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/*
* Copyright (C) 2003-2014 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "AudiofileDecoderPlugin.hxx"
#include "../DecoderAPI.hxx"
#include "input/InputStream.hxx"
#include "CheckAudioFormat.hxx"
#include "tag/TagHandler.hxx"
#include "fs/Path.hxx"
#include "util/Error.hxx"
#include "util/Domain.hxx"
#include "Log.hxx"
#include <audiofile.h>
#include <af_vfs.h>
#include <assert.h>
#include <stdio.h>
static constexpr Domain audiofile_domain("audiofile");
static void
audiofile_error_func(long, const char *msg)
{
LogWarning(audiofile_domain, msg);
}
static bool
audiofile_init(const config_param &)
{
afSetErrorHandler(audiofile_error_func);
return true;
}
struct AudioFileInputStream {
Decoder *const decoder;
InputStream &is;
size_t Read(void *buffer, size_t size) {
/* libaudiofile does not like partial reads at all,
and wil abort playback; therefore always force full
reads */
return decoder_read_full(decoder, is, buffer, size)
? size
: 0;
}
};
gcc_pure
static double
audiofile_get_duration(AFfilehandle fh)
{
double frame_count = afGetFrameCount(fh, AF_DEFAULT_TRACK);
double rate = afGetRate(fh, AF_DEFAULT_TRACK);
return frame_count / rate;
}
static ssize_t
audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length)
{
AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
return afis.Read(data, length);
}
static AFfileoffset
audiofile_file_length(AFvirtualfile *vfile)
{
AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
InputStream &is = afis.is;
return is.GetSize();
}
static AFfileoffset
audiofile_file_tell(AFvirtualfile *vfile)
{
AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
InputStream &is = afis.is;
return is.GetOffset();
}
static void
audiofile_file_destroy(AFvirtualfile *vfile)
{
assert(vfile->closure != nullptr);
vfile->closure = nullptr;
}
static AFfileoffset
audiofile_file_seek(AFvirtualfile *vfile, AFfileoffset _offset,
int is_relative)
{
AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
InputStream &is = afis.is;
InputStream::offset_type offset = _offset;
if (is_relative)
offset += is.GetOffset();
if (is.LockSeek(offset, IgnoreError())) {
return is.GetOffset();
} else {
return -1;
}
}
static AFvirtualfile *
setup_virtual_fops(AudioFileInputStream &afis)
{
AFvirtualfile *vf = new AFvirtualfile();
vf->closure = &afis;
vf->write = nullptr;
vf->read = audiofile_file_read;
vf->length = audiofile_file_length;
vf->destroy = audiofile_file_destroy;
vf->seek = audiofile_file_seek;
vf->tell = audiofile_file_tell;
return vf;
}
static SampleFormat
audiofile_bits_to_sample_format(int bits)
{
switch (bits) {
case 8:
return SampleFormat::S8;
case 16:
return SampleFormat::S16;
case 24:
return SampleFormat::S24_P32;
case 32:
return SampleFormat::S32;
}
return SampleFormat::UNDEFINED;
}
static SampleFormat
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))) {
FormatDebug(audiofile_domain,
"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 void
audiofile_stream_decode(Decoder &decoder, InputStream &is)
{
if (!is.IsSeekable()) {
LogWarning(audiofile_domain, "not seekable");
return;
}
AudioFileInputStream afis{&decoder, is};
AFvirtualfile *const vf = setup_virtual_fops(afis);
const AFfilehandle fh = afOpenVirtualFile(vf, "r", nullptr);
if (fh == AF_NULL_FILEHANDLE)
return;
Error error;
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format,
afGetRate(fh, AF_DEFAULT_TRACK),
audiofile_setup_sample_format(fh),
afGetVirtualChannels(fh, AF_DEFAULT_TRACK),
error)) {
LogError(error);
afCloseFile(fh);
return;
}
const double total_time = audiofile_get_duration(fh);
const uint16_t kbit_rate = (uint16_t)
(is.GetSize() * 8.0 / total_time / 1000.0 + 0.5);
const unsigned frame_size = (unsigned)
afGetVirtualFrameSize(fh, AF_DEFAULT_TRACK, true);
decoder_initialized(decoder, audio_format, true, total_time);
DecoderCommand cmd;
do {
/* pick 1020 since its divisible for 8,16,24, and
32-bit audio */
char chunk[1020];
const int nframes =
afReadFrames(fh, AF_DEFAULT_TRACK, chunk,
sizeof(chunk) / frame_size);
if (nframes <= 0)
break;
cmd = decoder_data(decoder, nullptr,
chunk, nframes * frame_size,
kbit_rate);
if (cmd == DecoderCommand::SEEK) {
AFframecount frame = decoder_seek_where(decoder) *
audio_format.sample_rate;
afSeekFrame(fh, AF_DEFAULT_TRACK, frame);
decoder_command_finished(decoder);
cmd = DecoderCommand::NONE;
}
} while (cmd == DecoderCommand::NONE);
afCloseFile(fh);
}
gcc_pure
static int
audiofile_get_duration(InputStream &is)
{
if (!is.IsSeekable())
return -1;
AudioFileInputStream afis{nullptr, is};
AFvirtualfile *vf = setup_virtual_fops(afis);
AFfilehandle fh = afOpenVirtualFile(vf, "r", nullptr);
if (fh == AF_NULL_FILEHANDLE)
return -1;
int duration = int(audiofile_get_duration(fh));
afCloseFile(fh);
return duration;
}
static bool
audiofile_scan_stream(InputStream &is,
const struct tag_handler *handler, void *handler_ctx)
{
int total_time = audiofile_get_duration(is);
if (total_time < 0)
return false;
tag_handler_invoke_duration(handler, handler_ctx, total_time);
return true;
}
static const char *const audiofile_suffixes[] = {
"wav", "au", "aiff", "aif", nullptr
};
static const char *const audiofile_mime_types[] = {
"audio/x-wav",
"audio/x-aiff",
nullptr
};
const struct DecoderPlugin audiofile_decoder_plugin = {
"audiofile",
audiofile_init,
nullptr,
audiofile_stream_decode,
nullptr,
nullptr,
audiofile_scan_stream,
nullptr,
audiofile_suffixes,
audiofile_mime_types,
};
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