From 30b087ab3d28f1585987fa3f6ae006562ae192e3 Mon Sep 17 00:00:00 2001 From: Paul Davis Date: Mon, 18 Sep 2017 12:39:17 -0400 Subject: globally change all use of "frame" to refer to audio into "sample". Generated by tools/f2s. Some hand-editing will be required in a few places to fix up comments related to timecode and video in order to keep the legible --- libs/ardour/audio_unit.cc | 84 +++++++++++++++++++++++------------------------ 1 file changed, 42 insertions(+), 42 deletions(-) (limited to 'libs/ardour/audio_unit.cc') diff --git a/libs/ardour/audio_unit.cc b/libs/ardour/audio_unit.cc index da18b969f5..74b2591df9 100644 --- a/libs/ardour/audio_unit.cc +++ b/libs/ardour/audio_unit.cc @@ -163,11 +163,11 @@ _render_callback(void *userData, 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; } @@ -445,11 +445,11 @@ AUPlugin::AUPlugin (AudioEngine& engine, Session& session, boost::shared_ptrGetElementCount (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)); @@ -738,7 +738,7 @@ AUPlugin::discover_parameters () kAudioUnitParameterUnit_Boolean = 2 kAudioUnitParameterUnit_Percent = 3 kAudioUnitParameterUnit_Seconds = 4 - kAudioUnitParameterUnit_SampleFrames = 5 + kAudioUnitParameterUnit_SampleSamples = 5 kAudioUnitParameterUnit_Phase = 6 kAudioUnitParameterUnit_Rate = 7 kAudioUnitParameterUnit_Hertz = 8 @@ -787,7 +787,7 @@ AUPlugin::discover_parameters () 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); @@ -950,12 +950,12 @@ AUPlugin::default_value (uint32_t port) 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; @@ -1035,7 +1035,7 @@ AUPlugin::activate () 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; } } @@ -1067,17 +1067,17 @@ int 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; } @@ -1110,7 +1110,7 @@ AUPlugin::configure_io (ChanCount in, ChanCount out) } } - streamFormat.mSampleRate = _session.frame_rate(); + streamFormat.mSampleRate = _session.sample_rate(); streamFormat.mFormatID = kAudioFormatLinearPCM; streamFormat.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked|kAudioFormatFlagIsNonInterleaved; @@ -1121,7 +1121,7 @@ AUPlugin::configure_io (ChanCount in, ChanCount out) #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. @@ -1582,13 +1582,13 @@ OSStatus 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!"); @@ -1611,7 +1611,7 @@ AUPlugin::render_callback(AudioUnitRenderActionFlags*, 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); @@ -1621,19 +1621,19 @@ AUPlugin::render_callback(AudioUnitRenderActionFlags*, 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; @@ -1684,7 +1684,7 @@ AUPlugin::connect_and_run (BufferSet& bufs, /* one MIDI port/buffer only */ MidiBuffer& m = bufs.get_midi (i); for (MidiBuffer::iterator i = m.begin(); i != m.end(); ++i) { - Evoral::Event ev (*i); + Evoral::Event 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)); @@ -1736,11 +1736,11 @@ AUPlugin::connect_and_run (BufferSet& bufs, } /* 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) { @@ -1787,7 +1787,7 @@ AUPlugin::connect_and_run (BufferSet& bufs, input_maxbuf = 0; if (ok) { - frames_processed += nframes; + samples_processed += nframes; return 0; } return -1; @@ -1802,11 +1802,11 @@ AUPlugin::get_beat_and_tempo_callback (Float64* outCurrentBeat, 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; @@ -1823,18 +1823,18 @@ AUPlugin::get_musical_time_location_callback (UInt32* outDeltaSampleOffsetToNe 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); } } @@ -1894,7 +1894,7 @@ AUPlugin::get_transport_state_callback (Boolean* outIsPlaying, /* 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) { @@ -1912,11 +1912,11 @@ AUPlugin::get_transport_state_callback (Boolean* outIsPlaying, 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); } } } @@ -3438,7 +3438,7 @@ AUPlugin::parameter_change_listener (void* /*arg*/, void* src, const AudioUnitEv 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; -- cgit v1.2.3