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Merge pull request #344 from mikebrady/short-packets

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Short packets – handles packets sent by the HTC 10's official AirPlay client. Needs a lot of testing!
This commit is contained in:
Mike Brady
2016-07-20 18:51:06 +01:00
committed by GitHub
parent 513b0e45e3 d12c7ec090
commit 828cbc855b
2 changed files with 89 additions and 66 deletions
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configure.ac
+1 -1
View File
@@ -2,7 +2,7 @@
# Process this file with autoconf to produce a configure script.
AC_PREREQ([2.50])
AC_INIT([shairport-sync], [2.8.4.4], [mikebrady@eircom.net])
AC_INIT([shairport-sync], [2.8.4.5], [mikebrady@eircom.net])
AM_INIT_AUTOMAKE
AC_CONFIG_SRCDIR([shairport.c])
AC_CONFIG_HEADERS([config.h])
player.c
+88 -65
View File
@@ -74,9 +74,13 @@ static AES_KEY aes;
#endif
static int sampling_rate, frame_size;
#define FRAME_BYTES(frame_size) (4 * frame_size)
static int bytes_per_audio_frame = 4;
// The maximum frame size change there can be is +/- 1;
static int max_frame_size_change = 1;
// #define FRAME_BYTES(frame_size) (4 * frame_size)
// maximal resampling shift - conservative
#define OUTFRAME_BYTES(frame_size) (4 * (frame_size + 3))
//#define OUTFRAME_BYTES(frame_size) (4 * (frame_size + 3))
#ifdef HAVE_LIBPOLARSSL
static aes_context dctx;
@@ -112,6 +116,7 @@ typedef struct audio_buffer_entry { // decoded audio packets
uint32_t timestamp;
seq_t sequence_number;
signed short *data;
int length; // the length of the decoded data
} abuf_t;
static abuf_t audio_buffer[BUFFER_FRAMES];
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
@@ -222,12 +227,21 @@ static inline int seq32_order(uint32_t a, uint32_t b) {
return (C & 0x80000000) == 0;
}
static int alac_decode(short *dest, uint8_t *buf, int len) {
static int alac_decode(short *dest, int *destlen, uint8_t *buf, int len) {
// parameters: where the decoded stuff goes, its length in samples,
// the incoming packet, the length of the incoming packet in bytes
// destlen should contain the allowed max number of samples on entry
if (len>MAX_PACKET) {
warn("Incoming audio packet size is too large at %d; it should not exceed %d.",len,MAX_PACKET);
return -1;
}
unsigned char packet[MAX_PACKET];
unsigned char packetp[MAX_PACKET];
assert(len <= MAX_PACKET);
// assert(len <= MAX_PACKET);
int reply = 0; //everything okay
int outsize=FRAME_BYTES(frame_size); // the size it should be
int outsize=bytes_per_audio_frame*(*destlen); // the size the output should be, in bytes
int toutsize = outsize;
if (encrypted) {
unsigned char iv[16];
@@ -244,14 +258,14 @@ static int alac_decode(short *dest, uint8_t *buf, int len) {
} else {
alac_decode_frame(decoder_info, buf, dest, &outsize);
}
if (outsize!=FRAME_BYTES(frame_size)) {
if(outsize<FRAME_BYTES(frame_size)) {
debug(2,"Output from alac_decode is smaller than expected. Encrypted = %d.",encrypted);
} else {
debug(2,"Output from alac_decode larger than expected -- truncated, but buffer overflow possible! Encrypted = %d.",encrypted);
}
reply = -1; // output frame is the wrong size
}
if(outsize>toutsize) {
debug(2,"Output from alac_decode larger (%d bytes, not frames) than expected (%d bytes) -- truncated, but buffer overflow possible! Encrypted = %d.",outsize, toutsize, encrypted);
reply = -1; // output packet is the wrong size
}
*destlen = outsize / bytes_per_audio_frame;
if ((outsize % bytes_per_audio_frame)!=0)
debug(1,"Number of audio frames (%d) does not correspond exactly to the number of bytes (%d) and the audio frame size (%d).",*destlen,outsize,bytes_per_audio_frame);
return reply;
}
@@ -290,7 +304,7 @@ static void free_decoder(void) { alac_free(decoder_info); }
static void init_buffer(void) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++)
audio_buffer[i].data = malloc(OUTFRAME_BYTES(frame_size));
audio_buffer[i].data = malloc(bytes_per_audio_frame*(frame_size+max_frame_size_change));
ab_resync();
}
@@ -375,8 +389,10 @@ void player_put_packet(seq_t seqno, uint32_t timestamp, uint8_t *data, int len)
// pthread_mutex_unlock(&ab_mutex);
if (abuf) {
if (alac_decode(abuf->data, data, len)==0) {
int datalen = frame_size;
if (alac_decode(abuf->data, &datalen, data, len)==0) {
abuf->ready = 1;
abuf->length = datalen;
abuf->timestamp = timestamp;
abuf->sequence_number = seqno;
@@ -686,11 +702,11 @@ static abuf_t *buffer_get_frame(void) {
// debug(2,"Zero length silence buffer needed with gross_frame_gap of %lld and dac_delay of %lld.",gross_frame_gap,dac_delay);
// the fs (number of frames of silence to play) can be zero in the DAC doesn't start ouotputting frames for a while -- it could get loaded up but not start responding for many milliseconds.
if (fs!=0) {
silence = malloc(FRAME_BYTES(fs));
silence = malloc(bytes_per_audio_frame*fs);
if (silence==NULL)
debug(1,"Failed to allocate %d byte silence buffer.",fs);
else {
memset(silence, 0, FRAME_BYTES(fs));
memset(silence, 0, bytes_per_audio_frame*fs);
// debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d,
// with %d packets.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write));
config.output->play(silence, fs);
@@ -822,7 +838,7 @@ static abuf_t *buffer_get_frame(void) {
if (!curframe->ready) {
// debug(1, "Supplying a silent frame for frame %u", read);
missing_packets++;
memset(curframe->data, 0, FRAME_BYTES(frame_size));
memset(curframe->data, 0, bytes_per_audio_frame*frame_size);
curframe->timestamp = 0;
}
curframe->ready = 0;
@@ -842,17 +858,17 @@ static inline short shortmean(short a, short b) {
}
// stuff: 1 means add 1; 0 means do nothing; -1 means remove 1
static int stuff_buffer_basic(short *inptr, short *outptr, int stuff) {
if ((stuff > 1) || (stuff < -1)) {
debug(1, "Stuff argument to stuff_buffer must be from -1 to +1.");
return frame_size;
static int stuff_buffer_basic(short *inptr, int length, short *outptr, int stuff) {
if ((stuff > 1) || (stuff < -1) || (length <100)) {
// debug(1, "Stuff argument to stuff_buffer must be from -1 to +1 and length >100.");
return length;
}
int i;
int stuffsamp = frame_size;
int stuffsamp = length;
if (stuff)
// stuffsamp = rand() % (frame_size - 1);
// stuffsamp = rand() % (length - 1);
stuffsamp =
(rand() % (frame_size - 2)) + 1; // ensure there's always a sample before and after the item
(rand() % (length - 2)) + 1; // ensure there's always a sample before and after the item
pthread_mutex_lock(&vol_mutex);
for (i = 0; i < stuffsamp; i++) { // the whole frame, if no stuffing
@@ -872,22 +888,28 @@ static int stuff_buffer_basic(short *inptr, short *outptr, int stuff) {
inptr++;
inptr++;
}
for (i = stuffsamp; i < frame_size + stuff; i++) {
// if you're removing, i.e. stuff < 0, copy that much less over. If you're adding, do all the rest.
int remainder = length;
if (stuff<0)
remainder = remainder+stuff; // don't run over the correct end of the output buffer
for (i = stuffsamp; i < remainder; i++) {
*outptr++ = dithered_vol(*inptr++);
*outptr++ = dithered_vol(*inptr++);
}
}
pthread_mutex_unlock(&vol_mutex);
return frame_size + stuff;
return length + stuff;
}
#ifdef HAVE_LIBSOXR
// stuff: 1 means add 1; 0 means do nothing; -1 means remove 1
static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) {
if ((stuff > 1) || (stuff < -1)) {
debug(1, "Stuff argument to sox_stuff_buffer must be from -1 to +1.");
return frame_size;
static int stuff_buffer_soxr(short *inptr, int length, short *outptr, int stuff) {
if ((stuff > 1) || (stuff < -1) || (length < 100)) {
// debug(1, "Stuff argument to sox_stuff_buffer must be from -1 to +1 and length must be > 100.");
return length;
}
int i;
short *ip, *op;
@@ -905,16 +927,16 @@ static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) {
size_t odone;
soxr_error_t error = soxr_oneshot(frame_size, frame_size + stuff, 2, /* Rates and # of chans. */
inptr, frame_size, NULL, /* Input. */
outptr, frame_size + stuff, &odone, /* Output. */
soxr_error_t error = soxr_oneshot(length, length + stuff, 2, /* Rates and # of chans. */
inptr, length, NULL, /* Input. */
outptr, length + stuff, &odone, /* Output. */
&io_spec, /* Input, output and transfer spec. */
NULL, NULL); /* Default configuration.*/
if (error)
die("soxr error: %s\n", "error: %s\n", soxr_strerror(error));
if (odone > frame_size + 1)
if (odone > length + 1)
die("odone = %d!\n", odone);
const int gpm = 5;
@@ -926,8 +948,8 @@ static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) {
}
// keep the last (dpm) samples, to mitigate the Gibbs phenomenon
op = outptr + (frame_size + stuff - gpm) * sizeof(short);
ip = inptr + (frame_size - gpm) * sizeof(short);
op = outptr + (length + stuff - gpm) * sizeof(short);
ip = inptr + (length - gpm) * sizeof(short);
for (i = 0; i < gpm; i++) {
*op++ = *ip++;
*op++ = *ip++;
@@ -937,7 +959,7 @@ static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) {
if (fix_volume != 65536.0) {
// pthread_mutex_lock(&vol_mutex);
op = outptr;
for (i = 0; i < frame_size + stuff; i++) {
for (i = 0; i < length + stuff; i++) {
*op = dithered_vol(*op);
op++;
*op = dithered_vol(*op);
@@ -949,13 +971,13 @@ static int stuff_buffer_soxr(short *inptr, short *outptr, int stuff) {
} else { // the whole frame, if no stuffing
// pthread_mutex_lock(&vol_mutex);
for (i = 0; i < frame_size; i++) {
for (i = 0; i < length; i++) {
*op++ = dithered_vol(*ip++);
*op++ = dithered_vol(*ip++);
};
// pthread_mutex_unlock(&vol_mutex);
}
return frame_size + stuff;
return length + stuff;
}
#endif
@@ -1012,13 +1034,13 @@ static void *player_thread_func(void *arg) {
initstate(time(NULL), rnstate, 256);
signed short *inbuf, *outbuf, *silence;
outbuf = malloc(OUTFRAME_BYTES(frame_size));
outbuf = malloc(bytes_per_audio_frame*(frame_size+max_frame_size_change));
if (outbuf==NULL)
debug(1,"Failed to allocate memory for an output buffer.");
silence = malloc(OUTFRAME_BYTES(frame_size));
silence = malloc(bytes_per_audio_frame*frame_size);
if (silence==NULL)
debug(1,"Failed to allocate memory for a silence buffer.");
memset(silence, 0, OUTFRAME_BYTES(frame_size));
memset(silence, 0, bytes_per_audio_frame*frame_size);
late_packet_message_sent = 0;
first_packet_timestamp = 0;
missing_packets = late_packets = too_late_packets = resend_requests = 0;
@@ -1066,21 +1088,20 @@ static void *player_thread_func(void *arg) {
while (!please_stop) {
abuf_t *inframe = buffer_get_frame();
if (inframe) {
inbuf = inframe->data;
if (inbuf) {
if (inframe->data) {
play_number++;
// if it's a supplied silent frame, let us know...
if (inframe->timestamp == 0) {
// debug(1,"Player has a supplied silent frame.");
last_seqno_read =
(SUCCESSOR(last_seqno_read) & 0xffff); // manage the packet out of sequence minder
if (inbuf==NULL)
debug(1,"NULL inbuf to play -- skipping it.");
if (inframe->data==NULL)
debug(1,"NULL inframe->data to play -- skipping it.");
else {
if (frame_size==0)
debug(1,"empty frame to play -- skipping it (1).");
if (inframe->length==0)
debug(1,"empty frame to play -- skipping it.");
else
config.output->play(inbuf, frame_size);
config.output->play(inframe->data, inframe->length);
}
} else {
// We have a frame of data. We need to see if we want to add or remove a frame from it to
@@ -1146,7 +1167,9 @@ static void *player_thread_func(void *arg) {
if (buffer_occupancy > maximum_buffer_occupancy)
maximum_buffer_occupancy = buffer_occupancy;
// here, we want to check (a) if we are meant to do synchronisation, (b) if we have a delay procedure, (b) if we can get the delay.
// here, we want to check (a) if we are meant to do synchronisation,
// (b) if we have a delay procedure, (c) if we can get the delay.
// If any of these are false, we don't do any synchronisation stuff
int resp = -1; // use this as a flag -- if negative, we can't rely on a real known delay
@@ -1223,29 +1246,29 @@ static void *player_thread_func(void *arg) {
// if no stuffing needed and no volume adjustment, then
// don't send to stuff_buffer_* and don't copy to outbuf; just send directly to the
// output device...
if (inbuf==NULL)
debug(1,"NULL inbuf to play -- skipping it.");
if (inframe->data==NULL)
debug(1,"NULL inframe->data to play -- skipping it.");
else {
if (frame_size==0)
if (inframe->length==0)
debug(1,"empty frame to play -- skipping it (2).");
else
config.output->play(inbuf, frame_size);
config.output->play(inframe->data, inframe->length);
}
} else {
#ifdef HAVE_LIBSOXR
switch (config.packet_stuffing) {
case ST_basic:
// if (amount_to_stuff) debug(1,"Basic stuff...");
play_samples = stuff_buffer_basic(inbuf, outbuf, amount_to_stuff);
play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, amount_to_stuff);
break;
case ST_soxr:
// if (amount_to_stuff) debug(1,"Soxr stuff...");
play_samples = stuff_buffer_soxr(inbuf, outbuf, amount_to_stuff);
play_samples = stuff_buffer_soxr(inframe->data, inframe->length, outbuf, amount_to_stuff);
break;
}
#else
// if (amount_to_stuff) debug(1,"Standard stuff...");
play_samples = stuff_buffer_basic(inbuf, outbuf, amount_to_stuff);
play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, amount_to_stuff);
#endif
/*
@@ -1303,24 +1326,24 @@ static void *player_thread_func(void *arg) {
if (fix_volume == 0x10000)
if (inbuf==NULL)
debug(1,"NULL inbuf to play -- skipping it.");
if (inframe->data==NULL)
debug(1,"NULL inframe->data to play -- skipping it.");
else {
if (frame_size==0)
if (inframe->length==0)
debug(1,"empty frame to play -- skipping it (3).");
else
config.output->play(inbuf, frame_size);
config.output->play(inframe->data, inframe->length);
}
else {
play_samples = stuff_buffer_basic(inbuf, outbuf, 0); // no stuffing, but volume adjustment
play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, 0); // no stuffing, but volume adjustment
if (outbuf==NULL)
debug(1,"NULL outbuf to play -- skipping it.");
else {
if (frame_size==0)
if (inframe->length==0)
debug(1,"empty frame to play -- skipping it (4).");
else
config.output->play(outbuf, frame_size);
config.output->play(outbuf, play_samples);
}
}
}