/*
* Squeezelite - lightweight headless squeezebox emulator
*
* (c) Adrian Smith 2012-2015, triode1@btinternet.com
* Ralph Irving 2015-2017, ralph_irving@hotmail.com
* Philippe, philippe_44@outlook.com
*
* 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 3 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, see .
*
*/
#include "squeezelite.h"
#include
// AAC_MAX_SAMPLES is the number of samples for one channel
#define FRAME_BUF (AAC_MAX_NSAMPS*2)
#if BYTES_PER_FRAME == 4
#define ALIGN(n) (n)
#else
#define ALIGN(n) (n << 16)
#endif
#define WRAPBUF_LEN 2048
static unsigned rates[] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350 };
struct chunk_table {
u32_t sample, offset;
};
struct helixaac {
HAACDecoder hAac;
u8_t type;
u8_t *write_buf;
u8_t *wrap_buf;
// following used for mp4 only
u32_t consume;
u32_t pos;
u32_t sample;
u32_t nextchunk;
void *stsc;
u32_t skip;
u64_t samples;
u64_t sttssamples;
bool empty;
struct chunk_table *chunkinfo;
#if !LINKALL
#endif
};
static struct helixaac *a;
extern log_level loglevel;
extern struct buffer *streambuf;
extern struct buffer *outputbuf;
extern struct streamstate stream;
extern struct outputstate output;
extern struct decodestate decode;
extern struct processstate process;
#define LOCK_S mutex_lock(streambuf->mutex)
#define UNLOCK_S mutex_unlock(streambuf->mutex)
#define LOCK_O mutex_lock(outputbuf->mutex)
#define UNLOCK_O mutex_unlock(outputbuf->mutex)
#if PROCESS
#define LOCK_O_direct if (decode.direct) mutex_lock(outputbuf->mutex)
#define UNLOCK_O_direct if (decode.direct) mutex_unlock(outputbuf->mutex)
#define IF_DIRECT(x) if (decode.direct) { x }
#define IF_PROCESS(x) if (!decode.direct) { x }
#else
#define LOCK_O_direct mutex_lock(outputbuf->mutex)
#define UNLOCK_O_direct mutex_unlock(outputbuf->mutex)
#define IF_DIRECT(x) { x }
#define IF_PROCESS(x)
#endif
#if LINKALL
#define HAAC(h, fn, ...) (AAC ## fn)(__VA_ARGS__)
#else
#define HAAC(h, fn, ...) (h)->AAC##fn(__VA_ARGS__)
#endif
// minimal code for mp4 file parsing to extract audio config and find media data
// adapted from faad2/common/mp4ff
u32_t mp4_desc_length(u8_t **buf) {
u8_t b;
u8_t num_bytes = 0;
u32_t length = 0;
do {
b = **buf;
*buf += 1;
num_bytes++;
length = (length << 7) | (b & 0x7f);
} while ((b & 0x80) && num_bytes < 4);
return length;
}
// read mp4 header to extract config data
static int read_mp4_header(unsigned long *samplerate_p, unsigned char *channels_p) {
size_t bytes = min(_buf_used(streambuf), _buf_cont_read(streambuf));
char type[5];
u32_t len;
while (bytes >= 8) {
// count trak to find the first playable one
static unsigned trak, play;
u32_t consume;
len = unpackN((u32_t *)streambuf->readp);
memcpy(type, streambuf->readp + 4, 4);
type[4] = '\0';
if (!strcmp(type, "moov")) {
trak = 0;
play = 0;
}
if (!strcmp(type, "trak")) {
trak++;
}
// extract audio config from within esds and pass to DecInit2
if (!strcmp(type, "esds") && bytes > len) {
u8_t *ptr = streambuf->readp + 12;
AACFrameInfo info;
if (*ptr++ == 0x03) {
mp4_desc_length(&ptr);
ptr += 4;
} else {
ptr += 3;
}
mp4_desc_length(&ptr);
ptr += 13;
if (*ptr++ != 0x05) {
LOG_WARN("error parsing esds");
return -1;
}
int desc_len = mp4_desc_length(&ptr);
int AOT = *ptr >> 3;
info.profile = AAC_PROFILE_LC;
info.sampRateCore = (*ptr++ & 0x07) << 1;
info.sampRateCore |= (*ptr >> 7) & 0x01;
info.sampRateCore = rates[info.sampRateCore];
info.nChans = (*ptr & 0x7f) >> 3;
*channels_p = info.nChans;
// Note that 24 bits frequencies are not handled
#if AAC_ENABLE_SBR
if (AOT == 5 || AOT == 29) {
*samplerate_p = rates[((ptr[0] & 0x03) << 1) | (ptr[1] >> 7)];
LOG_WARN("AAC stream with SBR => high CPU required (use LMS proxied mode)");
} else if (desc_len > 2 && ((ptr[1] << 3) | (ptr[2] >> 5)) == 0x2b7 && (ptr[2] & 0x1f) == 0x05 && (ptr[3] & 0x80)) {
*samplerate_p = rates[(ptr[3] & 0x78) >> 3];
LOG_WARN("AAC stream with extended SBR => high CPU required (use LMS proxied mode)");
} else if (AOT == 2) {
*samplerate_p = info.sampRateCore;
} else {
*samplerate_p = 44100;
LOG_ERROR("AAC audio object type %d not handled", AOT);
}
#else
*samplerate_p = info.sampRateCore;
#endif
HAAC(a, SetRawBlockParams, a->hAac, 0, &info);
LOG_DEBUG("playable aac track: %u (p:%x, r:%d, c:%d, desc_len:%d)", trak, AOT, info.sampRateCore, info.nChans, desc_len);
play = trak;
}
// extract the total number of samples from stts
if (!strcmp(type, "stts") && bytes > len) {
u32_t i;
u8_t *ptr = streambuf->readp + 12;
u32_t entries = unpackN((u32_t *)ptr);
ptr += 4;
for (i = 0; i < entries; ++i) {
u32_t count = unpackN((u32_t *)ptr);
u32_t size = unpackN((u32_t *)(ptr + 4));
a->sttssamples += count * size;
ptr += 8;
}
LOG_DEBUG("total number of samples contained in stts: " FMT_u64, a->sttssamples);
}
// stash sample to chunk info, assume it comes before stco
if (!strcmp(type, "stsc") && bytes > len && !a->chunkinfo) {
a->stsc = malloc(len - 12);
if (a->stsc == NULL) {
LOG_WARN("malloc fail");
return -1;
}
memcpy(a->stsc, streambuf->readp + 12, len - 12);
}
// build offsets table from stco and stored stsc
if (!strcmp(type, "stco") && bytes > len && play == trak) {
u32_t i;
// extract chunk offsets
u8_t *ptr = streambuf->readp + 12;
u32_t entries = unpackN((u32_t *)ptr);
ptr += 4;
a->chunkinfo = malloc(sizeof(struct chunk_table) * (entries + 1));
if (a->chunkinfo == NULL) {
LOG_WARN("malloc fail");
return -1;
}
for (i = 0; i < entries; ++i) {
a->chunkinfo[i].offset = unpackN((u32_t *)ptr);
a->chunkinfo[i].sample = 0;
ptr += 4;
}
a->chunkinfo[i].sample = 0;
a->chunkinfo[i].offset = 0;
// fill in first sample id for each chunk from stored stsc
if (a->stsc) {
u32_t stsc_entries = unpackN((u32_t *)a->stsc);
u32_t sample = 0;
u32_t last = 0, last_samples = 0;
u8_t *ptr = (u8_t *)a->stsc + 4;
while (stsc_entries--) {
u32_t first = unpackN((u32_t *)ptr);
u32_t samples = unpackN((u32_t *)(ptr + 4));
if (last) {
for (i = last - 1; i < first - 1; ++i) {
a->chunkinfo[i].sample = sample;
sample += last_samples;
}
}
if (stsc_entries == 0) {
for (i = first - 1; i < entries; ++i) {
a->chunkinfo[i].sample = sample;
sample += samples;
}
}
last = first;
last_samples = samples;
ptr += 12;
}
free(a->stsc);
a->stsc = NULL;
}
}
// found media data, advance to start of first chunk and return
if (!strcmp(type, "mdat")) {
_buf_inc_readp(streambuf, 8);
a->pos += 8;
bytes -= 8;
if (play) {
LOG_DEBUG("type: mdat len: %u pos: %u", len, a->pos);
if (a->chunkinfo && a->chunkinfo[0].offset > a->pos) {
u32_t skip = a->chunkinfo[0].offset - a->pos;
LOG_DEBUG("skipping: %u", skip);
if (skip <= bytes) {
_buf_inc_readp(streambuf, skip);
a->pos += skip;
} else {
a->consume = skip;
}
}
a->sample = a->nextchunk = 1;
return 1;
} else {
LOG_DEBUG("type: mdat len: %u, no playable track found", len);
return -1;
}
}
// parse key-value atoms within ilst ---- entries to get encoder padding within iTunSMPB entry for gapless
if (!strcmp(type, "----") && bytes > len) {
u8_t *ptr = streambuf->readp + 8;
u32_t remain = len - 8, size;
if (!memcmp(ptr + 4, "mean", 4) && (size = unpackN((u32_t *)ptr)) < remain) {
ptr += size; remain -= size;
}
if (!memcmp(ptr + 4, "name", 4) && (size = unpackN((u32_t *)ptr)) < remain && !memcmp(ptr + 12, "iTunSMPB", 8)) {
ptr += size; remain -= size;
}
if (!memcmp(ptr + 4, "data", 4) && remain > 16 + 48) {
// data is stored as hex strings: 0 start end samples
u32_t b, c; u64_t d;
if (sscanf((const char *)(ptr + 16), "%x %x %x " FMT_x64, &b, &b, &c, &d) == 4) {
LOG_DEBUG("iTunSMPB start: %u end: %u samples: " FMT_u64, b, c, d);
if (a->sttssamples && a->sttssamples < b + c + d) {
LOG_DEBUG("reducing samples as stts count is less");
d = a->sttssamples - (b + c);
}
a->skip = b;
a->samples = d;
}
}
}
// default to consuming entire box
consume = len;
// read into these boxes so reduce consume
if (!strcmp(type, "moov") || !strcmp(type, "trak") || !strcmp(type, "mdia") || !strcmp(type, "minf") || !strcmp(type, "stbl") ||
!strcmp(type, "udta") || !strcmp(type, "ilst")) {
consume = 8;
}
// special cases which mix mix data in the enclosing box which we want to read into
if (!strcmp(type, "stsd")) consume = 16;
if (!strcmp(type, "mp4a")) consume = 36;
if (!strcmp(type, "meta")) consume = 12;
// consume rest of box if it has been parsed (all in the buffer) or is not one we want to parse
if (bytes >= consume) {
LOG_DEBUG("type: %s len: %u consume: %u", type, len, consume);
_buf_inc_readp(streambuf, consume);
a->pos += consume;
bytes -= consume;
} else if ( !(!strcmp(type, "esds") || !strcmp(type, "stts") || !strcmp(type, "stsc") ||
!strcmp(type, "stco") || !strcmp(type, "----")) ) {
LOG_DEBUG("type: %s len: %u consume: %u - partial consume: %u", type, len, consume, bytes);
_buf_inc_readp(streambuf, bytes);
a->pos += bytes;
a->consume = consume - bytes;
break;
} else if (len > streambuf->size) {
// can't process an atom larger than streambuf!
LOG_ERROR("atom %s too large for buffer %u %u", type, len, streambuf->size);
return -1;
} else {
// make sure there is 'len' contiguous space
_buf_unwrap(streambuf, len);
break;
}
}
return 0;
}
static decode_state helixaac_decode(void) {
size_t bytes_total, bytes_wrap;
int res, bytes;
static AACFrameInfo info;
s16_t *iptr;
u8_t *sptr;
bool endstream;
frames_t frames;
LOCK_S;
bytes_total = _buf_used(streambuf);
bytes_wrap = min(bytes_total, _buf_cont_read(streambuf));
if (stream.state <= DISCONNECT && !bytes_total) {
UNLOCK_S;
return DECODE_COMPLETE;
}
if (a->consume) {
u32_t consume = min(a->consume, bytes_wrap);
LOG_DEBUG("consume: %u of %u", consume, a->consume);
_buf_inc_readp(streambuf, consume);
a->pos += consume;
a->consume -= consume;
UNLOCK_S;
return DECODE_RUNNING;
}
if (decode.new_stream) {
int found = 0;
static unsigned char channels;
static unsigned long samplerate;
if (a->type == '2') {
// adts stream - seek for header
long n = HAAC(a, FindSyncWord, streambuf->readp, bytes_wrap);
LOG_DEBUG("Sync search in %d bytes %d", bytes_wrap, n);
if (n >= 0) {
u8_t *p = streambuf->readp + n;
int bytes = bytes_wrap - n;
if (!HAAC(a, Decode, a->hAac, &p, &bytes, (s16_t*) a->write_buf)) {
HAAC(a, GetLastFrameInfo, a->hAac, &info);
channels = info.nChans;
samplerate = info.sampRateOut;
found = 1;
} else if (n == 0) n++;
HAAC(a, FlushCodec, a->hAac);
bytes_total -= n;
bytes_wrap -= n;
_buf_inc_readp(streambuf, n);
} else {
found = -1;
}
} else {
// mp4 - read header
found = read_mp4_header(&samplerate, &channels);
}
if (found == 1) {
LOCK_O;
output.next_sample_rate = decode_newstream(samplerate, output.supported_rates);
IF_DSD( output.next_fmt = PCM; )
output.track_start = outputbuf->writep;
if (output.fade_mode) _checkfade(true);
decode.new_stream = false;
UNLOCK_O;
LOG_INFO("setting track start, samplerate: %u channels: %u", samplerate, channels);
bytes_total = _buf_used(streambuf);
bytes_wrap = min(bytes_total, _buf_cont_read(streambuf));
// come back later if we don' thave enough data
if (bytes_total < WRAPBUF_LEN) {
UNLOCK_S;
LOG_INFO("need more audio data");
return DECODE_RUNNING;
}
} else if (found == -1) {
LOG_WARN("error reading stream header");
UNLOCK_S;
return DECODE_ERROR;
} else {
// not finished header parsing come back next time
UNLOCK_S;
LOG_DEBUG("header not found yet");
return DECODE_RUNNING;
}
}
// we always have at least WRAPBUF_LEN unless it's the end of a stream
if (bytes_wrap < WRAPBUF_LEN && bytes_wrap != bytes_total) {
// build a linear buffer if we are crossing the end of streambuf
memcpy(a->wrap_buf, streambuf->readp, bytes_wrap);
memcpy(a->wrap_buf + bytes_wrap, streambuf->buf, min(WRAPBUF_LEN, bytes_total) - bytes_wrap);
sptr = a->wrap_buf;
bytes = bytes_wrap = min(WRAPBUF_LEN, bytes_total);
} else {
sptr = streambuf->readp;
bytes = bytes_wrap;
}
// decode function changes iptr, so can't use streambuf->readp (same for bytes)
res = HAAC(a, Decode, a->hAac, &sptr, &bytes, (s16_t*) a->write_buf);
if (res < 0) {
LOG_WARN("AAC decode error %d", res);
}
HAAC(a, GetLastFrameInfo, a->hAac, &info);
iptr = (s16_t*) a->write_buf;
bytes = bytes_wrap - bytes;
endstream = false;
if (a->chunkinfo && a->chunkinfo[a->nextchunk].offset && a->sample++ == a->chunkinfo[a->nextchunk].sample) {
// mp4 end of chunk - skip to next offset
if (a->chunkinfo[a->nextchunk].offset > a->pos) {
u32_t skip = a->chunkinfo[a->nextchunk].offset - a->pos;
if (skip != bytes) {
LOG_DEBUG("skipping to next chunk pos: %u consumed: %u != skip: %u", a->pos, bytes, skip);
}
if (bytes_total >= skip) {
_buf_inc_readp(streambuf, skip);
a->pos += skip;
} else {
a->consume = skip;
}
a->nextchunk++;
} else {
LOG_ERROR("error: need to skip backwards!");
endstream = true;
}
} else if (bytes > 0) {
// adts and mp4 when not at end of chunk
_buf_inc_readp(streambuf, bytes);
a->pos += bytes;
} else {
// error which doesn't advance streambuf - end
endstream = true;
}
UNLOCK_S;
if (endstream) {
LOG_WARN("unable to decode further");
return DECODE_ERROR;
}
if (!info.outputSamps) {
a->empty = true;
return DECODE_RUNNING;
}
frames = info.outputSamps / info.nChans;
if (a->skip) {
u32_t skip;
if (a->empty) {
a->empty = false;
a->skip -= frames;
LOG_DEBUG("gapless: first frame empty, skipped %u frames at start", frames);
}
skip = min(frames, a->skip);
LOG_DEBUG("gapless: skipping %u frames at start", skip);
frames -= skip;
a->skip -= skip;
iptr += skip * info.nChans;
}
if (a->samples) {
if (a->samples < frames) {
LOG_DEBUG("gapless: trimming %u frames from end", frames - a->samples);
frames = (frames_t)a->samples;
}
a->samples -= frames;
}
LOG_SDEBUG("write %u frames", frames);
LOCK_O_direct;
while (frames > 0) {
frames_t f;
frames_t count;
ISAMPLE_T *optr;
IF_DIRECT(
f = _buf_cont_write(outputbuf) / BYTES_PER_FRAME;
optr = (ISAMPLE_T *)outputbuf->writep;
);
IF_PROCESS(
f = process.max_in_frames;
optr = (ISAMPLE_T *)process.inbuf;
);
f = min(f, frames);
count = f;
if (info.nChans == 2) {
#if BYTES_PER_FRAME == 4
memcpy(optr, iptr, count * BYTES_PER_FRAME);
iptr += count * 2;
#else
while (count--) {
*optr++ = ALIGN(*iptr++);
*optr++ = ALIGN(*iptr++);
}
#endif
} else if (info.nChans == 1) {
while (count--) {
*optr++ = ALIGN(*iptr);
*optr++ = ALIGN(*iptr++);
}
} else {
LOG_WARN("unsupported number of channels");
}
frames -= f;
IF_DIRECT(
_buf_inc_writep(outputbuf, f * BYTES_PER_FRAME);
);
IF_PROCESS(
process.in_frames = f;
if (frames) LOG_ERROR("unhandled case");
);
}
UNLOCK_O_direct;
return DECODE_RUNNING;
}
static void helixaac_open(u8_t size, u8_t rate, u8_t chan, u8_t endianness) {
LOG_INFO("opening %s stream", size == '2' ? "adts" : "mp4");
a->type = size;
a->pos = a->consume = a->sample = a->nextchunk = 0;
if (a->chunkinfo) {
free(a->chunkinfo);
}
if (a->stsc) {
free(a->stsc);
}
a->chunkinfo = NULL;
a->stsc = NULL;
a->skip = 0;
a->samples = 0;
a->sttssamples = 0;
a->empty = false;
if (a->hAac) {
// always free decoder as flush only works when no parameter has changed
HAAC(a, FreeDecoder, a->hAac);
} else {
a->write_buf = malloc(FRAME_BUF * 4);
a->wrap_buf = malloc(WRAPBUF_LEN);
}
a->hAac = HAAC(a, InitDecoder);
}
static void helixaac_close(void) {
HAAC(a, FreeDecoder, a->hAac);
a->hAac = NULL;
if (a->chunkinfo) {
free(a->chunkinfo);
a->chunkinfo = NULL;
}
if (a->stsc) {
free(a->stsc);
a->stsc = NULL;
}
free(a->write_buf);
free(a->wrap_buf);
}
static bool load_helixaac() {
#if !LINKALL
void *handle = dlopen(LIBHELIX-AAC, RTLD_NOW);
char *err;
if (!handle) {
LOG_INFO("dlerror: %s", dlerror());
return false;
}
// load symbols here
if ((err = dlerror()) != NULL) {
LOG_INFO("dlerror: %s", err);
return false;
}
LOG_INFO("loaded "LIBHELIX-AAC"");
#endif
return true;
}
struct codec *register_helixaac(void) {
static struct codec ret = {
'a', // id
"aac", // types
WRAPBUF_LEN, // min read
20480, // min space
helixaac_open, // open
helixaac_close, // close
helixaac_decode, // decode
};
a = malloc(sizeof(struct helixaac));
if (!a) {
return NULL;
}
a->hAac = NULL;
a->chunkinfo = NULL;
a->stsc = NULL;
if (!load_helixaac()) {
return NULL;
}
LOG_INFO("using helix-aac to decode aac");
return &ret;
}