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pcm.c
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pcm.c
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/*
* Squeezelite - lightweight headless squeezebox emulator
*
* (c) Adrian Smith 2012-2015, triode1@btinternet.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 <http://www.gnu.org/licenses/>.
*
*/
#include "squeezelite.h"
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 LOCK_O_not_direct if (!decode.direct) mutex_lock(outputbuf->mutex)
#define UNLOCK_O_not_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 LOCK_O_not_direct
#define UNLOCK_O_not_direct
#define IF_DIRECT(x) { x }
#define IF_PROCESS(x)
#endif
#define MAX_DECODE_FRAMES 4096
static u32_t sample_rates[] = {
11025, 22050, 32000, 44100, 48000, 8000, 12000, 16000, 24000, 96000, 88200, 176400, 192000, 352800, 384000, 705600, 768000
};
static u32_t sample_rate;
static u32_t sample_size;
static u32_t channels;
static bool bigendian;
static bool limit;
static u32_t audio_left;
static u32_t bytes_per_frame;
typedef enum { UNKNOWN = 0, WAVE, AIFF } header_format;
static void _check_header(void) {
u8_t *ptr = streambuf->readp;
unsigned bytes = min(_buf_used(streambuf), _buf_cont_read(streambuf));
header_format format = UNKNOWN;
// simple parsing of wav and aiff headers and get to samples
if (bytes > 12) {
if (!memcmp(ptr, "RIFF", 4) && !memcmp(ptr+8, "WAVE", 4)) {
LOG_INFO("WAVE");
format = WAVE;
} else if (!memcmp(ptr, "FORM", 4) && (!memcmp(ptr+8, "AIFF", 4) || !memcmp(ptr+8, "AIFC", 4))) {
LOG_INFO("AIFF");
format = AIFF;
}
}
if (format != UNKNOWN) {
ptr += 12;
bytes -= 12;
while (bytes >= 8) {
char id[5];
unsigned len;
memcpy(id, ptr, 4);
id[4] = '\0';
if (format == WAVE) {
len = *(ptr+4) | *(ptr+5) << 8 | *(ptr+6) << 16| *(ptr+7) << 24;
} else {
len = *(ptr+4) << 24 | *(ptr+5) << 16 | *(ptr+6) << 8 | *(ptr+7);
}
LOG_INFO("header: %s len: %d", id, len);
if (format == WAVE && !memcmp(ptr, "data", 4)) {
ptr += 8;
_buf_inc_readp(streambuf, ptr - streambuf->readp);
// len could be wrong for stream, reading up to EOF.
if (stream.state == STREAMING_FILE){
audio_left = len;
LOG_INFO("audio size: %u", audio_left);
limit = true;
}else {
LOG_INFO("unknown or invalid audio size, reading up to EOF");
}
return;
}
if (format == AIFF && !memcmp(ptr, "SSND", 4) && bytes >= 16) {
unsigned offset = *(ptr+8) << 24 | *(ptr+9) << 16 | *(ptr+10) << 8 | *(ptr+11);
// following 4 bytes is blocksize - ignored
ptr += 8 + 8;
_buf_inc_readp(streambuf, ptr + offset - streambuf->readp);
// len could be wrong for stream, reading up to EOF.
if (stream.state == STREAMING_FILE){
audio_left = len - 8 - offset;
LOG_INFO("audio size: %u", audio_left);
limit = true;
} else {
LOG_INFO("unknown or invalid audio size, reading up to EOF");
}
return;
}
if (format == WAVE && !memcmp(ptr, "fmt ", 4) && bytes >= 24) {
// override the server parsed values with our own
channels = *(ptr+10) | *(ptr+11) << 8;
sample_rate = *(ptr+12) | *(ptr+13) << 8 | *(ptr+14) << 16 | *(ptr+15) << 24;
sample_size = (*(ptr+22) | *(ptr+23) << 8) / 8;
bigendian = 0;
LOG_INFO("pcm size: %u rate: %u chan: %u bigendian: %u", sample_size, sample_rate, channels, bigendian);
}
if (format == AIFF && !memcmp(ptr, "COMM", 4) && bytes >= 26) {
int exponent;
// override the server parsed values with our own
channels = *(ptr+8) << 8 | *(ptr+9);
sample_size = (*(ptr+14) << 8 | *(ptr+15)) / 8;
bigendian = 1;
// sample rate is encoded as IEEE 80 bit extended format
// make some assumptions to simplify processing - only use first 32 bits of mantissa
exponent = ((*(ptr+16) & 0x7f) << 8 | *(ptr+17)) - 16383 - 31;
sample_rate = *(ptr+18) << 24 | *(ptr+19) << 16 | *(ptr+20) << 8 | *(ptr+21);
while (exponent < 0) { sample_rate >>= 1; ++exponent; }
while (exponent > 0) { sample_rate <<= 1; --exponent; }
LOG_INFO("pcm size: %u rate: %u chan: %u bigendian: %u", sample_size, sample_rate, channels, bigendian);
}
if (bytes >= len + 8) {
ptr += len + 8;
bytes -= (len + 8);
} else {
LOG_WARN("run out of data");
return;
}
}
} else if (stream.state == STREAMING_FILE) {
LOG_WARN("unknown format - can't parse header in file, fault back to server parameters");
} else {
LOG_INFO("unknown format - can't parse header in stream, using server parameters");
}
}
static decode_state pcm_decode(void) {
unsigned bytes, in, out;
frames_t frames, count;
u32_t *optr;
u8_t *iptr;
u8_t tmp[16];
LOCK_S;
// mc2 - 05/09/2015
// When Wav or Aiff is streamed the header is preserved, so we
// could verify the real file forma. Usefull when resampling
// from server, in that case server parameters are wrong.
//if (decode.new_stream && stream.state == STREAMING_FILE) {
if (decode.new_stream) {
_check_header();
}
LOCK_O_direct;
bytes = min(_buf_used(streambuf), _buf_cont_read(streambuf));
IF_DIRECT(
out = min(_buf_space(outputbuf), _buf_cont_write(outputbuf)) / BYTES_PER_FRAME;
);
IF_PROCESS(
out = process.max_in_frames;
);
if ((stream.state <= DISCONNECT && bytes == 0) || (limit && audio_left == 0)) {
UNLOCK_O_direct;
UNLOCK_S;
return DECODE_COMPLETE;
}
if (decode.new_stream) {
LOG_INFO("setting track_start");
LOCK_O_not_direct;
output.track_start = outputbuf->writep;
decode.new_stream = false;
#if DSD
if (sample_size == 3 &&
is_stream_dop(((u8_t *)streambuf->readp) + (bigendian?0:2),
((u8_t *)streambuf->readp) + (bigendian?0:2) + sample_size,
sample_size * channels, bytes / (sample_size * channels))) {
LOG_INFO("file contains DOP");
if (output.dsdfmt == DOP_S24_LE || output.dsdfmt == DOP_S24_3LE)
output.next_fmt = output.dsdfmt;
else
output.next_fmt = DOP;
output.next_sample_rate = sample_rate;
output.fade = FADE_INACTIVE;
} else {
output.next_sample_rate = decode_newstream(sample_rate, output.supported_rates);
output.next_fmt = PCM;
if (output.fade_mode) _checkfade(true);
}
#else
output.next_sample_rate = decode_newstream(sample_rate, output.supported_rates);
if (output.fade_mode) _checkfade(true);
#endif
UNLOCK_O_not_direct;
IF_PROCESS(
out = process.max_in_frames;
);
bytes_per_frame = channels * sample_size;
}
IF_DIRECT(
optr = (u32_t *)outputbuf->writep;
);
IF_PROCESS(
optr = (u32_t *)process.inbuf;
);
iptr = (u8_t *)streambuf->readp;
in = bytes / bytes_per_frame;
// handle frame wrapping round end of streambuf
// - only need if resizing of streambuf does not avoid this, could occur in localfile case
if (in == 0 && bytes > 0 && _buf_used(streambuf) >= bytes_per_frame) {
memcpy(tmp, iptr, bytes);
memcpy(tmp + bytes, streambuf->buf, bytes_per_frame - bytes);
iptr = tmp;
in = 1;
}
frames = min(in, out);
frames = min(frames, MAX_DECODE_FRAMES);
if (limit && frames * bytes_per_frame > audio_left) {
LOG_INFO("reached end of audio");
frames = audio_left / bytes_per_frame;
}
count = frames * channels;
if (channels == 2) {
if (sample_size == 1) {
while (count--) {
*optr++ = *iptr++ << 24;
}
} else if (sample_size == 2) {
if (bigendian) {
while (count--) {
*optr++ = *(iptr) << 24 | *(iptr+1) << 16;
iptr += 2;
}
} else {
while (count--) {
*optr++ = *(iptr) << 16 | *(iptr+1) << 24;
iptr += 2;
}
}
} else if (sample_size == 3) {
if (bigendian) {
while (count--) {
*optr++ = *(iptr) << 24 | *(iptr+1) << 16 | *(iptr+2) << 8;
iptr += 3;
}
} else {
while (count--) {
*optr++ = *(iptr) << 8 | *(iptr+1) << 16 | *(iptr+2) << 24;
iptr += 3;
}
}
} else if (sample_size == 4) {
if (bigendian) {
while (count--) {
*optr++ = *(iptr) << 24 | *(iptr+1) << 16 | *(iptr+2) << 8 | *(iptr+3);
iptr += 4;
}
} else {
while (count--) {
*optr++ = *(iptr) | *(iptr+1) << 8 | *(iptr+2) << 16 | *(iptr+3) << 24;
iptr += 4;
}
}
}
} else if (channels == 1) {
if (sample_size == 1) {
while (count--) {
*optr = *iptr++ << 24;
*(optr+1) = *optr;
optr += 2;
}
} else if (sample_size == 2) {
if (bigendian) {
while (count--) {
*optr = *(iptr) << 24 | *(iptr+1) << 16;
*(optr+1) = *optr;
iptr += 2;
optr += 2;
}
} else {
while (count--) {
*optr = *(iptr) << 16 | *(iptr+1) << 24;
*(optr+1) = *optr;
iptr += 2;
optr += 2;
}
}
} else if (sample_size == 3) {
if (bigendian) {
while (count--) {
*optr = *(iptr) << 24 | *(iptr+1) << 16 | *(iptr+2) << 8;
*(optr+1) = *optr;
iptr += 3;
optr += 2;
}
} else {
while (count--) {
*optr = *(iptr) << 8 | *(iptr+1) << 16 | *(iptr+2) << 24;
*(optr+1) = *optr;
iptr += 3;
optr += 2;
}
}
} else if (sample_size == 4) {
if (bigendian) {
while (count--) {
*optr++ = *(iptr) << 24 | *(iptr+1) << 16 | *(iptr+2) << 8 | *(iptr+3);
*(optr+1) = *optr;
iptr += 4;
optr += 2;
}
} else {
while (count--) {
*optr++ = *(iptr) | *(iptr+1) << 8 | *(iptr+2) << 16 | *(iptr+3) << 24;
*(optr+1) = *optr;
iptr += 4;
optr += 2;
}
}
}
} else {
LOG_ERROR("unsupported channels");
}
LOG_SDEBUG("decoded %u frames", frames);
_buf_inc_readp(streambuf, frames * bytes_per_frame);
if (limit) {
audio_left -= frames * bytes_per_frame;
}
IF_DIRECT(
_buf_inc_writep(outputbuf, frames * BYTES_PER_FRAME);
);
IF_PROCESS(
process.in_frames = frames;
);
UNLOCK_O_direct;
UNLOCK_S;
return DECODE_RUNNING;
}
static void pcm_open(u8_t size, u8_t rate, u8_t chan, u8_t endianness) {
sample_size = size - '0' + 1;
sample_rate = sample_rates[rate - '0'];
channels = chan - '0';
bigendian = (endianness == '0');
limit = false;
LOG_INFO("pcm size: %u rate: %u chan: %u bigendian: %u", sample_size, sample_rate, channels, bigendian);
buf_adjust(streambuf, sample_size * channels);
}
static void pcm_close(void) {
buf_adjust(streambuf, 1);
}
struct codec *register_pcm(void) {
static struct codec ret = {
'p', // id
"aif,pcm", // types
4096, // min read
102400, // min space
pcm_open, // open
pcm_close, // close
pcm_decode, // decode
};
LOG_INFO("using pcm to decode aif,pcm");
return &ret;
}