/* ============================================================================= * delete_node * ============================================================================= */ static node_t* delete_node (rbtree_t* s, node_t* p) { /* * If strictly internal, copy successor's element to p and then make p * point to successor */ if (LDNODE(p, l) != NULL && LDNODE(p, r) != NULL) { node_t* s = SUCCESSOR(p); STF(p, k, LDNODE(s, k)); STF(p, v, LDNODE(s, v)); p = s; } /* p has 2 children */ /* Start fixup at replacement node, if it exists */ node_t* replacement = ((LDNODE(p, l) != NULL) ? LDNODE(p, l) : LDNODE(p, r)); if (replacement != NULL) { /* Link replacement to parent */ /* TODO: precompute pp = p->p and substitute below ... */ STF (replacement, p, LDNODE(p, p)); node_t* pp = LDNODE(p, p); if (pp == NULL) { STF(s, root, replacement); } else if (p == LDNODE(pp, l)) { STF(pp, l, replacement); } else { STF(pp, r, replacement); } /* Null out links so they are OK to use by fixAfterDeletion */ STF(p, l, NULL); STF(p, r, NULL); STF(p, p, NULL); /* Fix replacement */ if (LDF(p,c) == BLACK) { FIX_AFTER_DELETION(s, replacement); } } else if (LDNODE(p, p) == NULL) { /* return if we are the only node */ STF(s, root, NULL); } else { /* No children. Use self as phantom replacement and unlink */ if (LDF(p, c) == BLACK) { FIX_AFTER_DELETION(s, p); } node_t* pp = LDNODE(p, p); if (pp != NULL) { if (p == LDNODE(pp, l)) { STF(pp,l, NULL); } else if (p == LDNODE(pp, r)) { STF(pp, r, NULL); } STF(p, p, NULL); } } return p; }
// 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); }
char * merge_sort (char *list, int length, int offset, int (*compare) (char *, char *)) { char *first_list; char *second_list; int first_length; int second_length; char *result; char **merge_point; char *cursor; int counter; #define SUCCESSOR(Pointer) \ (*((char **) (((char *) (Pointer)) + offset))) if (length == 1) return list; if (length == 2) { if ((*compare) (list, SUCCESSOR (list)) > 0) { result = SUCCESSOR (list); SUCCESSOR (result) = list; SUCCESSOR (list) = NULL; return result; } return list; } first_list = list; first_length = (length + 1) / 2; second_length = length / 2; for (cursor = list, counter = first_length - 1; counter; cursor = SUCCESSOR (cursor), counter--) continue; second_list = SUCCESSOR (cursor); SUCCESSOR (cursor) = NULL; first_list = merge_sort (first_list, first_length, offset, compare); second_list = merge_sort (second_list, second_length, offset, compare); merge_point = &result; while (first_list && second_list) if ((*compare) (first_list, second_list) < 0) { cursor = SUCCESSOR (first_list); *merge_point = first_list; merge_point = &SUCCESSOR (first_list); first_list = cursor; } else { cursor = SUCCESSOR (second_list); *merge_point = second_list; merge_point = &SUCCESSOR (second_list); second_list = cursor; } if (first_list) *merge_point = first_list; else *merge_point = second_list; return result; #undef SUCCESSOR }
static struct name * merge_sort_sll (struct name *list, int length, int (*compare) (struct name const*, struct name const*)) { struct name *first_list; struct name *second_list; int first_length; int second_length; struct name *result; struct name **merge_point; struct name *cursor; int counter; # define SUCCESSOR(name) ((name)->next) if (length == 1) return list; if (length == 2) { if ((*compare) (list, SUCCESSOR (list)) > 0) { result = SUCCESSOR (list); SUCCESSOR (result) = list; SUCCESSOR (list) = 0; return result; } return list; } first_list = list; first_length = (length + 1) / 2; second_length = length / 2; for (cursor = list, counter = first_length - 1; counter; cursor = SUCCESSOR (cursor), counter--) continue; second_list = SUCCESSOR (cursor); SUCCESSOR (cursor) = 0; first_list = merge_sort_sll (first_list, first_length, compare); second_list = merge_sort_sll (second_list, second_length, compare); merge_point = &result; while (first_list && second_list) if ((*compare) (first_list, second_list) < 0) { cursor = SUCCESSOR (first_list); *merge_point = first_list; merge_point = &SUCCESSOR (first_list); first_list = cursor; } else { cursor = SUCCESSOR (second_list); *merge_point = second_list; merge_point = &SUCCESSOR (second_list); second_list = cursor; } if (first_list) *merge_point = first_list; else *merge_point = second_list; return result; #undef SUCCESSOR }