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player.c
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player.c
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/*
* Slave-clocked ALAC stream player. This file is part of Shairport.
* Copyright (c) James Laird 2011, 2013
* All rights reserved.
*
* Modifications for audio synchronisation
* and related work, copyright (c) Mike Brady 2014
* All rights reserved.
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or
* sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <sys/types.h>
#include <pthread.h>
#include <math.h>
#include <sys/stat.h>
#include <signal.h>
#include <sys/syslog.h>
#include <assert.h>
#include <fcntl.h>
#include <stdlib.h>
#include <errno.h>
#include <limits.h>
#include "config.h"
#ifdef HAVE_LIBPOLARSSL
#include <polarssl/aes.h>
#include <polarssl/havege.h>
#endif
#ifdef HAVE_LIBSSL
#include <openssl/aes.h>
#endif
#ifdef HAVE_LIBSOXR
#include <soxr.h>
#endif
#include "common.h"
#include "player.h"
#include "rtp.h"
#include "rtsp.h"
#include "alac.h"
// parameters from the source
static unsigned char *aesiv;
#ifdef HAVE_LIBSSL
static AES_KEY aes;
#endif
static int sampling_rate, frame_size;
#define FRAME_BYTES(frame_size) (4*frame_size)
// maximal resampling shift - conservative
#define OUTFRAME_BYTES(frame_size) (4*(frame_size+3))
#ifdef HAVE_LIBPOLARSSL
static aes_context dctx;
#endif
static pthread_t player_thread;
static int please_stop;
static int connection_state_to_output; // if true, then play incoming stuff; if false drop everything
static alac_file *decoder_info;
// debug variables
static int late_packet_message_sent;
static uint64_t packet_count = 0;
static int32_t last_seqno_read;
// interthread variables
static double software_mixer_volume = 1.0;
static int fix_volume = 0x10000;
static pthread_mutex_t vol_mutex = PTHREAD_MUTEX_INITIALIZER;
// default buffer size
// needs to be a power of 2 because of the way BUFIDX(seqno) works
#define BUFFER_FRAMES 512
#define MAX_PACKET 2048
// DAC buffer occupancy stuff
#define DAC_BUFFER_QUEUE_DESIRED_LENGTH 6615
#define DAC_BUFFER_QUEUE_MINIMUM_LENGTH 5000
typedef struct audio_buffer_entry { // decoded audio packets
int ready;
uint32_t timestamp;
seq_t sequence_number;
signed short *data;
} abuf_t;
static abuf_t audio_buffer[BUFFER_FRAMES];
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
// mutex-protected variables
static seq_t ab_read, ab_write;
static int ab_buffering = 1, ab_synced = 0;
static uint32_t first_packet_timestamp = 0;
static int flush_requested = 0;
static uint32_t flush_rtp_timestamp;
static uint64_t time_of_last_audio_packet;
static int shutdown_requested;
// mutexes and condition variables
static pthread_mutex_t ab_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t flush_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_cond_t flowcontrol;
static int64_t first_packet_time_to_play; // nanoseconds
static audio_parameters audio_information;
// stats
static uint64_t missing_packets,late_packets,too_late_packets,resend_requests;
static void ab_resync(void) {
int i;
for (i=0; i<BUFFER_FRAMES; i++) {
audio_buffer[i].ready = 0;
audio_buffer[i].sequence_number = 0;
}
ab_synced = 0;
last_seqno_read = -1;
ab_buffering = 1;
}
// the sequence number is a 16-bit unsigned number which wraps pretty often
// to work out if one seqno is 'after' another therefore depends whether wrap has occurred
// this function works out the actual ordinate of the seqno, i.e. the distance up from
// the zeroth element, at ab_read, taking due account of wrap.
static inline seq_t SUCCESSOR(seq_t x) {
uint32_t p = x & 0xffff;
p+=1;
p = p & 0xffff;
return p;
}
static inline seq_t PREDECESSOR(seq_t x) {
uint32_t p = (x & 0xffff)+0x10000;
p-=1;
p = p & 0xffff;
return p;
}
// anything with ORDINATE in it must be proctected by the ab_mutex
static inline int32_t ORDINATE(seq_t x) {
int32_t p = x & 0xffff;
int32_t q = ab_read & 0x0ffff;
int32_t t = (p+0x10000-q) & 0xffff;
// we definitely will get a positive number in t at this point, but it might be a
// positive alias of a negative number, i.e. x might actually be "before" ab_read
// So, if the result is greater than 32767, we will assume its an
// alias and subtract 65536 from it
if (t >= 32767) {
// debug(1,"OOB: %u, ab_r: %u, ab_w: %u",x,ab_read,ab_write);
t -= 65536;
}
return t;
}
// wrapped number between two seq_t.
int32_t seq_diff(seq_t a, seq_t b) {
int32_t diff = ORDINATE(b) - ORDINATE(a);
return diff;
}
// the sequence numbers will wrap pretty often.
// this returns true if the second arg is after the first
static inline int seq_order(seq_t a, seq_t b) {
int32_t d = ORDINATE(b) - ORDINATE(a);
return d > 0;
}
static inline seq_t seq_sum(seq_t a, seq_t b) {
uint32_t p = a & 0xffff;
uint32_t q = b & 0x0ffff;
uint32_t r = (a+b) & 0xffff;
return r;
}
// now for 32-bit wrapping in timestamps
// this returns true if the second arg is strictly after the first
// on the assumption that the gap between them is never greater than (2^31)-1
// Represent a and b in 64 bits
static inline int seq32_order(uint32_t a, uint32_t b) {
if (a==b)
return 0;
int64_t A = a & 0xffffffff;
int64_t B = b & 0xffffffff;
int64_t C = B-A;
// if bit 31 is set, it means either b is before (i.e. less than) a or
// b is (2^31)-1 ahead of a.
// If we assume the gap between b and a should never reach 2 billion, then
// bit 31 == 0 means b is strictly after a
return (C & 0x80000000) == 0;
}
static void alac_decode(short *dest, uint8_t *buf, int len) {
unsigned char packet[MAX_PACKET];
unsigned char packetp[MAX_PACKET];
assert(len<=MAX_PACKET);
unsigned char iv[16];
int aeslen = len & ~0xf;
memcpy(iv, aesiv, sizeof(iv));
#ifdef HAVE_LIBPOLARSSL
aes_crypt_cbc(&dctx,AES_DECRYPT,aeslen,iv,buf, packet);
#endif
#ifdef HAVE_LIBSSL
AES_cbc_encrypt(buf, packet, aeslen, &aes, iv, AES_DECRYPT);
#endif
memcpy(packet+aeslen, buf+aeslen, len-aeslen);
int outsize;
alac_decode_frame(decoder_info, packet, dest, &outsize);
assert(outsize == FRAME_BYTES(frame_size));
}
static int init_decoder(int32_t fmtp[12]) {
alac_file *alac;
frame_size = fmtp[1]; // stereo samples
sampling_rate = fmtp[11];
int sample_size = fmtp[3];
if (sample_size != 16)
die("only 16-bit samples supported!");
alac = alac_create(sample_size, 2);
if (!alac)
return 1;
decoder_info = alac;
alac->setinfo_max_samples_per_frame = frame_size;
alac->setinfo_7a = fmtp[2];
alac->setinfo_sample_size = sample_size;
alac->setinfo_rice_historymult = fmtp[4];
alac->setinfo_rice_initialhistory = fmtp[5];
alac->setinfo_rice_kmodifier = fmtp[6];
alac->setinfo_7f = fmtp[7];
alac->setinfo_80 = fmtp[8];
alac->setinfo_82 = fmtp[9];
alac->setinfo_86 = fmtp[10];
alac->setinfo_8a_rate = fmtp[11];
alac_allocate_buffers(alac);
return 0;
}
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));
ab_resync();
}
static void free_buffer(void) {
int i;
for (i=0; i<BUFFER_FRAMES; i++)
free(audio_buffer[i].data);
}
void player_put_packet(seq_t seqno,uint32_t timestamp, uint8_t *data, int len) {
packet_count++;
pthread_mutex_lock(&ab_mutex);
time_of_last_audio_packet = get_absolute_time_in_fp();
if (connection_state_to_output) { // if we are supposed to be processing these packets
if ((flush_rtp_timestamp!=0) && ((timestamp==flush_rtp_timestamp) || seq32_order(timestamp,flush_rtp_timestamp))) {
debug(2,"Dropping flushed packet in player_put_packet, seqno %u, timestamp %u, flushing to timestamp: %u.",seqno,timestamp,flush_rtp_timestamp);
} else {
if ((flush_rtp_timestamp!=0x0) && (!seq32_order(timestamp,flush_rtp_timestamp))) // if we have gone past the flush boundary time
flush_rtp_timestamp=0x0;
abuf_t *abuf = 0;
if (!ab_synced) {
debug(2, "syncing to seqno %u.", seqno);
ab_write = seqno;
ab_read = seqno;
ab_synced = 1;
}
if (ab_write == seqno) { // expected packet
abuf = audio_buffer + BUFIDX(seqno);
ab_write = SUCCESSOR(seqno);
} else if (seq_order(ab_write, seqno)) { // newer than expected
//if (ORDINATE(seqno)>(BUFFER_FRAMES*7)/8)
// debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and seqno: %u.",seq_diff(ab_read,seqno),ab_read,ab_write,seqno);
int32_t gap = seq_diff(ab_write,PREDECESSOR(seqno))+1;
if (gap<=0)
debug(1,"Unexpected gap size: %d.",gap);
int i;
for (i=0;i<gap;i++) {
abuf = audio_buffer + BUFIDX(seq_sum(ab_write,i));
abuf->ready = 0; // to be sure, to be sure
abuf->timestamp = 0;
abuf->sequence_number = 0;
}
// debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write);
abuf = audio_buffer + BUFIDX(seqno);
rtp_request_resend(ab_write,gap);
resend_requests++;
ab_write = SUCCESSOR(seqno);
} else if (seq_order(ab_read, seqno)) { // late but not yet played
late_packets++;
abuf = audio_buffer + BUFIDX(seqno);
} else { // too late.
too_late_packets++;
/*
if (!late_packet_message_sent) {
debug(1, "too-late packet received: %u; ab_read: %u; ab_write: %u.", seqno, ab_read, ab_write);
late_packet_message_sent=1;
}
*/
}
// pthread_mutex_unlock(&ab_mutex);
if (abuf) {
alac_decode(abuf->data, data, len);
abuf->ready = 1;
abuf->timestamp = timestamp;
abuf->sequence_number = seqno;
}
// pthread_mutex_lock(&ab_mutex);
}
int rc = pthread_cond_signal(&flowcontrol);
if (rc)
debug(1,"Error signalling flowcontrol.");
}
pthread_mutex_unlock(&ab_mutex);
}
static inline short lcg_rand(void) {
static unsigned long lcg_prev = 12345;
lcg_prev = lcg_prev * 69069 + 3;
return lcg_prev & 0xffff;
}
static inline short dithered_vol(short sample) {
short rand_a, rand_b;
long out;
out = (long)sample * fix_volume;
if (fix_volume < 0x10000) {
rand_b = rand_a;
rand_a = lcg_rand();
out += rand_a;
out -= rand_b;
}
return out>>16;
}
// get the next frame, when available. return 0 if underrun/stream reset.
static abuf_t *buffer_get_frame(void) {
int16_t buf_fill;
uint64_t local_time_now;
// struct timespec tn;
abuf_t *abuf = 0;
int i;
abuf_t *curframe;
pthread_mutex_lock(&ab_mutex);
int wait;
int32_t dac_delay = 0;
do {
// get the time
local_time_now = get_absolute_time_in_fp();
// if config.timeout (default 120) seconds have elapsed since the last audio packet was received, then we should stop.
// config.timeout of zero means don't check..., but iTunes may be confused by a long gap followed by a resumption...
if ((time_of_last_audio_packet!=0) && (shutdown_requested==0) && (config.timeout!=0)) {
uint64_t ct = config.timeout; // go from int to 64-bit int
if (local_time_now-time_of_last_audio_packet>=ct<<32) {
debug(1,"As Yeats almost said, \"Too long a silence / can make a stone of the heart\"");
rtsp_request_shutdown_stream();
shutdown_requested=1;
}
}
int rco = get_requested_connection_state_to_output();
if (connection_state_to_output != rco) {
connection_state_to_output=rco;
// change happening
if (connection_state_to_output==0) { //going off
pthread_mutex_lock(&flush_mutex);
flush_requested=1;
pthread_mutex_unlock(&flush_mutex);
}
}
pthread_mutex_lock(&flush_mutex);
if (flush_requested==1) {
if (config.output->flush)
config.output->flush();
ab_resync();
first_packet_timestamp = 0;
first_packet_time_to_play = 0;
flush_requested=0;
}
pthread_mutex_unlock(&flush_mutex);
uint32_t flush_limit = 0;
if (ab_synced) {
do {
curframe = audio_buffer + BUFIDX(ab_read);
if (curframe->ready) {
if (curframe->sequence_number!=ab_read) {
// some kind of sync problem has occurred.
if (BUFIDX(curframe->sequence_number)==BUFIDX(ab_read)) {
// it looks like some kind of aliasing has happened
if (seq_order(ab_read,curframe->sequence_number)) {
ab_read=curframe->sequence_number;
debug(1,"Aliasing of buffer index -- reset.");
}
} else {
debug(1,"Inconsistent sequence numbers detected");
}
}
if ((flush_rtp_timestamp!=0) && ((curframe->timestamp==flush_rtp_timestamp) || seq32_order(curframe->timestamp,flush_rtp_timestamp))) {
debug(1,"Dropping flushed packet seqno %u, timestamp %u",curframe->sequence_number,curframe->timestamp);
curframe->ready=0;
flush_limit++;
ab_read=SUCCESSOR(ab_read);
}
if ((flush_rtp_timestamp!=0) && (!seq32_order(curframe->timestamp,flush_rtp_timestamp))) // if we have gone past the flush boundary time
flush_rtp_timestamp=0;
}
} while ((flush_rtp_timestamp!=0) && (flush_limit<=8820) && (curframe->ready==0));
if (flush_limit==8820) {
debug(1,"Flush hit the 8820 frame limit!");
flush_limit=0;
}
dac_delay = 0;
curframe = audio_buffer + BUFIDX(ab_read);
if (config.output->delay) {
dac_delay = config.output->delay();
if (dac_delay==-1) {
debug(1,"Error getting dac_delay at start of loop.");
dac_delay=0;
}
}
if (curframe->ready) {
/*
// This is broken -- it causes infinite loops. It's in the wrong place, and anyway isn't used much.
if ((flush_rtp_timestamp) && (flush_rtp_timestamp>=curframe->timestamp)) {
debug(1,"Dropping flushed packet seqno %u, timestamp %u",curframe->sequence_number,curframe->timestamp);
curframe->ready=0;
if (flush_rtp_timestamp==curframe->timestamp) // if we are finished...
flush_rtp_timestamp=0;
ab_read==SUCCESSOR(ab_read);
} else
*/
if (ab_buffering) { // if we are getting packets but not yet forwarding them to the player
if (first_packet_timestamp==0) { // if this is the very first packet
// debug(1,"First frame seen, time %u, with %d frames...",curframe->timestamp,seq_diff(ab_read, ab_write));
uint32_t reference_timestamp;
uint64_t reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp,&reference_timestamp_time);
if (reference_timestamp) { // if we have a reference time
// debug(1,"First frame seen with timestamp...");
first_packet_timestamp=curframe->timestamp; // we will keep buffering until we are supposed to start playing this
// here, see if we should start playing. We need to know when to allow the packets to be sent to the player
// we will get a fix every second or so, which will be stored as a pair consisting of
// the time when the packet with a particular timestamp should be played.
// it might not be the timestamp of our first packet, however, so we might have to do some calculations.
int64_t delta = ((int64_t)first_packet_timestamp-(int64_t)reference_timestamp);
first_packet_time_to_play = reference_timestamp_time+((delta+(int64_t)config.latency)<<32)/44100; // using the latency requested...
if (local_time_now>=first_packet_time_to_play) {
debug(2,"First packet is late! It should have played before now. Flushing 0.1 seconds");
player_flush(first_packet_timestamp+4410);
}
}
}
if (first_packet_time_to_play!=0) {
uint32_t filler_size = frame_size;
uint32_t max_dac_delay = DAC_BUFFER_QUEUE_DESIRED_LENGTH;
// if (dac_delay==0) // i.e. if this is the first fill
filler_size = 4410; // 0.1 second -- the maximum we'll add to the DAC
if (local_time_now>=first_packet_time_to_play) {
// we've gone past the time...
// debug(1,"Run past the exact start time by %llu frames, with time now of %llx, fpttp of %llx and dac_delay of %d and %d packets; flush.",(((tn-first_packet_time_to_play)*44100)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read, ab_write));
if (config.output->flush)
config.output->flush();
ab_resync();
first_packet_timestamp = 0;
first_packet_time_to_play = 0;
} else {
uint64_t gross_frame_gap = ((first_packet_time_to_play-local_time_now)*44100)>>32;
int64_t exact_frame_gap = gross_frame_gap-dac_delay;
if (exact_frame_gap<=0) {
// we've gone past the time...
// debug(1,"Run a bit past the exact start time by %lld frames, with time now of %llx, fpttp of %llx and dac_delay of %d and %d packets; flush.",-exact_frame_gap,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read, ab_write));
if (config.output->flush)
config.output->flush();
ab_resync();
first_packet_timestamp = 0;
first_packet_time_to_play = 0;
} else {
uint32_t fs=filler_size;
if (fs>(max_dac_delay-dac_delay))
fs=max_dac_delay-dac_delay;
if ((exact_frame_gap<=fs) || (exact_frame_gap<=frame_size*2)) {
fs=exact_frame_gap;
// debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d, with %d packets, ab_read is %04x, ab_write is %04x.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write),ab_read,ab_write);
ab_buffering = 0;
}
signed short *silence;
silence = malloc(FRAME_BYTES(fs));
memset(silence, 0, FRAME_BYTES(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);
free(silence);
}
}
}
}
}
}
wait = (ab_buffering || (dac_delay>=DAC_BUFFER_QUEUE_DESIRED_LENGTH) || (!ab_synced)) && (!please_stop);
// wait = (ab_buffering || (seq_diff(ab_read, ab_write) < (config.latency-22000)/(352)) || (!ab_synced)) && (!please_stop);
if (wait) {
uint64_t time_to_wait_for_wakeup_fp = ((uint64_t)1<<32)/44100; // this is time period of one frame
time_to_wait_for_wakeup_fp *= 4*352; // four full 352-frame packets
time_to_wait_for_wakeup_fp /= 3; //four thirds of a packet time
#ifdef COMPILE_FOR_LINUX
uint64_t time_of_wakeup_fp = local_time_now+time_to_wait_for_wakeup_fp;
uint64_t sec = time_of_wakeup_fp>>32;
uint64_t nsec = ((time_of_wakeup_fp&0xffffffff)*1000000000)>>32;
struct timespec time_of_wakeup;
time_of_wakeup.tv_sec = sec;
time_of_wakeup.tv_nsec = nsec;
pthread_cond_timedwait(&flowcontrol,&ab_mutex,&time_of_wakeup);
// int rc = pthread_cond_timedwait(&flowcontrol,&ab_mutex,&time_of_wakeup);
// if (rc!=0)
// debug(1,"pthread_cond_timedwait returned error code %d.",rc);
#endif
#ifdef COMPILE_FOR_OSX
uint64_t sec = time_to_wait_for_wakeup_fp>>32;;
uint64_t nsec = ((time_to_wait_for_wakeup_fp&0xffffffff)*1000000000)>>32;
struct timespec time_to_wait;
time_to_wait.tv_sec = sec;
time_to_wait.tv_nsec = nsec;
pthread_cond_timedwait_relative_np(&flowcontrol,&ab_mutex,&time_to_wait);
#endif
}
} while (wait);
if (please_stop) {
pthread_mutex_unlock(&ab_mutex);
return 0;
}
seq_t read = ab_read;
// check if t+8, t+16, t+32, t+64, t+128, ... (buffer_start_fill / 2)
// packets have arrived... last-chance resend
if (!ab_buffering) {
for (i = 8; i < (seq_diff(ab_read,ab_write) / 2); i = (i * 2)) {
seq_t next = seq_sum(ab_read,i);
abuf = audio_buffer + BUFIDX(next);
if (!abuf->ready) {
rtp_request_resend(next, 1);
// debug(1,"Resend %u.",next);
resend_requests++;
}
}
}
if (!curframe->ready) {
// debug(1, " %d. Supplying a silent frame.", read);
missing_packets++;
memset(curframe->data, 0, FRAME_BYTES(frame_size));
curframe->timestamp=0;
}
curframe->ready = 0;
ab_read=SUCCESSOR(ab_read);
pthread_mutex_unlock(&ab_mutex);
return curframe;
}
static inline short shortmean(short a, short b) {
long al = (long)a;
long bl = (long)b;
long longmean = (al+bl)/2;
short r = (short)longmean;
if (r!=longmean)
debug(1,"Error calculating average of two shorts");
return r;
}
// 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;
}
int i;
int stuffsamp = frame_size;
if (stuff)
// stuffsamp = rand() % (frame_size - 1);
stuffsamp = (rand() % (frame_size-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
*outptr++ = dithered_vol(*inptr++);
*outptr++ = dithered_vol(*inptr++);
};
if (stuff) {
if (stuff==1) {
debug(3, "+++++++++");
// interpolate one sample
//*outptr++ = dithered_vol(((long)inptr[-2] + (long)inptr[0]) >> 1);
//*outptr++ = dithered_vol(((long)inptr[-1] + (long)inptr[1]) >> 1);
*outptr++ = dithered_vol(shortmean(inptr[-2],inptr[0]));
*outptr++ = dithered_vol(shortmean(inptr[-1],inptr[1]));
} else if (stuff==-1) {
debug(3, "---------");
inptr++;
inptr++;
}
for (i=stuffsamp; i<frame_size + stuff; i++) {
*outptr++ = dithered_vol(*inptr++);
*outptr++ = dithered_vol(*inptr++);
}
}
pthread_mutex_unlock(&vol_mutex);
return frame_size + 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;
}
int i;
short *ip,*op;
ip=inptr;
op=outptr;
if (stuff) {
// debug(1,"Stuff %d.",stuff);
soxr_io_spec_t io_spec;
io_spec.itype = SOXR_INT16_I;
io_spec.otype = SOXR_INT16_I;
io_spec.scale = 1.0; // this seems to crash if not = 1.0
io_spec.e = NULL;
io_spec.flags = 0;
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. */
&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)
die("odone = %d!\n", odone);
const int gpm = 5;
// keep the first (dpm) samples, to mitigate the Gibbs phenomenon
for (i=0;i<gpm;i++) {
*op++ = *ip++;
*op++ = *ip++;
}
// 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);
for (i=0;i<gpm;i++) {
*op++ = *ip++;
*op++ = *ip++;
}
// finally, adjust the volume, if necessary
if (software_mixer_volume!=1.0) {
// pthread_mutex_lock(&vol_mutex);
op=outptr;
for (i=0; i<frame_size+stuff; i++) {
*op = dithered_vol(*op);
op++;
*op = dithered_vol(*op);
op++;
};
// pthread_mutex_unlock(&vol_mutex);
}
} else { // the whole frame, if no stuffing
// pthread_mutex_lock(&vol_mutex);
for (i=0; i<frame_size; i++) {
*op++ = dithered_vol(*ip++);
*op++ = dithered_vol(*ip++);
};
// pthread_mutex_unlock(&vol_mutex);
}
return frame_size + stuff;
}
#endif
typedef struct stats { // statistics for running averages
int64_t sync_error,correction,drift;
} stats_t;
static void *player_thread_func(void *arg) {
connection_state_to_output = get_requested_connection_state_to_output();
//this is about half a minute
#define trend_interval 3758
stats_t statistics[trend_interval];
int number_of_statistics,oldest_statistic,newest_statistic;
int64_t tsum_of_sync_errors,tsum_of_corrections,tsum_of_insertions_and_deletions,tsum_of_drifts;
int64_t previous_sync_error,previous_correction;
int64_t minimum_dac_queue_size = 1000000;
int32_t minimum_buffer_occupancy = BUFFER_FRAMES;
int32_t maximum_buffer_occupancy = 0;
audio_information.valid=0;
int play_samples;
int64_t current_delay;
int play_number = 0;
time_of_last_audio_packet = 0;
shutdown_requested = 0;
number_of_statistics = oldest_statistic = newest_statistic = 0;
tsum_of_sync_errors = tsum_of_corrections = tsum_of_insertions_and_deletions = tsum_of_drifts = 0;
const int print_interval = trend_interval; // don't ask...
// I think it's useful to keep this prime to prevent it from falling into a pattern with some other process.
char rnstate[256];
initstate(time(NULL),rnstate,256);
signed short *inbuf, *outbuf, *silence;
outbuf = malloc(OUTFRAME_BYTES(frame_size));
silence = malloc(OUTFRAME_BYTES(frame_size));
memset(silence, 0, OUTFRAME_BYTES(frame_size));
late_packet_message_sent=0;
missing_packets=late_packets=too_late_packets=resend_requests=0;
flush_rtp_timestamp=0; // it seems this number has a special significance -- it seems to be used as a null operand, so we'll use it like that too
int sync_error_out_of_bounds = 0; // number of times in a row that there's been a serious sync error
while (!please_stop) {
abuf_t *inframe = buffer_get_frame();
if (inframe) {
inbuf = inframe->data;
if (inbuf) {
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
config.output->play(inbuf, frame_size);
} else {
// We have a frame of data. We need to see if we want to add or remove a frame from it to keep in sync.
// So we calculate the timing error for the first frame in the DAC.
// If it's ahead of time, we add one audio frame to this frame to delay a subsequent frame
// If it's late, we remove an audio frame from this frame to bring a subsequent frame forward in time
uint32_t reference_timestamp;
uint64_t reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp,&reference_timestamp_time);
int64_t rt,nt;
rt = reference_timestamp;
nt = inframe->timestamp;
uint64_t local_time_now = get_absolute_time_in_fp();
//struct timespec tn;
//clock_gettime(CLOCK_MONOTONIC,&tn);
//uint64_t local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000;
int64_t td_in_frames;
int64_t td = local_time_now-reference_timestamp_time;
// debug(1,"td is %lld.",td);
if (td>=0) {
td_in_frames = (td*44100)>>32;
} else {
td_in_frames = -((-td*44100)>>32);
}
if (config.output->delay) {
current_delay = config.output->delay();
if (current_delay==-1) {
debug(1,"Delay error when checking running latency.");
current_delay=0;
}
} else {
current_delay = 0;
}
if (current_delay<minimum_dac_queue_size)
minimum_dac_queue_size=current_delay;
uint32_t bo = seq_diff(ab_read,ab_write);
if (bo<minimum_buffer_occupancy)
minimum_buffer_occupancy=bo;
if (bo>maximum_buffer_occupancy)
maximum_buffer_occupancy=bo;
// this is the actual delay, including the latency we actually want, which will fluctuate a good bit about a potentially rising or falling trend.
int64_t delay = td_in_frames+rt-(nt-current_delay);
// This is the timing error for the next audio frame in the DAC.
int64_t sync_error = delay-config.latency;
// before we finally commit to this frame, check its sequencing and timing
// check sequencing
if (last_seqno_read==-1)
last_seqno_read=inframe->sequence_number;
else {
last_seqno_read = (SUCCESSOR(last_seqno_read) & 0xffff);
if (inframe->sequence_number!=last_seqno_read)
debug(1,"Player: packets out of sequence: expected: %d, got: %d, sync error: %d frames.",last_seqno_read,inframe->sequence_number,sync_error);
last_seqno_read=inframe->sequence_number; // reset warning...
}
int amount_to_stuff = 0;
// require a certain error before bothering to fix it...
if (sync_error>config.tolerance) {
amount_to_stuff = -1;
}
if (sync_error<-config.tolerance) {
amount_to_stuff = 1;
}
// only allow stuffing if there is enough time to do it -- check DAC buffer...
if (current_delay<DAC_BUFFER_QUEUE_MINIMUM_LENGTH) {
// debug(1,"DAC buffer too short to allow stuffing.");
amount_to_stuff=0;
}
// try to keep the corrections definitely below 1 in 1000 audio frames
if (amount_to_stuff) {
uint32_t x = random()%1000;
if (x>352)
amount_to_stuff=0;
}
if ((amount_to_stuff==0) && (fix_volume==0x10000)) {
// 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...
config.output->play(inbuf, frame_size);
} 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);
break;
case ST_soxr:
// if (amount_to_stuff) debug(1,"Soxr stuff...");
play_samples = stuff_buffer_soxr(inbuf, outbuf,amount_to_stuff);
break;
}
#else
// if (amount_to_stuff) debug(1,"Standard stuff...");
play_samples = stuff_buffer_basic(inbuf, outbuf,amount_to_stuff);
#endif
/*
{
int co;
int is_silent=1;
short *p = outbuf;
for (co=0;co<play_samples;co++) {
if (*p!=0)
is_silent=0;
p++;
}
if (is_silent)
debug(1,"Silence!");
}
*/
config.output->play(outbuf, play_samples);
}
// check for loss of sync
// timestamp of zero means an inserted silent frame in place of a missing frame
if ((inframe->timestamp!=0) && (!please_stop) && (config.resyncthreshold!=0) && (abs(sync_error)>config.resyncthreshold)) {
sync_error_out_of_bounds++;
// debug(1,"Sync error out of bounds: Error: %lld; previous error: %lld; DAC: %lld; timestamp: %llx, time now %llx",sync_error,previous_sync_error,current_delay,inframe->timestamp,local_time_now);
if (sync_error_out_of_bounds>3) {
debug(1,"Lost sync with source for %d consecutive packets -- flushing and resyncing. Error: %lld.",sync_error_out_of_bounds,sync_error);
sync_error_out_of_bounds = 0;
player_flush(nt);
}
} else {
sync_error_out_of_bounds = 0;
}
// mark the frame as finished
inframe->timestamp=0;
inframe->sequence_number = 0;
// debug(1,"Sync error %lld frames. Amount to stuff %d." ,sync_error,amount_to_stuff);
// new stats calculation. We want a running average of sync error, drift, adjustment, number of additions+subtractions
if (number_of_statistics==trend_interval) {
// here we remove the oldest statistical data and take it from the summaries as well
tsum_of_sync_errors-=statistics[oldest_statistic].sync_error;
tsum_of_drifts -= statistics[oldest_statistic].drift;
if (statistics[oldest_statistic].correction>0)
tsum_of_insertions_and_deletions-=statistics[oldest_statistic].correction;
else
tsum_of_insertions_and_deletions+=statistics[oldest_statistic].correction;
tsum_of_corrections-=statistics[oldest_statistic].correction;
oldest_statistic=(oldest_statistic+1)%trend_interval;
number_of_statistics--;
}
statistics[newest_statistic].sync_error = sync_error;
statistics[newest_statistic].correction = amount_to_stuff;
if (number_of_statistics==0)
statistics[newest_statistic].drift = 0;
else
statistics[newest_statistic].drift = sync_error-previous_sync_error-previous_correction;
previous_sync_error = sync_error;
previous_correction = amount_to_stuff;
tsum_of_sync_errors += sync_error;
tsum_of_drifts += statistics[newest_statistic].drift;
if (amount_to_stuff>0)
tsum_of_insertions_and_deletions+=amount_to_stuff;
else
tsum_of_insertions_and_deletions-=amount_to_stuff;
tsum_of_corrections+=amount_to_stuff;
newest_statistic=(newest_statistic+1)%trend_interval;
number_of_statistics++;