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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 <inttypes.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;
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))
#ifdef HAVE_LIBPOLARSSL
static aes_context dctx;
#endif
//static pthread_t player_thread = NULL;
static int please_stop;
static int encrypted; // Normally the audio is encrypted, but it may not be
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 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_MINIMUM_LENGTH 600
typedef struct audio_buffer_entry { // decoded audio packets
int ready;
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)
// 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, time_since_play_started; // 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; // int32_t from seq_t, i.e. uint16_t, so okay
int32_t q = ab_read; // int32_t from seq_t, i.e. uint16_t, so okay
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 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);
int reply = 0; //everything okay
int outsize=bytes_per_audio_frame*(*destlen); // the size the output should be, in bytes
int toutsize = outsize;
if (encrypted) {
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);
alac_decode_frame(decoder_info, packet, dest, &outsize);
} else {
alac_decode_frame(decoder_info, buf, dest, &outsize);
}
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;
}
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(bytes_per_audio_frame*(frame_size+max_frame_size_change));
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) {
// ignore a request to flush that has been made before the first packet...
if (packet_count==0) {
pthread_mutex_lock(&flush_mutex);
flush_requested = 0;
flush_rtp_timestamp = 0;
pthread_mutex_unlock(&flush_mutex);
}
pthread_mutex_lock(&ab_mutex);
packet_count++;
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)
// debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
if ((flush_rtp_timestamp != 0) &&
((timestamp == flush_rtp_timestamp) || seq32_order(timestamp, flush_rtp_timestamp))) {
debug(3, "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, seqno);
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) {
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;
if (config.playback_mode==ST_mono) {
signed short *v = abuf->data;
int i;
int both;
for (i=frame_size;i;i--) {
int both = *v + *(v+1);
if (both > INT16_MAX) {
both = INT16_MAX;
} else if (both < INT16_MIN) {
both = INT16_MIN;
}
short sboth = (short)both;
*v++ = sboth;
*v++ = sboth;
}
}
} else {
debug(1,"Bad audio packet detected and discarded.");
abuf->ready = 0;
abuf->timestamp = 0;
abuf->sequence_number = 0;
}
}
// pthread_mutex_lock(&ab_mutex);
}
int rc = pthread_cond_signal(&flowcontrol);
if (rc)
debug(1, "Error signalling flowcontrol.");
}
pthread_mutex_unlock(&ab_mutex);
}
int32_t rand_in_range(int32_t exclusive_range_limit) {
static uint32_t lcg_prev = 12345;
// returns a pseudo random integer in the range 0 to (exclusive_range_limit-1) inclusive
int64_t sp = lcg_prev;
int64_t rl = exclusive_range_limit;
lcg_prev = lcg_prev * 69069 + 3; // crappy psrg
sp = sp*rl; // 64 bit calculation. INtersting part if above the 32 rightmost bits;
return sp >> 32;
}
static inline short dithered_vol(short sample) {
long out;
out = (long)sample * fix_volume;
if (fix_volume < 0x10000) {
// add a TPDF dither -- see http:https://www.users.qwest.net/%7Evolt42/cadenzarecording/DitherExplained.pdf
// and the discussion around https://www.hydrogenaud.io/forums/index.php?showtopic=16963&st=25
// I think, for a 32 --> 16 bits, the range of
// random numbers needs to be from -2^16 to 2^16, i.e. from -65536 to 65536 inclusive, not from -32768 to +32767
// See the original paper at http:https://www.ece.rochester.edu/courses/ECE472/resources/Papers/Lipshitz_1992.pdf
// by Lipshitz, Wannamaker and Vanderkooy, 1992.
long tpdf = rand_in_range(65536+1) - rand_in_range(65536+1);
// Check there's no clipping -- if there is,
if (tpdf>=0) {
if (LONG_MAX-tpdf>=out)
out += tpdf;
else
out = LONG_MAX;
} else {
if (LONG_MIN-tpdf<=out)
out += tpdf;
else
out = LONG_MIN;
}
}
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;
int notified_buffer_empty = 0; // diagnostic only
pthread_mutex_lock(&ab_mutex);
int wait;
long dac_delay = 0; // long because alsa returns a long
do {
// get the time
local_time_now = get_absolute_time_in_fp(); // type okay
// 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.dont_check_timeout == 0)) {
uint64_t ct = config.timeout; // go from int to 64-bit int
if ((local_time_now > time_of_last_audio_packet) &&
(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;
time_since_play_started = 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 ((ab_read!=ab_write) && (curframe->ready)) { // it could be synced and empty, under exceptional circumstances, with the frame unused, thus apparently 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;
}
curframe = audio_buffer + BUFIDX(ab_read);
if (curframe->ready) {
notified_buffer_empty=0; // at least one buffer now -- diagnostic only.
if (ab_buffering) { // if we are getting packets but not yet forwarding them to the player
int have_sent_prefiller_silence; // set true when we have send some silent frames to the DAC
uint32_t reference_timestamp;
uint64_t reference_timestamp_time,remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time);
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));
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
have_sent_prefiller_silence = 0;
// Here, calculate when we should start playing. We need to know when to allow the
// packets to be sent to the player.
// We will send packets of silence from now until that time and then we will send the
// first packet, which will be followed by the subsequent packets.
// 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, neglecting
// latencies, etc.
// It probably won't be the timestamp of our first packet, however, so we might have
// to do some calculations.
// To calculate when the first packet will be played, we figure out the exact time the
// packet should be played according to its timestamp and the reference time.
// We then need to add the desired latency, typically 88200 frames.
// Then we need to offset this by the backend latency offset. For example, if we knew
// that the audio back end has a latency of 100 ms, we would
// ask for the first packet to be emitted 100 ms earlier than it should, i.e. -4410
// frames, so that when it got through the audio back end,
// if would be in sync. To do this, we would give it a latency offset of -100 ms, i.e.
// -4410 frames.
int64_t delta = ((int64_t)first_packet_timestamp - (int64_t)reference_timestamp)+config.latency+config.audio_backend_latency_offset; // uint32_t to int64_t is okay and int32t to int64t promotion is okay.
if (delta>=0) {
uint64_t delta_fp_sec = (delta << 32) / 44100; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time+delta_fp_sec;
} else {
int64_t abs_delta = -delta;
uint64_t delta_fp_sec = (abs_delta << 32) / 44100; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
if (local_time_now >= first_packet_time_to_play) {
debug(
1,
"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) {
// recalculate first_packet_time_to_play -- the latency might change
int64_t delta = ((int64_t)first_packet_timestamp - (int64_t)reference_timestamp)+config.latency+config.audio_backend_latency_offset; // uint32_t to int64_t is okay and int32t to int64t promotion is okay.
if (delta>=0) {
uint64_t delta_fp_sec = (delta << 32) / 44100; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time+delta_fp_sec;
} else {
int64_t abs_delta = -delta;
uint64_t delta_fp_sec = (abs_delta << 32) / 44100; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
int64_t max_dac_delay = 4410;
int64_t filler_size = max_dac_delay; // 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;
time_since_play_started = 0;
} else {
// first_packet_time_to_play is definitely later than local_time_now
if ((config.output->delay) && (have_sent_prefiller_silence != 0)) {
int resp = config.output->delay(&dac_delay);
if (resp != 0) {
debug(1, "Error %d getting dac_delay in buffer_get_frame.",resp);
dac_delay = 0;
}
} else
dac_delay = 0;
int64_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 {
int64_t fs = filler_size;
if (fs > (max_dac_delay - dac_delay))
fs = max_dac_delay - dac_delay;
if (fs<0) {
debug(2,"frame size (fs) < 0 with max_dac_delay of %lld and dac_delay of %ld",max_dac_delay, dac_delay);
fs=0;
}
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;
//if (fs==0)
// 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(bytes_per_audio_frame*fs);
if (silence==NULL)
debug(1,"Failed to allocate %d byte silence buffer.",fs);
else {
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);
free(silence);
have_sent_prefiller_silence = 1;
}
}
}
}
}
if (ab_buffering == 0) {
// not the time of the playing of the first frame
uint64_t reference_timestamp_time; // don't need this...
get_reference_timestamp_stuff(&play_segment_reference_frame, &reference_timestamp_time, &play_segment_reference_frame_remote_time);
#ifdef CONFIG_METADATA
send_ssnc_metadata('prsm', NULL, 0, 0); // "resume", but don't wait if the queue is locked
#endif
}
}
}
}
// Here, we work out whether to release a packet or wait
// We release a buffer when the time is right.
// To work out when the time is right, we need to take account of (1) the actual time the packet
// should be released,
// (2) the latency requested, (3) the audio backend latency offset and (4) the desired length of
// the audio backend's buffer
// The time is right if the current time is later or the same as
// The packet time + (latency + latency offset - backend_buffer_length).
// Note: the last three items are expressed in frames and must be converted to time.
int do_wait = 0; // don't wait unless we can really prove we must
if ((ab_synced) && (curframe) && (curframe->ready) && (curframe->timestamp)) {
do_wait = 1; // if the current frame exists and is ready, then wait unless it's time to let it go...
uint32_t reference_timestamp;
uint64_t reference_timestamp_time,remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time); // all types okay
if (reference_timestamp) { // if we have a reference time
uint32_t packet_timestamp = curframe->timestamp; // types okay
int64_t delta = (int64_t)packet_timestamp - (int64_t)reference_timestamp; // uint32_t to int64_t is okay.
int64_t offset = config.latency + config.audio_backend_latency_offset -
config.audio_backend_buffer_desired_length; // all arguments are int32_t, so expression promotion okay
int64_t net_offset = delta + offset; // okay
uint64_t time_to_play = reference_timestamp_time; // type okay
if (net_offset >= 0) {
uint64_t net_offset_fp_sec = (net_offset << 32) / 44100; // int64_t which is positive
time_to_play += net_offset_fp_sec; // using the latency requested...
// debug(2,"Net Offset: %lld, adjusted: %lld.",net_offset,net_offset_fp_sec);
} else {
int64_t abs_net_offset = -net_offset;
uint64_t net_offset_fp_sec = (abs_net_offset << 32) / 44100; // int64_t which is positive
time_to_play -= net_offset_fp_sec;
// debug(2,"Net Offset: %lld, adjusted: -%lld.",net_offset,net_offset_fp_sec);
}
if (local_time_now >= time_to_play) {
do_wait = 0;
}
}
}
if (do_wait==0)
if ((ab_synced!=0) && (ab_read==ab_write)) { // the buffer is empty!
if (notified_buffer_empty==0) {
debug(1,"Buffers exhausted.");
notified_buffer_empty=1;
}
do_wait=1;
}
wait = (ab_buffering || (do_wait != 0) || (!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_AND_FREEBSD_AND_CYGWIN
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, "Supplying a silent frame for frame %u", read);
missing_packets++;
memset(curframe->data, 0, bytes_per_audio_frame*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, 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 = length;
if (stuff)
// stuffsamp = rand() % (length - 1);
stuffsamp =
(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
*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++;
}
// 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 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, 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;
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(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 > length + 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 + (length + stuff - gpm) * sizeof(short);
ip = inptr + (length - gpm) * sizeof(short);
for (i = 0; i < gpm; i++) {
*op++ = *ip++;
*op++ = *ip++;
}
// finally, adjust the volume, if necessary
if (fix_volume != 65536.0) {
// pthread_mutex_lock(&vol_mutex);
op = outptr;
for (i = 0; i < length + 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 < length; i++) {
*op++ = dithered_vol(*ip++);
*op++ = dithered_vol(*ip++);
};
// pthread_mutex_unlock(&vol_mutex);
}
return length + stuff;
}
#endif
typedef struct stats { // statistics for running averages
int64_t sync_error, correction, drift;
} stats_t;
static void *player_thread_func(void *arg) {
struct inter_threads_record itr;
itr.please_stop = 0;
// create and start the timing, control and audio receiver threads
pthread_t rtp_audio_thread, rtp_control_thread, rtp_timing_thread;
pthread_create(&rtp_audio_thread, NULL, &rtp_audio_receiver, (void *)&itr);
pthread_create(&rtp_control_thread, NULL, &rtp_control_receiver, (void *)&itr);
pthread_create(&rtp_timing_thread, NULL, &rtp_timing_receiver, (void *)&itr);
session_corrections = 0;
play_segment_reference_frame = 0; // zero signals that we are not in a play segment
// check that there are enough buffers to accommodate the desired latency and the latency offset