufferSet& bufs,
- framepos_t start, framepos_t end, double speed,
+ samplepos_t start, samplepos_t end, double speed,
ChanMapping in_map, ChanMapping out_map,
- pframes_t nframes, framecnt_t offset)
+ pframes_t nframes, samplecnt_t offset)
{
Plugin::connect_and_run(bufs, start, end, speed, in_map, out_map, nframes, offset);
- transport_frame = start;
+ transport_sample = start;
transport_speed = speed;
AudioUnitRenderActionFlags flags = 0;
/* one MIDI port/buffer only */
MidiBuffer& m = bufs.get_midi (i);
for (MidiBuffer::iterator i = m.begin(); i != m.end(); ++i) {
- Evoral::Event<framepos_t> ev (*i);
+ Evoral::Event<samplepos_t> ev (*i);
if (ev.is_channel_event()) {
const uint8_t* b = ev.buffer();
DEBUG_TRACE (DEBUG::AudioUnits, string_compose ("%1: MIDI event %2\n", name(), ev));
}
/* does this really mean anything ? */
- ts.mSampleTime = frames_processed;
+ ts.mSampleTime = samples_processed;
ts.mFlags = kAudioTimeStampSampleTimeValid;
DEBUG_TRACE (DEBUG::AudioUnits, string_compose ("%1 render flags=%2 time=%3 nframes=%4 bus=%5 buffers=%6\n",
- name(), flags, frames_processed, nframes, bus, buffers->mNumberBuffers));
+ name(), flags, samples_processed, nframes, bus, buffers->mNumberBuffers));
if ((err = unit->Render (&flags, &ts, bus, nframes, buffers)) == noErr) {
input_maxbuf = 0;
if (ok) {
- frames_processed += nframes;
+ samples_processed += nframes;
return 0;
}
return -1;
DEBUG_TRACE (DEBUG::AudioUnits, "AU calls ardour beat&tempo callback\n");
if (outCurrentBeat) {
- *outCurrentBeat = tmap.quarter_note_at_frame (transport_frame + input_offset);
+ *outCurrentBeat = tmap.quarter_note_at_sample (transport_sample + input_offset);
}
if (outCurrentTempo) {
- *outCurrentTempo = tmap.tempo_at_frame (transport_frame + input_offset).quarter_notes_per_minute();
+ *outCurrentTempo = tmap.tempo_at_sample (transport_sample + input_offset).quarter_notes_per_minute();
}
return noErr;
DEBUG_TRACE (DEBUG::AudioUnits, "AU calls ardour music time location callback\n");
- TempoMetric metric = tmap.metric_at (transport_frame + input_offset);
- Timecode::BBT_Time bbt = _session.tempo_map().bbt_at_frame (transport_frame + input_offset);
+ TempoMetric metric = tmap.metric_at (transport_sample + input_offset);
+ Timecode::BBT_Time bbt = _session.tempo_map().bbt_at_sample (transport_sample + input_offset);
if (outDeltaSampleOffsetToNextBeat) {
if (bbt.ticks == 0) {
/* on the beat */
*outDeltaSampleOffsetToNextBeat = 0;
} else {
- double const next_beat = ceil (tmap.quarter_note_at_frame (transport_frame + input_offset));
- framepos_t const next_beat_frame = tmap.frame_at_quarter_note (next_beat);
+ double const next_beat = ceil (tmap.quarter_note_at_sample (transport_sample + input_offset));
+ samplepos_t const next_beat_sample = tmap.sample_at_quarter_note (next_beat);
- *outDeltaSampleOffsetToNextBeat = next_beat_frame - (transport_frame + input_offset);
+ *outDeltaSampleOffsetToNextBeat = next_beat_sample - (transport_sample + input_offset);
}
}
/* this assumes that the AU can only call this host callback from render context,
where input_offset is valid.
*/
- *outCurrentSampleInTimeLine = transport_frame + input_offset;
+ *outCurrentSampleInTimeLine = transport_sample + input_offset;
}
if (outIsCycling) {
Timecode::BBT_Time bbt;
if (outCycleStartBeat) {
- *outCycleStartBeat = tmap.quarter_note_at_frame (loc->start() + input_offset);
+ *outCycleStartBeat = tmap.quarter_note_at_sample (loc->start() + input_offset);
}
if (outCycleEndBeat) {
- *outCycleEndBeat = tmap.quarter_note_at_frame (loc->end() + input_offset);
+ *outCycleEndBeat = tmap.quarter_note_at_sample (loc->end() + input_offset);
}
}
}
if (event->mEventType == kAudioUnitEvent_PropertyChange) {
if (event->mArgument.mProperty.mPropertyID == kAudioUnitProperty_Latency) {
DEBUG_TRACE (DEBUG::AudioUnits, string_compose("AU Latency Change Event %1 <> %2\n", new_value, unit->Latency()));
- guint lat = unit->Latency() * _session.frame_rate();
+ guint lat = unit->Latency() * _session.sample_rate();
g_atomic_int_set (&_current_latency, lat);
}
return;
projects / ardour.git / blobdiff