void player_put_packet(seq_t seqno, uint8_t *data, int len) {
abuf_t *abuf = 0;
int16_t buf_fill;
pthread_mutex_lock(&ab_mutex);
if (!ab_synced) {
debug(1, "syncing to first seqno %04X\n", seqno);
ab_write = seqno-1;
ab_read = seqno;
ab_synced = 1;
}
if (seq_diff(ab_write, seqno) == 1) { // expected packet
abuf = audio_buffer + BUFIDX(seqno);
ab_write = seqno;
} else if (seq_order(ab_write, seqno)) { // newer than expected
// Be careful with new packets that are in advance
// Those more than a buffer size in advance will cause an Overrun
// When buffering the valid threshold should be the buffer fill target itself which should re-sync
if (ab_buffering && (seq_diff(ab_read, seqno) > config.buffer_start_fill)) {
warn("out of range re-sync %04X (%04X:%04X)", seqno, ab_read, ab_write);
ab_resync();
ab_synced = 1;
ab_read = seqno;
ab_write = seqno;
abuf = audio_buffer + BUFIDX(seqno);
} else {
debug(1, "advance packet %04X (%04X:%04X)\n", seqno, ab_read, ab_write);
rtp_request_resend(ab_write+1, seqno-1);
abuf = audio_buffer + BUFIDX(seqno);
ab_write = seqno;
}
} else if (seq_order(ab_read, seqno)) { // late but not yet played
abuf = audio_buffer + BUFIDX(seqno);
if (abuf->ready) { // discard this frame if frame previously received
abuf =0;
debug(1, "duplicate packet %04X (%04X:%04X)\n", seqno, ab_read, ab_write);
}
} else { // too late.
debug(1, "late packet %04X (%04X:%04X)\n", seqno, ab_read, ab_write);
}
buf_fill = seq_diff(ab_read, ab_write);
pthread_mutex_unlock(&ab_mutex);
if (abuf) {
alac_decode(abuf->data, data, len);
abuf->ready = 1;
}
pthread_mutex_lock(&ab_mutex);
if (ab_buffering && buf_fill >= config.buffer_start_fill) {
debug(1, "buffering over. starting play (%04X:%04X)\n", ab_read, ab_write);
ab_buffering = 0;
}
pthread_mutex_unlock(&ab_mutex);
}
void buffer_put_packet(seq_t seqno, char *data, int len) {
volatile abuf_t *abuf = 0;
short read;
short buf_fill;
pthread_mutex_lock(&ab_mutex);
if (!ab_synced) {
ab_write = seqno;
ab_read = seqno-1;
ab_synced = 1;
}
if (seqno == ab_write+1) { // expected packet
abuf = audio_buffer + BUFIDX(seqno);
ab_write = seqno;
} else if (seq_order(ab_write, seqno)) { // newer than expected
rtp_request_resend(ab_write, seqno-1);
abuf = audio_buffer + BUFIDX(seqno);
ab_write = seqno;
} else if (seq_order(ab_read, seqno)) { // late but not yet played
abuf = audio_buffer + BUFIDX(seqno);
} else { // too late.
fprintf(stderr, "\nlate packet %04X (%04X:%04X)\n", seqno, ab_read, ab_write);
}
buf_fill = ab_write - ab_read;
pthread_mutex_unlock(&ab_mutex);
if (abuf) {
alac_decode(abuf->data, data, len);
abuf->ready = 1;
}
if (ab_buffering && buf_fill >= buffer_start_fill) {
ab_buffering = 0;
pthread_cond_signal(&ab_buffer_ready);
}
if (!ab_buffering) {
// check if the t+10th packet has arrived... last-chance resend
read = ab_read + 10;
abuf = audio_buffer + BUFIDX(read);
if (abuf->ready != 1) {
rtp_request_resend(read, read);
abuf->ready = -1;
}
}
}
void player_put_packet(seq_t seqno, uint8_t *data, int len) {
abuf_t *abuf = 0;
int16_t buf_fill;
pthread_mutex_lock(&ab_mutex);
if (!ab_synced) {
debug(2, "syncing to first seqno %04X\n", seqno);
ab_write = seqno-1;
ab_read = seqno;
ab_synced = 1;
}
if (seq_diff(ab_write, seqno) == 1) { // expected packet
abuf = audio_buffer + BUFIDX(seqno);
ab_write = seqno;
} else if (seq_order(ab_write, seqno)) { // newer than expected
rtp_request_resend(ab_write+1, seqno-1);
abuf = audio_buffer + BUFIDX(seqno);
ab_write = seqno;
} else if (seq_order(ab_read, seqno)) { // late but not yet played
abuf = audio_buffer + BUFIDX(seqno);
} else { // too late.
debug(1, "late packet %04X (%04X:%04X)", seqno, ab_read, ab_write);
}
buf_fill = seq_diff(ab_read, ab_write);
pthread_mutex_unlock(&ab_mutex);
if (abuf) {
alac_decode(abuf->data, data, len);
abuf->ready = 1;
}
pthread_mutex_lock(&ab_mutex);
if (ab_buffering && buf_fill >= config.buffer_start_fill) {
debug(1, "buffering over. starting play\n");
ab_buffering = 0;
bf_est_reset(buf_fill);
}
pthread_mutex_unlock(&ab_mutex);
}
void rtp_request_resend(seq_t first, seq_t last) {
if (seq_order(last, first))
return;
fprintf(stderr, "requesting resend on %d packets (port %d)\n", last-first+1, controlport);
char req[8]; // *not* a standard RTCP NACK
req[0] = 0x80;
req[1] = 0x55|0x80; // Apple 'resend'
*(unsigned short *)(req+2) = htons(1); // our seqnum
*(unsigned short *)(req+4) = htons(first); // missed seqnum
*(unsigned short *)(req+6) = htons(last-first+1); // count
rtp_client.sin_port = htons(controlport);
sendto(rtp_sockets[1], req, sizeof(req), 0, (struct sockaddr *)&rtp_client, sizeof(struct sockaddr_in));
}
static inline void rtp_request_resend(seq_t first, seq_t last) {
if (seq_order(last, first))
return;
#ifdef DEBUG
if ((last-first) > 1000) {
fprintf(stderr, "resend too much last=%d first=%d\n",last, first);
}
fprintf(stderr, "requesting resend on %d packets (port %d) last=%d first=%d\n", last-first+1, controlport, last, first);
#endif
char req[8]; // *not* a standard RTCP NACK
req[0] = 0x80;
req[1] = 0x55|0x80; // Apple 'resend'
*(unsigned short *)(req+2) = htons(1); // our seqnum
*(unsigned short *)(req+4) = htons(first); // missed seqnum
*(unsigned short *)(req+6) = htons(last-first+1); // count
#ifdef AF_INET6
rtp_client.sin6_port = htons(controlport);
#else
rtp_client.sin_port = htons(controlport);
#endif
sendto(rtp_sockets[1], req, sizeof(req), 0, (struct sockaddr *)&rtp_client, sizeof(rtp_client));
}
// 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.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 (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;
}
curframe = audio_buffer + BUFIDX(ab_read);
if (curframe->ready) {
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,remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_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, 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);
first_packet_time_to_play =
reference_timestamp_time +
((delta + (int64_t)config.latency + (int64_t)config.audio_backend_latency_offset)
<< 32) /
44100;
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) {
uint32_t filler_size = frame_size;
uint32_t max_dac_delay = 4410;
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;
time_since_play_started = 0;
} else {
if (config.output->delay) {
dac_delay = config.output->delay();
if (dac_delay == -1) {
debug(1, "Error getting dac_delay in buffer_get_frame.");
dac_delay = 0;
}
} else
dac_delay = 0;
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);
if (ab_buffering == 0) {
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 = 1;
if ((ab_synced) && (curframe) && (curframe->ready) && (curframe->timestamp)) {
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 (reference_timestamp) { // if we have a reference time
uint32_t packet_timestamp = curframe->timestamp;
int64_t delta = ((int64_t)packet_timestamp - (int64_t)reference_timestamp);
int64_t offset = (int64_t)config.latency + config.audio_backend_latency_offset -
(int64_t)config.audio_backend_buffer_desired_length;
int64_t net_offset = delta + offset;
int64_t time_to_play = reference_timestamp_time;
int64_t net_offset_fp_sec;
if (net_offset >= 0) {
net_offset_fp_sec = (net_offset << 32) / 44100;
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 {
net_offset_fp_sec = ((-net_offset) << 32) / 44100;
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;
}
}
}
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
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);
void player_put_packet(seq_t seqno, uint32_t timestamp, uint8_t *data, int len) {
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) &&
((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 void buffer_put_packet(seq_t seqno, char *data, int len) {
abuf_t *abuf = 0;
abuf_t *tmpbuf = 0;
unsigned short buf_fill;
pthread_mutex_lock(&ab_mutex);
/* sync is required. */
if (!ab_synced) {
ab_write = seqno;
if (seqno == 0) {
ab_read = 0;
} else {
ab_read = seqno-1;
}
ab_synced = 1;
}
/* Check seq_no is what we want. */
buf_fill=BUF_FILL_SIZE(seqno,ab_read);
if (buf_fill > 1000) {
pthread_mutex_unlock(&ab_mutex);
return;
}
/* Check buf slot is occupied. */
tmpbuf = audio_buffer + BUFIDX(seqno); // buffering fits here!
if (tmpbuf->ready) {
pthread_mutex_unlock(&ab_mutex);
return;
}
/* if seqno moves faster than expected. */
if (seq_order(ab_write, seqno)) { // late but not yet played
rtp_request_resend(ab_write+1, seqno-1);
}
/* update ab_write */
ab_write = seqno;
abuf = audio_buffer + BUFIDX(seqno); // buffering fits here!
/* Make sure the slot of abuf is empty! */
if (!abuf->ready) {
memcpy(abuf->data, data, len);
abuf->len=len;
abuf->ready = 1;
}
buf_fill=BUF_FILL_SIZE(ab_write,ab_read);
#ifdef DEBUG
fprintf(stderr,"seqno=%d ab_write=%d ab_read=%d buf_fill=%d ab_buffering=%d \n",seqno,ab_write,ab_read,buf_fill,ab_buffering);
#endif
if (ab_buffering && (buf_fill >= (buffer_start_fill-1))) {
ab_buffering = 0;
pthread_cond_signal(&ab_buffer_ready);
}
if (buf_fill >= (BUFFER_FRAMES-1)) { // overrunning!
#ifdef DEBUG
fprintf(stderr,"overrunning.\n");
#endif
pthread_cond_wait(&ab_buffer_full, &ab_mutex);
buf_fill=BUF_FILL_SIZE(ab_write,ab_read);
}
pthread_mutex_unlock(&ab_mutex);
}
