/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../inputPlugin.h"
#ifdef HAVE_FAAD
#define AAC_MAX_CHANNELS 6
#include "../utils.h"
#include "../log.h"
#include <faad.h>
/* all code here is either based on or copied from FAAD2's frontend code */
typedef struct {
InputStream *inStream;
size_t bytesIntoBuffer;
size_t bytesConsumed;
off_t fileOffset;
unsigned char *buffer;
int atEof;
} AacBuffer;
static void aac_buffer_shift(AacBuffer * b, size_t length)
{
assert(length >= b->bytesConsumed);
assert(length <= b->bytesConsumed + b->bytesIntoBuffer);
memmove(b->buffer, b->buffer + length,
b->bytesConsumed + b->bytesIntoBuffer - length);
length -= b->bytesConsumed;
b->bytesConsumed = 0;
b->bytesIntoBuffer -= length;
}
static void fillAacBuffer(AacBuffer * b)
{
size_t bread;
if (b->bytesIntoBuffer >= FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS)
/* buffer already full */
return;
aac_buffer_shift(b, b->bytesConsumed);
if (!b->atEof) {
size_t rest = FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS -
b->bytesIntoBuffer;
bread = readFromInputStream(b->inStream,
(void *)(b->buffer +
b->
bytesIntoBuffer),
1, rest);
if (bread == 0 && inputStreamAtEOF(b->inStream))
b->atEof = 1;
b->bytesIntoBuffer += bread;
}
if ((b->bytesIntoBuffer > 3 && memcmp(b->buffer, "TAG", 3) == 0) ||
(b->bytesIntoBuffer > 11 &&
memcmp(b->buffer, "LYRICSBEGIN", 11) == 0) ||
(b->bytesIntoBuffer > 8 && memcmp(b->buffer, "APETAGEX", 8) == 0))
b->bytesIntoBuffer = 0;
}
static void advanceAacBuffer(AacBuffer * b, size_t bytes)
{
b->fileOffset += bytes;
b->bytesConsumed = bytes;
b->bytesIntoBuffer -= bytes;
}
static int adtsSampleRates[] =
{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
/**
* Check whether the buffer head is an AAC frame, and return the frame
* length. Returns 0 if it is not a frame.
*/
static size_t adts_check_frame(AacBuffer * b)
{
if (b->bytesIntoBuffer <= 7)
return 0;
/* check syncword */
if (!((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)))
return 0;
return (((unsigned int)b->buffer[3] & 0x3) << 11) |
(((unsigned int)b->buffer[4]) << 3) |
(b->buffer[5] >> 5);
}
/**
* Find the next AAC frame in the buffer. Returns 0 if no frame is
* found or if not enough data is available.
*/
static size_t adts_find_frame(AacBuffer * b)
{
const unsigned char *p;
size_t frame_length;
while ((p = memchr(b->buffer, 0xff, b->bytesIntoBuffer)) != NULL) {
/* discard data before 0xff */
if (p > b->buffer)
aac_buffer_shift(b, p - b->buffer);
if (b->bytesIntoBuffer <= 7)
/* not enough data yet */
return 0;
/* is it a frame? */
frame_length = adts_check_frame(b);
if (frame_length > 0)
/* yes, it is */
return frame_length;
/* it's just some random 0xff byte; discard and and
continue searching */
aac_buffer_shift(b, 1);
}
/* nothing at all; discard the whole buffer */
aac_buffer_shift(b, b->bytesIntoBuffer);
return 0;
}
static void adtsParse(AacBuffer * b, float *length)
{
unsigned int frames, frameLength;
int sampleRate = 0;
float framesPerSec;
/* Read all frames to ensure correct time and bitrate */
for (frames = 0;; frames++) {
fillAacBuffer(b);
frameLength = adts_find_frame(b);
if (frameLength > 0) {
if (frames == 0) {
sampleRate = adtsSampleRates[(b->
buffer[2] & 0x3c)
>> 2];
}
if (frameLength > b->bytesIntoBuffer)
break;
advanceAacBuffer(b, frameLength);
} else
break;
}
framesPerSec = (float)sampleRate / 1024.0;
if (framesPerSec != 0)
*length = (float)frames / framesPerSec;
}
static void initAacBuffer(InputStream * inStream, AacBuffer * b)
{
memset(b, 0, sizeof(AacBuffer));
b->inStream = inStream;
b->buffer = xmalloc(FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
memset(b->buffer, 0, FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
}
static void aac_parse_header(AacBuffer * b, float *length)
{
size_t fileread;
size_t tagsize;
if (length)
*length = -1;
fileread = b->inStream->size;
fillAacBuffer(b);
tagsize = 0;
if (b->bytesIntoBuffer >= 10 && !memcmp(b->buffer, "ID3", 3)) {
tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) |
(b->buffer[8] << 7) | (b->buffer[9] << 0);
tagsize += 10;
advanceAacBuffer(b, tagsize);
fillAacBuffer(b);
}
if (length == NULL)
return;
if (b->bytesIntoBuffer >= 2 &&
(b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
adtsParse(b, length);
seekInputStream(b->inStream, tagsize, SEEK_SET);
b->bytesIntoBuffer = 0;
b->bytesConsumed = 0;
b->fileOffset = tagsize;
fillAacBuffer(b);
} else if (memcmp(b->buffer, "ADIF", 4) == 0) {
int bitRate;
int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
bitRate =
((unsigned int)(b->
buffer[4 +
skipSize] & 0x0F) << 19) | ((unsigned
int)b->
buffer[5
+
skipSize]
<< 11) |
((unsigned int)b->
buffer[6 + skipSize] << 3) | ((unsigned int)b->buffer[7 +
skipSize]
& 0xE0);
if (fileread != 0 && bitRate != 0)
*length = fileread * 8.0 / bitRate;
else
*length = fileread;
}
}
static float getAacFloatTotalTime(char *file)
{
AacBuffer b;
float length;
faacDecHandle decoder;
faacDecConfigurationPtr config;
uint32_t sampleRate;
unsigned char channels;
InputStream inStream;
long bread;
if (openInputStream(&inStream, file) < 0)
return -1;
initAacBuffer(&inStream, &b);
aac_parse_header(&b, &length);
if (length < 0) {
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
faacDecSetConfiguration(decoder, config);
fillAacBuffer(&b);
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
&sampleRate, &channels);
#else
bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
#endif
if (bread >= 0 && sampleRate > 0 && channels > 0)
length = 0;
faacDecClose(decoder);
}
if (b.buffer)
free(b.buffer);
closeInputStream(&inStream);
return length;
}
static int getAacTotalTime(char *file)
{
int file_time = -1;
float length;
if ((length = getAacFloatTotalTime(file)) >= 0)
file_time = length + 0.5;
return file_time;
}
static int aac_decode(char *path)
{
float file_time;
float totalTime;
faacDecHandle decoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
long bread;
uint32_t sampleRate;
unsigned char channels;
unsigned int sampleCount;
char *sampleBuffer;
size_t sampleBufferLen;
/*float * seekTable;
long seekTableEnd = -1;
int seekPositionFound = 0; */
mpd_uint16 bitRate = 0;
AacBuffer b;
InputStream inStream;
if ((totalTime = getAacFloatTotalTime(path)) < 0)
return -1;
if (openInputStream(&inStream, path) < 0)
return -1;
initAacBuffer(&inStream, &b);
aac_parse_header(&b, NULL);
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
config->downMatrix = 1;
#endif
#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
config->dontUpSampleImplicitSBR = 0;
#endif
faacDecSetConfiguration(decoder, config);
fillAacBuffer(&b);
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
&sampleRate, &channels);
#else
bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
#endif
if (bread < 0) {
ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
if (b.buffer)
free(b.buffer);
return -1;
}
dc.audio_format.bits = 16;
dc.total_time = totalTime;
file_time = 0.0;
advanceAacBuffer(&b, bread);
while (1) {
fillAacBuffer(&b);
if (b.bytesIntoBuffer == 0)
break;
#ifdef HAVE_FAAD_BUFLEN_FUNCS
sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer,
b.bytesIntoBuffer);
#else
sampleBuffer = faacDecDecode(decoder, &frameInfo, b.buffer);
#endif
if (frameInfo.error > 0) {
ERROR("error decoding AAC file: %s\n", path);
ERROR("faad2 error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
sampleRate = frameInfo.samplerate;
#endif
dc.audio_format.channels = frameInfo.channels;
dc.audio_format.sampleRate = sampleRate;
advanceAacBuffer(&b, frameInfo.bytesconsumed);
sampleCount = (unsigned long)(frameInfo.samples);
if (sampleCount > 0) {
bitRate = frameInfo.bytesconsumed * 8.0 *
frameInfo.channels * sampleRate /
frameInfo.samples / 1000 + 0.5;
file_time +=
(float)(frameInfo.samples) / frameInfo.channels /
sampleRate;
}
sampleBufferLen = sampleCount * 2;
switch (ob_send(sampleBuffer, sampleBufferLen,
file_time, bitRate, NULL)) {
case DC_ACTION_NONE: break;
case DC_ACTION_SEEK:
/*
* this plugin doesn't support seek because nobody
* has bothered, yet...
*/
dc_action_seek_fail(DC_SEEK_ERROR);
break;
default: goto out;
}
}
out:
faacDecClose(decoder);
if (b.buffer)
free(b.buffer);
return 0;
}
static MpdTag *aacTagDup(char *file)
{
MpdTag *ret = NULL;
int file_time = getAacTotalTime(file);
if (file_time >= 0) {
if ((ret = id3Dup(file)) == NULL)
ret = newMpdTag();
ret->time = file_time;
} else {
DEBUG("aacTagDup: Failed to get total song time from: %s\n",
file);
}
return ret;
}
static const char *aac_suffixes[] = { "aac", NULL };
static const char *aac_mimeTypes[] = { "audio/aac", "audio/aacp", NULL };
InputPlugin aacPlugin = {
"aac",
NULL,
NULL,
NULL,
NULL,
aac_decode,
aacTagDup,
INPUT_PLUGIN_STREAM_FILE,
aac_suffixes,
aac_mimeTypes
};
#else
InputPlugin aacPlugin;
#endif /* HAVE_FAAD */