// get the next frame, when available. return 0 if underrun/stream reset. static short *buffer_get_frame(void) { int16_t buf_fill; seq_t read, next; abuf_t *abuf = 0; int i; if (ab_buffering) return 0; pthread_mutex_lock(&ab_mutex); buf_fill = seq_diff(ab_read, ab_write); if (buf_fill < 1 || !ab_synced) { if (buf_fill < 1) warn("underrun %i (%04X:%04X)", buf_fill, ab_read, ab_write); ab_resync(); pthread_mutex_unlock(&ab_mutex); return 0; } if (buf_fill >= BUFFER_FRAMES) { // overrunning! uh-oh. restart at a sane distance warn("overrun %i (%04X:%04X)", buf_fill, ab_read, ab_write); read = ab_read; ab_read = ab_write - config.buffer_start_fill; ab_reset((ab_write + 1 - BUFFER_FRAMES), ab_read); // reset any ready frames in those we've skipped (avoiding wrap around) } read = ab_read; ab_read++; buf_fill = seq_diff(ab_read, ab_write); bf_est_update(buf_fill); // check if t+16, t+32, t+64, t+128, ... resend focus boundary // packets have arrived... last-chance resend if (!ab_buffering) { for (i = 16; i <= resend_focus(config.buffer_start_fill); i = (i * 2)) { next = ab_read + i; abuf = audio_buffer + BUFIDX(next); if ((!abuf->ready) && (next < ab_write)){ rtp_request_resend(next, next); debug(1, "last chance resend T+%i, %04X (%04X:%04X)\n", i, next, ab_read, ab_write); } } } abuf_t *curframe = audio_buffer + BUFIDX(read); if (!curframe->ready) { debug(1, "missing frame %04X\n", read); memset(curframe->data, 0, FRAME_BYTES(frame_size)); } curframe->ready = 0; pthread_mutex_unlock(&ab_mutex); return curframe->data; }
// get the next frame, when available. return 0 if underrun/stream reset. static short *buffer_get_frame(void) { int16_t buf_fill; seq_t read, next; abuf_t *abuf = 0; int i; if (ab_buffering) return 0; pthread_mutex_lock(&ab_mutex); buf_fill = seq_diff(ab_read, ab_write); if (buf_fill < 1 || !ab_synced) { if (buf_fill < 1) warn("underrun."); ab_buffering = 1; pthread_mutex_unlock(&ab_mutex); return 0; } if (buf_fill >= BUFFER_FRAMES) { // overrunning! uh-oh. restart at a sane distance warn("overrun."); ab_read = ab_write - config.buffer_start_fill; } read = ab_read; ab_read++; buf_fill = seq_diff(ab_read, ab_write); bf_est_update(buf_fill); // check if t+16, t+32, t+64, t+128, ... (buffer_start_fill / 2) // packets have arrived... last-chance resend if (!ab_buffering) { for (i = 16; i < (config.buffer_start_fill / 2); i = (i * 2)) { next = ab_read + i; abuf = audio_buffer + BUFIDX(next); if (!abuf->ready) { rtp_request_resend(next, next); } } } abuf_t *curframe = audio_buffer + BUFIDX(read); if (!curframe->ready) { debug(1, "missing frame %04X.", read); memset(curframe->data, 0, FRAME_BYTES(frame_size)); } curframe->ready = 0; pthread_mutex_unlock(&ab_mutex); return curframe->data; }
static void alac_decode(short *dest, uint8_t *buf, int len) { unsigned char packet[MAX_PACKET]; assert(len<=MAX_PACKET); unsigned char iv[16]; int aeslen = len & ~0xf; memcpy(iv, aesiv, sizeof(iv)); AES_cbc_encrypt(buf, packet, aeslen, &aes, iv, AES_DECRYPT); memcpy(packet+aeslen, buf+aeslen, len-aeslen); int outsize; alac_decode_frame(decoder_info, packet, dest, &outsize); assert(outsize == FRAME_BYTES(frame_size)); }
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); int 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); } assert(outsize == FRAME_BYTES(frame_size)); }
// 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);