/*
- Copyright (C) 2006 Paul Davis
+ Copyright (C) 2006 Paul Davis
+ Written by Taybin Rutkin
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
*/
-#include <pbd/error.h>
-#include <ardour/coreaudiosource.h>
-#include <ardour/utils.h>
+#include <algorithm>
+#define __STDC_FORMAT_MACROS
+#include <inttypes.h>
+
+#include "pbd/error.h"
+#include "ardour/coreaudiosource.h"
+#include "ardour/utils.h"
#include <appleutility/CAAudioFile.h>
#include <appleutility/CAStreamBasicDescription.h>
#include <AudioToolbox/AudioFormat.h>
+using namespace std;
using namespace ARDOUR;
using namespace PBD;
CoreAudioSource::CoreAudioSource (Session& s, const XMLNode& node)
- : AudioFileSource (s, node)
+ : Source (s, node),
+ AudioFileSource (s, node)
{
init ();
}
-CoreAudioSource::CoreAudioSource (Session& s, const string& path, int chn, Flag flags)
+CoreAudioSource::CoreAudioSource (Session& s, const string& path, bool, int chn, Flag flags)
/* files created this way are never writable or removable */
- : AudioFileSource (s, path, Flag (flags & ~(Writable|Removable|RemovableIfEmpty|RemoveAtDestroy)))
+ : Source (s, DataType::AUDIO, path, Source::Flag (flags & ~(Writable|Removable|RemovableIfEmpty|RemoveAtDestroy))),
+ AudioFileSource (s, path,
+ Source::Flag (flags & ~(Writable|Removable|RemovableIfEmpty|RemoveAtDestroy)))
{
_channel = chn;
init ();
void
CoreAudioSource::init ()
{
- tmpbuf = 0;
- tmpbufsize = 0;
-
- cerr << "CoreAudioSource::init() " << name() << endl;
-
/* note that we temporarily truncated _id at the colon */
try {
af.Open(_path.c_str());
- CAStreamBasicDescription file_asbd (af.GetFileDataFormat());
- n_channels = file_asbd.NumberChannels();
- cerr << "number of channels: " << n_channels << endl;
+ CAStreamBasicDescription file_format (af.GetFileDataFormat());
+ n_channels = file_format.NumberChannels();
if (_channel >= n_channels) {
error << string_compose("CoreAudioSource: file only contains %1 channels; %2 is invalid as a channel number (%3)", n_channels, _channel, name()) << endmsg;
_length = af.GetNumberFrames();
- CAStreamBasicDescription client_asbd(file_asbd);
- client_asbd.SetCanonical(client_asbd.NumberChannels(), false);
- af.SetClientFormat (client_asbd);
+ CAStreamBasicDescription client_format (file_format);
+
+ /* set canonial form (PCM, native float packed, 32 bit, with the correct number of channels
+ and interleaved (since we plan to deinterleave ourselves)
+ */
+
+ client_format.SetCanonical(client_format.NumberChannels(), true);
+ af.SetClientFormat (client_format);
+
} catch (CAXException& cax) {
- error << string_compose ("CoreAudioSource: %1 (%2)", cax.mOperation, name()) << endmsg;
+
+ error << string_compose(_("CoreAudioSource: cannot open file \"%1\" for %2"),
+ _path, (writable() ? "read+write" : "reading")) << endmsg;
throw failed_constructor ();
}
}
CoreAudioSource::~CoreAudioSource ()
{
- cerr << "CoreAudioSource::~CoreAudioSource() " << name() << endl;
GoingAway (); /* EMIT SIGNAL */
-
- if (tmpbuf) {
- delete [] tmpbuf;
- }
-
- cerr << "deletion done" << endl;
}
-nframes_t
-CoreAudioSource::read_unlocked (Sample *dst, nframes_t start, nframes_t cnt) const
+int
+CoreAudioSource::safe_read (Sample* dst, nframes_t start, nframes_t cnt, AudioBufferList& abl) const
{
- try {
- af.Seek (start);
- } catch (CAXException& cax) {
- error << string_compose("CoreAudioSource: %1 to %2 (%3)", cax.mOperation, start, _name.substr (1)) << endmsg;
- return 0;
- }
-
- AudioBufferList abl;
- abl.mNumberBuffers = 1;
- abl.mBuffers[0].mNumberChannels = n_channels;
+ nframes_t nread = 0;
- UInt32 new_cnt = cnt;
- if (n_channels == 1) {
- abl.mBuffers[0].mDataByteSize = cnt * sizeof(Sample);
- abl.mBuffers[0].mData = dst;
+ while (nread < cnt) {
+
+ try {
+ af.Seek (start+nread);
+ } catch (CAXException& cax) {
+ error << string_compose("CoreAudioSource: %1 to %2 (%3)", cax.mOperation, start+nread, _name.substr (1)) << endmsg;
+ return -1;
+ }
+
+ UInt32 new_cnt = cnt - nread;
+
+ abl.mBuffers[0].mDataByteSize = new_cnt * n_channels * sizeof(Sample);
+ abl.mBuffers[0].mData = dst + nread;
+
try {
af.Read (new_cnt, &abl);
} catch (CAXException& cax) {
error << string_compose("CoreAudioSource: %1 (%2)", cax.mOperation, _name);
+ return -1;
}
- _read_data_count = new_cnt * sizeof(float);
- return new_cnt;
- }
-
- UInt32 real_cnt = cnt * n_channels;
- {
- Glib::Mutex::Lock lm (_tmpbuf_lock);
-
- if (tmpbufsize < real_cnt) {
-
- if (tmpbuf) {
- delete [] tmpbuf;
+ if (new_cnt == 0) {
+ /* EOF */
+ if (start+cnt == _length) {
+ /* we really did hit the end */
+ nread = cnt;
}
- tmpbufsize = real_cnt;
- tmpbuf = new float[tmpbufsize];
+ break;
}
- abl.mBuffers[0].mDataByteSize = tmpbufsize * sizeof(Sample);
- abl.mBuffers[0].mData = tmpbuf;
+ nread += new_cnt;
+ }
+
+ if (nread < cnt) {
+ return -1;
+ } else {
+ return 0;
+ }
+}
+
+
+nframes_t
+CoreAudioSource::read_unlocked (Sample *dst, sframes_t start, nframes_t cnt) const
+{
+ nframes_t file_cnt;
+ AudioBufferList abl;
+
+ abl.mNumberBuffers = 1;
+ abl.mBuffers[0].mNumberChannels = n_channels;
+
+ if (start > _length) {
- cerr << "channel: " << _channel << endl;
+ /* read starts beyond end of data, just memset to zero */
- try {
- af.Read (real_cnt, &abl);
- } catch (CAXException& cax) {
- error << string_compose("CoreAudioSource: %1 (%2)", cax.mOperation, _name);
- }
- float *ptr = tmpbuf + _channel;
- real_cnt /= n_channels;
+ file_cnt = 0;
+
+ } else if (start + cnt > _length) {
+
+ /* read ends beyond end of data, read some, memset the rest */
- /* stride through the interleaved data */
+ file_cnt = _length - start;
+
+ } else {
- for (uint32_t n = 0; n < real_cnt; ++n) {
- dst[n] = *ptr;
- ptr += n_channels;
+ /* read is entirely within data */
+
+ file_cnt = cnt;
+ }
+
+ if (file_cnt != cnt) {
+ nframes_t delta = cnt - file_cnt;
+ memset (dst+file_cnt, 0, sizeof (Sample) * delta);
+ }
+
+ if (file_cnt) {
+
+ if (n_channels == 1) {
+ if (safe_read (dst, start, file_cnt, abl) == 0) {
+ _read_data_count = cnt * sizeof (Sample);
+ return cnt;
+ }
+ return 0;
}
}
+ Sample* interleave_buf = get_interleave_buffer (file_cnt * n_channels);
+
+ if (safe_read (interleave_buf, start, file_cnt, abl) != 0) {
+ return 0;
+ }
+
_read_data_count = cnt * sizeof(float);
-
- return real_cnt;
+
+ Sample *ptr = interleave_buf + _channel;
+
+ /* stride through the interleaved data */
+
+ for (uint32_t n = 0; n < file_cnt; ++n) {
+ dst[n] = *ptr;
+ ptr += n_channels;
+ }
+
+ return cnt;
}
float
}
int
-CoreAudioSource::update_header (nframes_t when, struct tm&, time_t)
+CoreAudioSource::update_header (sframes_t when, struct tm&, time_t)
{
return 0;
}
goto out;
}
- _info.format_name = CFStringRefToStdString(name);
+ _info.format_name = "";
+
+ if (absd.mFormatID == kAudioFormatLinearPCM) {
+ if (absd.mFormatFlags & kAudioFormatFlagIsBigEndian) {
+ _info.format_name += "big-endian";
+ } else {
+ _info.format_name += "little-endian";
+ }
+
+ char buf[32];
+ snprintf (buf, sizeof (buf), " %" PRIu32 " bit", absd.mBitsPerChannel);
+ _info.format_name += buf;
+ _info.format_name += '\n';
+
+ if (absd.mFormatFlags & kAudioFormatFlagIsFloat) {
+ _info.format_name += "float";
+ } else {
+ if (absd.mFormatFlags & kAudioFormatFlagIsSignedInteger) {
+ _info.format_name += "signed";
+ } else {
+ _info.format_name += "unsigned";
+ }
+ /* integer is typical, do not show it */
+ }
+
+ if (_info.channels > 1) {
+ if (absd.mFormatFlags & kAudioFormatFlagIsNonInterleaved) {
+ _info.format_name += " noninterleaved";
+ }
+ /* interleaved is the normal case, do not show it */
+ }
+
+ _info.format_name += ' ';
+ }
+
+ switch (absd.mFormatID) {
+ case kAudioFormatLinearPCM:
+ _info.format_name += "PCM";
+ break;
+
+ case kAudioFormatAC3:
+ _info.format_name += "AC3";
+ break;
+
+ case kAudioFormat60958AC3:
+ _info.format_name += "60958 AC3";
+ break;
+
+ case kAudioFormatMPEGLayer1:
+ _info.format_name += "MPEG-1";
+ break;
+
+ case kAudioFormatMPEGLayer2:
+ _info.format_name += "MPEG-2";
+ break;
+
+ case kAudioFormatMPEGLayer3:
+ _info.format_name += "MPEG-3";
+ break;
+
+ case kAudioFormatAppleIMA4:
+ _info.format_name += "IMA-4";
+ break;
+
+ case kAudioFormatMPEG4AAC:
+ _info.format_name += "AAC";
+ break;
+
+ case kAudioFormatMPEG4CELP:
+ _info.format_name += "CELP";
+ break;
+
+ case kAudioFormatMPEG4HVXC:
+ _info.format_name += "HVXC";
+ break;
+
+ case kAudioFormatMPEG4TwinVQ:
+ _info.format_name += "TwinVQ";
+ break;
+
+ /* these really shouldn't show up, but we should do something
+ somewhere else to make sure that doesn't happen. until
+ that is guaranteed, print something anyway.
+ */
+
+ case kAudioFormatTimeCode:
+ _info.format_name += "timecode";
+ break;
+
+ case kAudioFormatMIDIStream:
+ _info.format_name += "MIDI";
+ break;
+
+ case kAudioFormatParameterValueStream:
+ _info.format_name += "parameter values";
+ break;
+ }
// XXX it would be nice to find a way to get this information if it exists