SoundSample::SoundSample(std::string name, int rev)
{
uMin = rev;
samples = NULL;
#ifdef HAVE_AUDIOFILE
lowPitchOffset = highPitchOffset = 1.0;
lowVolumeOffset = highVolumeOffset = 1.0;
lastDir = 0;
lastValue = 0;
file = afOpenFile(name.c_str(), "r", 0);
if (file == AF_NULL_FILEHANDLE)
{
QMessageBox::critical(theWindow, "Error",
"SoundSample::afOpenFD() - failed!",
QMessageBox::Ok,
QMessageBox::Ok);
}
else
{
afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
//afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32);
channels = afGetChannels(file, AF_DEFAULT_TRACK);
samplesPerSec = (int)afGetRate(file, AF_DEFAULT_TRACK);
numSamples = afGetFrameCount(file, AF_DEFAULT_TRACK);
sampleBlocksize = (int)afGetVirtualFrameSize(file, AF_DEFAULT_TRACK, 1);
bytesPerSample = (int)afGetVirtualFrameSize(file, AF_DEFAULT_TRACK, 1) / channels;
if (channels != 2)
{
std::strstream buf;
buf << "SoundSample::not a stereo sound file" << name;
std::string s;
buf >> s;
QMessageBox::critical(theWindow, "Error",
s.c_str(),
QMessageBox::Ok,
QMessageBox::Ok);
}
if (samplesPerSec != 48000)
{
std::strstream buf;
buf << "SoundSample::48kHz required but " << name << " has " << samplesPerSec;
std::string s;
buf >> s;
QMessageBox::critical(theWindow, "Error",
s.c_str(),
QMessageBox::Ok,
QMessageBox::Ok);
}
static enum sample_format
audiofile_setup_sample_format(AFfilehandle af_fp)
{
int fs, bits;
afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
if (!audio_valid_sample_format(audiofile_bits_to_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);
return audiofile_bits_to_sample_format(bits);
}
static enum audiotap_status audiofile_read_init(struct audiotap **audiotap,
const char *file,
struct tapenc_params *params,
uint8_t machine,
uint8_t videotype,
uint8_t *halfwaves){
uint32_t freq;
enum audiotap_status error = AUDIOTAP_LIBRARY_ERROR;
AFfilehandle fh;
if (status.audiofile_init_status != LIBRARY_OK
|| status.tapencoder_init_status != LIBRARY_OK)
return AUDIOTAP_LIBRARY_UNAVAILABLE;
fh=afOpenFile(file,"r", NULL);
if (fh == AF_NULL_FILEHANDLE)
return AUDIOTAP_LIBRARY_ERROR;
do{
if ( (freq=(uint32_t)afGetRate(fh, AF_DEFAULT_TRACK)) == -1)
break;
if (afSetVirtualChannels(fh, AF_DEFAULT_TRACK, 1) == -1)
break;
if (afSetVirtualSampleFormat(fh, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 32) == -1)
break;
if (afGetVirtualFrameSize(fh, AF_DEFAULT_TRACK, 0) != 4)
break;
error = AUDIOTAP_OK;
}while(0);
if(error != AUDIOTAP_OK){
afCloseFile(fh);
return error;
}
*halfwaves = 1;
return audio2tap_audio_open_common(audiotap,
freq,
params,
machine,
videotype,
&audiofile_read_functions,
fh);
}
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);
}
int main (int argc, char **argv)
{
AFfilesetup setup;
if ((setup = afNewFileSetup()) == AF_NULL_FILESETUP)
{
fprintf(stderr, "Could not allocate file setup.\n");
exit(EXIT_FAILURE);
}
afInitFileFormat(setup, AF_FILE_IRCAM);
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32);
char *testFileName;
if (!createTemporaryFile("floatto24", &testFileName))
{
fprintf(stderr, "Could not create temporary file.\n");
exit(EXIT_FAILURE);
}
AFfilehandle file = afOpenFile(testFileName, "w", setup);
if (file == AF_NULL_FILEHANDLE)
{
printf("could not open file for writing\n");
exit(EXIT_FAILURE);
}
afFreeFileSetup(setup);
AFframecount framesWritten = afWriteFrames(file, AF_DEFAULT_TRACK, samples,
frameCount);
if (framesWritten != frameCount)
{
fprintf(stderr, "Wrong number of frames read.\n");
exit(EXIT_FAILURE);
}
if (afCloseFile(file) != 0)
{
fprintf(stderr, "Closing file returned non-zero status.\n");
exit(EXIT_FAILURE);
}
file = afOpenFile(testFileName, "r", AF_NULL_FILESETUP);
if (file == AF_NULL_FILEHANDLE)
{
fprintf(stderr, "Could not open file for writing.\n");
exit(EXIT_FAILURE);
}
if (afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 24) != 0)
{
fprintf(stderr, "afSetVirtualSampleFormat returned non-zero status.\n");
exit(EXIT_FAILURE);
}
int readSamples[frameCount];
for (int i=0; i<frameCount; i++)
readSamples[i] = -1000 - i;
AFframecount framesRead = afReadFrames(file, AF_DEFAULT_TRACK, readSamples,
frameCount);
if (framesRead != frameCount)
{
fprintf(stderr, "Wrong number of frames read.\n");
exit(EXIT_FAILURE);
}
for (int i=0; i<framesRead; i++)
{
#ifdef DEBUG
printf("[%d] = %d\n", i, readSamples[i]);
#endif
if (readSamples[i] == -1000 - i)
{
fprintf(stderr, "Data in destination array untouched.\n");
exit(EXIT_FAILURE);
}
/*
Ensure that the high-order 8 bits represent
sign extension: only 0x00 (+) or 0xff (-) is
valid.
*/
if ((readSamples[i] & 0xff000000) != 0x000000 &&
(readSamples[i] & 0xff000000) != 0xff000000)
{
fprintf(stderr, "Data is not within range of "
"{-2^23, ..., 2^23-1}.\n");
exit(EXIT_FAILURE);
}
if (readSamples[i] != referenceConvertedSamples[i])
{
fprintf(stderr, "Data doesn't match reference data.\n");
exit(EXIT_FAILURE);
}
}
if (afCloseFile(file) != 0)
{
fprintf(stderr, "Closing file returned non-zero status.\n");
exit(EXIT_FAILURE);
}
unlink(testFileName);
free(testFileName);
exit(EXIT_SUCCESS);
}
int main (int argc, char **argv)
{
int i = 1;
char *infilename, *outfilename;
int fileFormat, outFileFormat = AF_FILE_UNKNOWN;
AFfilehandle infile, outfile;
AFfilesetup outfilesetup;
int sampleFormat, sampleWidth, channelCount;
double sampleRate;
int outSampleFormat = -1, outSampleWidth = -1,
outChannelCount = -1;
double outMaxAmp = 1.0;
AFframecount totalFrames;
if (argc == 2)
{
if (!strcmp(argv[1], "--version") || !strcmp(argv[1], "-v"))
{
printversion();
exit(EXIT_SUCCESS);
}
if (!strcmp(argv[1], "--help") || !strcmp(argv[1], "-h"))
{
printusage();
exit(EXIT_SUCCESS);
}
}
if (argc < 3)
usageerror();
infilename = argv[1];
outfilename = argv[2];
i = 3;
while (i < argc)
{
if (!strcmp(argv[i], "format"))
{
if (i + 1 >= argc)
usageerror();
if (!strcmp(argv[i+1], "aiff"))
outFileFormat = AF_FILE_AIFF;
else if (!strcmp(argv[i+1], "aifc"))
outFileFormat = AF_FILE_AIFFC;
else if (!strcmp(argv[i+1], "wave"))
outFileFormat = AF_FILE_WAVE;
else if (!strcmp(argv[i+1], "next"))
outFileFormat = AF_FILE_NEXTSND;
else if (!strcmp(argv[i+1], "bics"))
outFileFormat = AF_FILE_BICSF;
else if (!strcmp(argv[i+1], "voc"))
outFileFormat = AF_FILE_VOC;
else if (!strcmp(argv[i+1], "nist"))
outFileFormat = AF_FILE_NIST_SPHERE;
else if (!strcmp(argv[i+1], "caf"))
outFileFormat = AF_FILE_CAF;
else
{
fprintf(stderr, "sfconvert: Unknown format %s.\n", argv[i+1]);
exit(EXIT_FAILURE);
}
/* Increment for argument. */
i++;
}
else if (!strcmp(argv[i], "channels"))
{
if (i + 1 >= argc)
usageerror();
outChannelCount = atoi(argv[i+1]);
if (outChannelCount < 1)
usageerror();
/* Increment for argument. */
i++;
}
else if (!strcmp(argv[i], "float"))
{
if (i + 1 >= argc)
usageerror();
outSampleFormat = AF_SAMPFMT_FLOAT;
outSampleWidth = 32;
outMaxAmp = atof(argv[i+1]);
/* Increment for argument. */
i++;
}
else if (!strcmp(argv[i], "integer"))
{
if (i + 2 >= argc)
usageerror();
outSampleWidth = atoi(argv[i+1]);
if (outSampleWidth < 1 || outSampleWidth > 32)
usageerror();
if (!strcmp(argv[i+2], "2scomp"))
outSampleFormat = AF_SAMPFMT_TWOSCOMP;
else if (!strcmp(argv[i+2], "unsigned"))
outSampleFormat = AF_SAMPFMT_UNSIGNED;
else
usageerror();
/* Increment for arguments. */
i += 2;
}
else
{
printf("Unrecognized command %s\n", argv[i]);
}
i++;
}
infile = afOpenFile(infilename, "r", AF_NULL_FILESETUP);
if (infile == AF_NULL_FILEHANDLE)
{
printf("Could not open file '%s' for reading.\n", infilename);
return 1;
}
/* Get audio format parameters from input file. */
fileFormat = afGetFileFormat(infile, NULL);
totalFrames = afGetFrameCount(infile, AF_DEFAULT_TRACK);
channelCount = afGetChannels(infile, AF_DEFAULT_TRACK);
sampleRate = afGetRate(infile, AF_DEFAULT_TRACK);
afGetSampleFormat(infile, AF_DEFAULT_TRACK, &sampleFormat, &sampleWidth);
/* Initialize output audio format parameters. */
outfilesetup = afNewFileSetup();
if (outFileFormat == -1)
outFileFormat = fileFormat;
if (outSampleFormat == -1 || outSampleWidth == -1)
{
outSampleFormat = sampleFormat;
outSampleWidth = sampleWidth;
}
if (outChannelCount == -1)
outChannelCount = channelCount;
afInitFileFormat(outfilesetup, outFileFormat);
afInitSampleFormat(outfilesetup, AF_DEFAULT_TRACK, outSampleFormat,
outSampleWidth);
afInitChannels(outfilesetup, AF_DEFAULT_TRACK, outChannelCount);
afInitRate(outfilesetup, AF_DEFAULT_TRACK, sampleRate);
outfile = afOpenFile(outfilename, "w", outfilesetup);
if (outfile == AF_NULL_FILEHANDLE)
{
printf("Could not open file '%s' for writing.\n", outfilename);
return 1;
}
/*
Set the output file's virtual audio format parameters
to match the audio format parameters of the input file.
*/
afSetVirtualChannels(outfile, AF_DEFAULT_TRACK, channelCount);
afSetVirtualSampleFormat(outfile, AF_DEFAULT_TRACK, sampleFormat,
sampleWidth);
afFreeFileSetup(outfilesetup);
copyaudiodata(infile, outfile, AF_DEFAULT_TRACK, totalFrames);
afCloseFile(infile);
afCloseFile(outfile);
printfileinfo(infilename);
putchar('\n');
printfileinfo(outfilename);
return EXIT_SUCCESS;
}
void print_power (char *filename)
{
AFfilehandle file;
double *sums, *frames;
int channelCount, windowSize, frameCount;
int i, c;
struct smooth *powsmooth;
int winStart, winEnd;
int lastWindow = FALSE;
double pow, maxpow;
double level, peak, minSample = 1, maxSample = -1;
file = afOpenFile(filename, "r", NULL);
if (file == AF_NULL_FILEHANDLE)
{
fprintf(stderr, "Could not open file %s.\n", filename);
return;
}
channelCount = afGetChannels(file, AF_DEFAULT_TRACK);
windowSize = afGetRate(file, AF_DEFAULT_TRACK) / 100;
frameCount = afGetFrameCount(file, AF_DEFAULT_TRACK);
sums = calloc(channelCount, sizeof (double));
for (c=0; c<channelCount; c++)
sums[c] = 0;
frames = calloc(channelCount * windowSize, sizeof (double));
afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_DOUBLE,
sizeof (double));
powsmooth = calloc(channelCount, sizeof (struct smooth));
for (c=0; c<channelCount; c++)
{
/* Use a 100-element (1 second) window. */
powsmooth[c].length = 100;
powsmooth[c].buf = calloc(powsmooth[c].length, sizeof (double));
powsmooth[c].start = 0;
powsmooth[c].n = 0;
}
winStart = 0;
winEnd = 0;
lastWindow = FALSE;
maxpow = 0;
do
{
winEnd = winStart + windowSize;
if (winEnd >= frameCount)
{
winEnd = frameCount;
lastWindow = TRUE;
}
afReadFrames(file, AF_DEFAULT_TRACK, frames, windowSize);
for (c=0; c<channelCount; c++)
{
sums[c] = 0;
for (i=0; i < winEnd - winStart; i++)
{
double sample;
sample = frames[i*channelCount + c];
sums[c] += sample*sample;
if (sample > maxSample)
maxSample = sample;
if (sample < minSample)
minSample = sample;
}
}
/* Compute power for each channel. */
for (c=0; c<channelCount; c++)
{
double pow;
int end;
pow = sums[c] / (winEnd - winStart);
end = (powsmooth[c].start + powsmooth[c].n) %
powsmooth[c].length;
powsmooth[c].buf[end] = pow;
if (powsmooth[c].n == powsmooth[c].length)
{
powsmooth[c].start = (powsmooth[c].start + 1) % powsmooth[c].length;
pow = get_smoothed_data(&powsmooth[c]);
if (pow > maxpow)
maxpow = pow;
}
else
{
powsmooth[c].n++;
}
}
winStart += windowSize;
} while (!lastWindow);
for (c = 0; c < channelCount; c++)
{
pow = get_smoothed_data(&powsmooth[c]);
if (pow > maxpow)
maxpow = pow;
}
free(sums);
free(frames);
for (c=0; c<channelCount; c++)
free(powsmooth[c].buf);
free(powsmooth);
level = sqrt(maxpow);
afCloseFile(file);
printf("file: %s\n", filename);
printf("level (dB): %f\n", 20 * log10(level));
printf("peak-: %f\n", minSample);
printf("peak+: %f\n", maxSample);
peak = abs(minSample);
if (peak < abs(maxSample))
peak = abs(maxSample);
printf("peak (dB): %f\n", 20 * log10(peak));
}
libspectrum_error
libspectrum_wav_read( libspectrum_tape *tape, const char *filename )
{
libspectrum_byte *buffer; size_t length;
libspectrum_byte *tape_buffer; size_t tape_length;
size_t data_length;
libspectrum_tape_block *block = NULL;
int frames;
/* Our filehandle from libaudiofile */
AFfilehandle handle;
/* The track we're using in the file */
int track = AF_DEFAULT_TRACK;
if( !filename ) {
libspectrum_print_error(
LIBSPECTRUM_ERROR_LOGIC,
"libspectrum_wav_read: no filename provided - wav files can only be loaded from a file"
);
return LIBSPECTRUM_ERROR_LOGIC;
}
handle = afOpenFile( filename, "r", NULL );
if( handle == AF_NULL_FILEHANDLE ) {
libspectrum_print_error(
LIBSPECTRUM_ERROR_LOGIC,
"libspectrum_wav_read: audiofile failed to open file:%s", filename
);
return LIBSPECTRUM_ERROR_LOGIC;
}
if( afSetVirtualSampleFormat( handle, track, AF_SAMPFMT_UNSIGNED, 8 ) ) {
afCloseFile( handle );
libspectrum_print_error(
LIBSPECTRUM_ERROR_LOGIC,
"libspectrum_wav_read: audiofile failed to set virtual sample format"
);
return LIBSPECTRUM_ERROR_LOGIC;
}
if( afSetVirtualChannels( handle, track, 1 ) ) {
afCloseFile( handle );
libspectrum_print_error(
LIBSPECTRUM_ERROR_LOGIC,
"libspectrum_wav_read: audiofile failed to set virtual channel count"
);
return LIBSPECTRUM_ERROR_LOGIC;
}
length = afGetFrameCount( handle, track );
tape_length = length;
if( tape_length%8 ) tape_length += 8 - (tape_length%8);
buffer = libspectrum_new0( libspectrum_byte,
tape_length * afGetChannels(handle, track) );
frames = afReadFrames( handle, track, buffer, length );
if( frames == -1 ) {
libspectrum_free( buffer );
afCloseFile( handle );
libspectrum_print_error(
LIBSPECTRUM_ERROR_CORRUPT,
"libspectrum_wav_read: can't calculate number of frames in audio file"
);
return LIBSPECTRUM_ERROR_CORRUPT;
}
if( !length ) {
libspectrum_free( buffer );
afCloseFile( handle );
libspectrum_print_error(
LIBSPECTRUM_ERROR_CORRUPT,
"libspectrum_wav_read: empty audio file, nothing to load"
);
return LIBSPECTRUM_ERROR_CORRUPT;
}
if( frames != length ) {
libspectrum_free( buffer );
afCloseFile( handle );
libspectrum_print_error(
LIBSPECTRUM_ERROR_CORRUPT,
"libspectrum_wav_read: read %d frames, but expected %lu\n", frames,
(unsigned long)length
);
return LIBSPECTRUM_ERROR_CORRUPT;
}
block = libspectrum_tape_block_alloc( LIBSPECTRUM_TAPE_BLOCK_RAW_DATA );
/* 44100 Hz 79 t-states 22050 Hz 158 t-states */
libspectrum_tape_block_set_bit_length( block,
3500000/afGetRate( handle, track ) );
libspectrum_set_pause_ms( block, 0 );
libspectrum_tape_block_set_bits_in_last_byte( block,
length % LIBSPECTRUM_BITS_IN_BYTE ?
length % LIBSPECTRUM_BITS_IN_BYTE : LIBSPECTRUM_BITS_IN_BYTE );
data_length = tape_length / LIBSPECTRUM_BITS_IN_BYTE;
libspectrum_tape_block_set_data_length( block, data_length );
tape_buffer = libspectrum_new0( libspectrum_byte, data_length );
libspectrum_byte *from = buffer;
libspectrum_byte *to = tape_buffer;
length = tape_length;
do {
libspectrum_byte val = 0;
int i;
for( i = 7; i >= 0; i-- ) {
if( *from++ > 127 ) val |= 1 << i;
}
*to++ = val;
} while ((length -= 8) > 0);
libspectrum_tape_block_set_data( block, tape_buffer );
libspectrum_tape_append_block( tape, block );
if( afCloseFile( handle ) ) {
libspectrum_free( buffer );
libspectrum_print_error(
LIBSPECTRUM_ERROR_UNKNOWN,
"libspectrum_wav_read: failed to close audio file"
);
return LIBSPECTRUM_ERROR_UNKNOWN;
}
libspectrum_free( buffer );
/* Successful completion */
return LIBSPECTRUM_ERROR_NONE;
}
AFfilehandle openTestFile(int sampleWidth)
{
AFfilehandle file = afOpenFile(m_testFileName.c_str(), "r", AF_NULL_FILESETUP);
afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_UNSIGNED, sampleWidth);
return file;
}
OSStatus setupOutput (AudioUnit *outputUnit, AFfilehandle file)
{
OSStatus status = noErr;
UInt32 size;
Boolean outWritable;
AudioStreamBasicDescription fileASBD, inputASBD, outputASBD;
AURenderCallbackStruct renderCallback;
/* Set virtual sample format to single-precision floating-point. */
afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32);
/* Get ASBD for virtual sample format. */
getASBDForFile(file, AF_DEFAULT_TRACK, &fileASBD);
status = AudioUnitGetPropertyInfo(*outputUnit,
kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output,
0, &size, &outWritable);
status = AudioUnitGetProperty(*outputUnit,
kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output,
0, &outputASBD, &size);
if (outWritable)
{
outputASBD = fileASBD;
status = AudioUnitSetProperty(*outputUnit,
kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output,
0, &outputASBD, size);
}
inputASBD = fileASBD;
status = AudioUnitSetProperty(*outputUnit,
kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
0, &inputASBD, size);
if (status != noErr)
{
fprintf(stderr, "Could not set input stream format.\n");
exit(EXIT_FAILURE);
}
/*
Set the render callback to a procedure which will
read from the file.
*/
renderCallback.inputProc = fileRenderProc;
renderCallback.inputProcRefCon = file;
status = AudioUnitSetProperty(*outputUnit,
kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0,
&renderCallback, sizeof (AURenderCallbackStruct));
if (status != noErr)
{
fprintf(stderr, "Could not set render callback.\n");
exit(EXIT_FAILURE);
}
return status;
}
VeSound *veSoundLoadFile_AudioFile(char *file) {
AFfilehandle fh;
VeSound *snd;
int nchan, nframe;
int fsz, i, n;
float *buf;
if (!(fh = afOpenFile(file,"r",0))) {
veError(MODULE,"audiofile library could not open sound file '%s'",
file);
return NULL;
}
afSetVirtualSampleFormat(fh,AF_DEFAULT_TRACK,AF_SAMPFMT_FLOAT,4);
// Hack.
// afSetVirtualRate(fh,AF_DEFAULT_TRACK,(double)veAudioGetSampFreq());
nchan = afGetChannels(fh,AF_DEFAULT_TRACK);
if (nchan < 1) {
veError(MODULE,"sound file has no tracks?");
afCloseFile(fh);
return NULL;
}
if (nchan > 2) {
veError(MODULE,"sound file has more than 2 tracks - rejecting it");
afCloseFile(fh);
return NULL;
}
nframe = afGetFrameCount(fh,AF_DEFAULT_TRACK);
fsz = veAudioGetFrameSize();
snd = veAllocObj(VeSound);
snd->nframes = nframe/fsz;
if (n % fsz)
snd->nframes++;
snd->data = veAlloc(snd->nframes*sizeof(float)*fsz,0);
buf = veAlloc(2*fsz*sizeof(float),0);
for (i = 0; i < snd->nframes; i++) {
memset(buf,0,fsz*sizeof(float)); /* zero buffer */
n = afReadFrames(fh,AF_DEFAULT_TRACK,buf,fsz);
if (n == -1) {
veError(MODULE,"sound file read failed for '%s': %s",
file, strerror(errno));
afCloseFile(fh);
veFree(buf);
veFree(snd->data);
veFree(snd);
return NULL;
}
if (nchan == 1)
/* remember: snd->data is a (float *) */
memcpy(snd->data+fsz*i,buf,fsz*sizeof(float));
else {
/* downmix stereo to mono (trivial) */
int k;
for (k = 0; k < fsz; k++)
snd->data[fsz*i+k] = (buf[2*k] + buf[2*k+1])/2.0;
}
}
afCloseFile(fh);
veFree(buf);
snd->file = veDupString(file);
/* initialize other fields */
snd->id = -1;
snd->name = NULL;
return snd;
}
