static boolByte _openSampleSourceAiff(void *sampleSourcePtr, const SampleSourceOpenAs openAs) {
SampleSource sampleSource = (SampleSource)sampleSourcePtr;
#if HAVE_LIBAUDIOFILE
SampleSourceAudiofileData extraData = (SampleSourceAudiofileData)(sampleSource->extraData);
#else
SampleSourcePcmData extraData = (SampleSourcePcmData)(sampleSource->extraData);
#endif
if(openAs == SAMPLE_SOURCE_OPEN_READ) {
#if HAVE_LIBAUDIOFILE
extraData->fileHandle = afOpenFile(sampleSource->sourceName->data, "r", NULL);
if(extraData->fileHandle != NULL) {
setNumChannels(afGetVirtualChannels(extraData->fileHandle, AF_DEFAULT_TRACK));
setSampleRate((float)afGetRate(extraData->fileHandle, AF_DEFAULT_TRACK));
}
#else
logInternalError("Executable was not built with a library to read AIFF files");
#endif
}
else if(openAs == SAMPLE_SOURCE_OPEN_WRITE) {
#if HAVE_LIBAUDIOFILE
AFfilesetup outfileSetup = afNewFileSetup();
afInitFileFormat(outfileSetup, AF_FILE_AIFF);
afInitByteOrder(outfileSetup, AF_DEFAULT_TRACK, AF_BYTEORDER_BIGENDIAN);
afInitChannels(outfileSetup, AF_DEFAULT_TRACK, getNumChannels());
afInitRate(outfileSetup, AF_DEFAULT_TRACK, getSampleRate());
afInitSampleFormat(outfileSetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, DEFAULT_BITRATE);
extraData->fileHandle = afOpenFile(sampleSource->sourceName->data, "w", outfileSetup);
#else
logInternalError("Executable was not built with a library to write AIFF files");
#endif
}
else {
logInternalError("Invalid type for openAs in AIFF file");
return false;
}
if(extraData->fileHandle == NULL) {
logError("AIFF file '%s' could not be opened for '%s'",
sampleSource->sourceName->data, openAs == SAMPLE_SOURCE_OPEN_READ ? "reading" : "writing");
return false;
}
sampleSource->openedAs = openAs;
return true;
}
main (int argc, char **argv)
{
AFfilehandle file;
AFframecount count, frameCount;
int channelCount, sampleFormat, sampleWidth;
float frameSize;
void *buffer;
double sampleRate;
ALport outport;
ALconfig outportconfig;
if (argc < 2)
usage();
file = afOpenFile(argv[1], "r", NULL);
if (file == AF_NULL_FILEHANDLE)
{
fprintf(stderr, "Could not open file %s.\n", argv[1]);
exit(EXIT_FAILURE);
}
frameCount = afGetFrameCount(file, AF_DEFAULT_TRACK);
frameSize = afGetVirtualFrameSize(file, AF_DEFAULT_TRACK, 1);
channelCount = afGetVirtualChannels(file, AF_DEFAULT_TRACK);
sampleRate = afGetRate(file, AF_DEFAULT_TRACK);
afGetVirtualSampleFormat(file, AF_DEFAULT_TRACK, &sampleFormat,
&sampleWidth);
if (sampleFormat == AF_SAMPFMT_UNSIGNED)
{
afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, sampleWidth);
}
printf("frame count: %lld\n", frameCount);
printf("frame size: %d bytes\n", (int) frameSize);
printf("channel count: %d\n", channelCount);
printf("sample rate: %.2f Hz\n", sampleRate);
buffer = malloc(BUFFERED_FRAME_COUNT * frameSize);
outportconfig = alNewConfig();
setwidth(outportconfig, sampleWidth);
setsampleformat(outportconfig, sampleFormat);
alSetChannels(outportconfig, channelCount);
count = afReadFrames(file, AF_DEFAULT_TRACK, buffer, BUFFERED_FRAME_COUNT);
outport = alOpenPort("irixread", "w", outportconfig);
setrate(outport, sampleRate);
do
{
printf("count = %lld\n", count);
alWriteFrames(outport, buffer, count);
count = afReadFrames(file, AF_DEFAULT_TRACK, buffer,
BUFFERED_FRAME_COUNT);
} while (count > 0);
waitport(outport);
alClosePort(outport);
alFreeConfig(outportconfig);
afCloseFile(file);
}
//***************************************************************************
bool Kwave::AudiofileDecoder::open(QWidget *widget, QIODevice &src)
{
metaData().clear();
Q_ASSERT(!m_source);
if (m_source) qWarning("AudiofileDecoder::open(), already open !");
// try to open the source
if (!src.open(QIODevice::ReadOnly)) {
qWarning("AudiofileDecoder::open(), failed to open source !");
return false;
}
// source successfully opened
m_source = &src;
m_src_adapter = new Kwave::VirtualAudioFile(*m_source);
Q_ASSERT(m_src_adapter);
if (!m_src_adapter) return false;
m_src_adapter->open(m_src_adapter, 0);
AFfilehandle fh = m_src_adapter->handle();
if (!fh || (m_src_adapter->lastError() >= 0)) {
QString reason;
switch (m_src_adapter->lastError()) {
case AF_BAD_NOT_IMPLEMENTED:
reason = i18n("Format or function is not implemented");
break;
case AF_BAD_MALLOC:
reason = i18n("Out of memory");
break;
case AF_BAD_HEADER:
reason = i18n("File header is damaged");
break;
case AF_BAD_CODEC_TYPE:
reason = i18n("Invalid codec type");
break;
case AF_BAD_OPEN:
reason = i18n("Opening the file failed");
break;
case AF_BAD_READ:
reason = i18n("Read access failed");
break;
case AF_BAD_SAMPFMT:
reason = i18n("Invalid sample format");
break;
default:
reason = reason.number(m_src_adapter->lastError());
}
QString text= i18n("An error occurred while opening the "\
"file:\n'%1'", reason);
Kwave::MessageBox::error(widget, text);
return false;
}
AFframecount length = afGetFrameCount(fh, AF_DEFAULT_TRACK);
unsigned int tracks = qMax(afGetVirtualChannels(fh, AF_DEFAULT_TRACK), 0);
unsigned int bits = 0;
double rate = 0.0;
int af_sample_format;
afGetVirtualSampleFormat(fh, AF_DEFAULT_TRACK, &af_sample_format,
reinterpret_cast<int *>(&bits));
Kwave::SampleFormat::Format fmt;
switch (af_sample_format)
{
case AF_SAMPFMT_TWOSCOMP:
fmt = Kwave::SampleFormat::Signed;
break;
case AF_SAMPFMT_UNSIGNED:
fmt = Kwave::SampleFormat::Unsigned;
break;
case AF_SAMPFMT_FLOAT:
fmt = Kwave::SampleFormat::Float;
break;
case AF_SAMPFMT_DOUBLE:
fmt = Kwave::SampleFormat::Double;
break;
default:
fmt = Kwave::SampleFormat::Unknown;
break;
}
// get sample rate, with fallback to 8kHz
rate = afGetRate(fh, AF_DEFAULT_TRACK);
if (rate < 1.0) {
qWarning("\n!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!\n"\
"WARNING: file has no sample rate!\n"\
" => using 8000 samples/sec as fallback\n"\
"!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!\n");
rate = 8000.0;
}
Kwave::SampleFormat::Map sf;
QString sample_format_name = sf.description(Kwave::SampleFormat(fmt), true);
if (static_cast<signed int>(bits) < 0) bits = 0;
int af_compression = afGetCompression(fh, AF_DEFAULT_TRACK);
const Kwave::Compression compression(
Kwave::Compression::fromAudiofile(af_compression)
);
Kwave::FileInfo info(metaData());
info.setRate(rate);
info.setBits(bits);
info.setTracks(tracks);
info.setLength(length);
info.set(INF_SAMPLE_FORMAT, Kwave::SampleFormat(fmt).toInt());
info.set(Kwave::INF_COMPRESSION, compression.toInt());
metaData().replace(Kwave::MetaDataList(info));
qDebug("-------------------------");
qDebug("info:");
qDebug("compression = %d", af_compression);
qDebug("channels = %d", info.tracks());
qDebug("rate = %0.0f", info.rate());
qDebug("bits/sample = %d", info.bits());
qDebug("length = %lu samples",
static_cast<unsigned long int>(info.length()));
qDebug("format = %d (%s)", af_sample_format,
DBG(sample_format_name));
qDebug("-------------------------");
// set up libaudiofile to produce Kwave's internal sample format
#if Q_BYTE_ORDER == Q_BIG_ENDIAN
afSetVirtualByteOrder(fh, AF_DEFAULT_TRACK, AF_BYTEORDER_BIGENDIAN);
#else
afSetVirtualByteOrder(fh, AF_DEFAULT_TRACK, AF_BYTEORDER_LITTLEENDIAN);
#endif
afSetVirtualSampleFormat(fh, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, SAMPLE_STORAGE_BITS);
return true;
}
static void
audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
{
AFvirtualfile *vf;
int fs, frame_count;
AFfilehandle af_fp;
int bits;
struct audio_format audio_format;
float total_time;
uint16_t bit_rate;
int ret, current = 0;
char chunk[CHUNK_SIZE];
enum decoder_command cmd;
if (!is->seekable) {
g_warning("not seekable");
return;
}
vf = setup_virtual_fops(is);
af_fp = afOpenVirtualFile(vf, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
g_warning("failed to input stream\n");
return;
}
afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
if (!audio_valid_sample_format(bits)) {
g_debug("input file has %d bit samples, converting to 16",
bits);
bits = 16;
}
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, bits);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
audio_format.bits = (uint8_t)bits;
audio_format.sample_rate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
audio_format.channels =
(uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
audio_format.sample_rate, audio_format.bits,
audio_format.channels);
afCloseFile(af_fp);
return;
}
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
total_time = ((float)frame_count / (float)audio_format.sample_rate);
bit_rate = (uint16_t)(is->size * 8.0 / total_time / 1000.0 + 0.5);
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder, &audio_format, true, total_time);
do {
ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
CHUNK_SIZE / fs);
if (ret <= 0)
break;
current += ret;
cmd = decoder_data(decoder, NULL,
chunk, ret * fs,
(float)current /
(float)audio_format.sample_rate,
bit_rate, NULL);
if (cmd == DECODE_COMMAND_SEEK) {
current = decoder_seek_where(decoder) *
audio_format.sample_rate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
decoder_command_finished(decoder);
cmd = DECODE_COMMAND_NONE;
}
} while (cmd == DECODE_COMMAND_NONE);
afCloseFile(af_fp);
}
static boolByte _openSampleSourceWave(void *sampleSourcePtr, const SampleSourceOpenAs openAs) {
SampleSource sampleSource = (SampleSource)sampleSourcePtr;
#if HAVE_LIBAUDIOFILE
SampleSourceAudiofileData extraData = sampleSource->extraData;
#else
SampleSourcePcmData extraData = (SampleSourcePcmData)sampleSource->extraData;
#endif
if(openAs == SAMPLE_SOURCE_OPEN_READ) {
#if HAVE_LIBAUDIOFILE
extraData->fileHandle = afOpenFile(sampleSource->sourceName->data, "r", NULL);
if(extraData->fileHandle != NULL) {
setNumChannels(afGetVirtualChannels(extraData->fileHandle, AF_DEFAULT_TRACK));
setSampleRate((float)afGetRate(extraData->fileHandle, AF_DEFAULT_TRACK));
}
#else
extraData->fileHandle = fopen(sampleSource->sourceName->data, "rb");
if(extraData->fileHandle != NULL) {
if(_readWaveFileInfo(sampleSource->sourceName->data, extraData)) {
setNumChannels(extraData->numChannels);
setSampleRate(extraData->sampleRate);
}
else {
fclose(extraData->fileHandle);
extraData->fileHandle = NULL;
}
}
#endif
}
else if(openAs == SAMPLE_SOURCE_OPEN_WRITE) {
#if HAVE_LIBAUDIOFILE
AFfilesetup outfileSetup = afNewFileSetup();
afInitFileFormat(outfileSetup, AF_FILE_WAVE);
afInitByteOrder(outfileSetup, AF_DEFAULT_TRACK, AF_BYTEORDER_LITTLEENDIAN);
afInitChannels(outfileSetup, AF_DEFAULT_TRACK, getNumChannels());
afInitRate(outfileSetup, AF_DEFAULT_TRACK, getSampleRate());
afInitSampleFormat(outfileSetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, DEFAULT_BITRATE);
extraData->fileHandle = afOpenFile(sampleSource->sourceName->data, "w", outfileSetup);
#else
extraData->fileHandle = fopen(sampleSource->sourceName->data, "wb");
if(extraData->fileHandle != NULL) {
extraData->numChannels = (unsigned short)getNumChannels();
extraData->sampleRate = (unsigned int)getSampleRate();
extraData->bitsPerSample = 16;
if(!_writeWaveFileInfo(extraData)) {
fclose(extraData->fileHandle);
extraData->fileHandle = NULL;
}
}
#endif
}
else {
logInternalError("Invalid type for openAs in WAVE file");
return false;
}
if(extraData->fileHandle == NULL) {
logError("WAVE file '%s' could not be opened for %s",
sampleSource->sourceName->data, openAs == SAMPLE_SOURCE_OPEN_READ ? "reading" : "writing");
return false;
}
sampleSource->openedAs = openAs;
return true;
}
static void
audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
{
GError *error = NULL;
AFvirtualfile *vf;
int fs, frame_count;
AFfilehandle af_fp;
struct audio_format audio_format;
float total_time;
uint16_t bit_rate;
int ret;
char chunk[CHUNK_SIZE];
enum decoder_command cmd;
if (!is->seekable) {
g_warning("not seekable");
return;
}
vf = setup_virtual_fops(is);
af_fp = afOpenVirtualFile(vf, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
g_warning("failed to input stream\n");
return;
}
if (!audio_format_init_checked(&audio_format,
afGetRate(af_fp, AF_DEFAULT_TRACK),
audiofile_setup_sample_format(af_fp),
afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK),
&error)) {
g_warning("%s", error->message);
g_error_free(error);
afCloseFile(af_fp);
return;
}
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
total_time = ((float)frame_count / (float)audio_format.sample_rate);
bit_rate = (uint16_t)(is->size * 8.0 / total_time / 1000.0 + 0.5);
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder, &audio_format, true, total_time);
do {
ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
CHUNK_SIZE / fs);
if (ret <= 0)
break;
cmd = decoder_data(decoder, NULL,
chunk, ret * fs,
bit_rate);
if (cmd == DECODE_COMMAND_SEEK) {
AFframecount frame = decoder_seek_where(decoder) *
audio_format.sample_rate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame);
decoder_command_finished(decoder);
cmd = DECODE_COMMAND_NONE;
}
} while (cmd == DECODE_COMMAND_NONE);
afCloseFile(af_fp);
}
