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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 <assert.h>
#include <errno.h>
#include <fcntl.h>
#include <inttypes.h>
#include <limits.h>
#include <math.h>
#include <pthread.h>
#include <stdio.h>
#include <stdlib.h>
#include <stdlib.h>
#include <string.h>
#include <sys/stat.h>
#include <sys/syslog.h>
#include <sys/types.h>
#include <unistd.h>
#include "config.h"
#ifdef HAVE_LIBMBEDTLS
#include <mbedtls/aes.h>
#include <mbedtls/havege.h>
#endif
#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
#ifdef CONFIG_CONVOLUTION
#include <FFTConvolver/convolver.h>
#endif
#if defined(HAVE_DBUS) || defined(HAVE_MPRIS)
#include "dacp.h"
#include <glib.h>
#endif
#ifdef HAVE_DBUS
#include "dbus-interface.h"
#include "dbus-service.h"
#endif
#ifdef HAVE_MPRIS
#include "mpris-interface.h"
#include "mpris-player-interface.h"
#include "mpris-service.h"
#endif
#include "common.h"
#include "player.h"
#include "rtp.h"
#include "rtsp.h"
#include "alac.h"
#ifdef HAVE_APPLE_ALAC
#include "apple_alac.h"
#endif
#include "loudness.h"
// 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
// static abuf_t audio_buffer[BUFFER_FRAMES];
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
// make timestamps and seqnos definitely monotonic
// add an epoch to the timestamp. The monotonic timestamp guaranteed to start between 2^32 and 2^33
// frames and continue up to 2^63-1 frames
// if should never get into the negative range
// which is about 2*10^8 * 1,000 seconds at 384,000 frames per second -- about 2 trillion seconds or
// over 50,000 years.
// also, it won't reach zero until then, if ever, so we can safely say that a null monotonic
// timestamp can mean something special
int64_t monotonic_timestamp(uint32_t timestamp, rtsp_conn_info *conn) {
int64_t previous_value;
int64_t return_value;
if (conn->timestamp_epoch == 0) {
if (timestamp > conn->maximum_timestamp_interval)
conn->timestamp_epoch = 1;
else
conn->timestamp_epoch = 2;
previous_value = conn->timestamp_epoch;
previous_value <<= 32;
previous_value += timestamp;
} else {
previous_value = conn->timestamp_epoch;
previous_value <<= 32;
previous_value += conn->last_timestamp;
if (timestamp < conn->last_timestamp) {
// the incoming timestamp is less than the last one.
// if the difference is more than a minute, assume it's really from the next epoch
if ((conn->last_timestamp - timestamp) > conn->maximum_timestamp_interval)
conn->timestamp_epoch++;
} else {
// the incoming timestamp is greater than the last one.
// if the difference is more than a minute, assume it's really from the previous epoch
if ((timestamp - conn->last_timestamp) > conn->maximum_timestamp_interval)
conn->timestamp_epoch--;
}
}
conn->last_timestamp = timestamp;
return_value = conn->timestamp_epoch;
return_value <<= 32;
return_value += timestamp;
if (previous_value > return_value) {
if ((previous_value - return_value) > conn->maximum_timestamp_interval)
debug(2, "interval between successive rtptimes greater than allowed!");
} else {
if ((return_value - previous_value) > conn->maximum_timestamp_interval)
debug(2, "interval between successive rtptimes greater than allowed!");
}
if (return_value < 0)
debug(1, "monotonic rtptime is negative!");
return return_value;
}
static void ab_resync(rtsp_conn_info *conn) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++) {
conn->audio_buffer[i].ready = 0;
conn->audio_buffer[i].sequence_number = 0;
}
conn->ab_synced = 0;
conn->last_seqno_read = -1;
conn->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, seq_t base) {
int32_t p = x; // int32_t from seq_t, i.e. uint16_t, so okay
int32_t q = base; // 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, seq_t base) {
int32_t diff = ORDINATE(b, base) - ORDINATE(a, base);
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, seq_t base) {
int32_t d = ORDINATE(b, base) - ORDINATE(a, base);
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, rtsp_conn_info *conn) {
// 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 = conn->input_bytes_per_frame * (*destlen); // the size the output should be, in bytes
int toutsize = outsize;
if (conn->stream.encrypted) {
unsigned char iv[16];
int aeslen = len & ~0xf;
memcpy(iv, conn->stream.aesiv, sizeof(iv));
#ifdef HAVE_LIBMBEDTLS
mbedtls_aes_crypt_cbc(&conn->dctx, MBEDTLS_AES_DECRYPT, aeslen, iv, buf, packet);
#endif
#ifdef HAVE_LIBPOLARSSL
aes_crypt_cbc(&conn->dctx, AES_DECRYPT, aeslen, iv, buf, packet);
#endif
#ifdef HAVE_LIBSSL
AES_cbc_encrypt(buf, packet, aeslen, &conn->aes, iv, AES_DECRYPT);
#endif
memcpy(packet + aeslen, buf + aeslen, len - aeslen);
#ifdef HAVE_APPLE_ALAC
if (config.use_apple_decoder) {
if (conn->decoder_in_use != 1 << decoder_apple_alac) {
debug(1, "Apple ALAC Decoder used on encrypted audio.");
conn->decoder_in_use = 1 << decoder_apple_alac;
}
apple_alac_decode_frame(packet, len, (unsigned char *)dest, &outsize);
outsize = outsize * 4; // bring the size to bytes
} else
#endif
{
if (conn->decoder_in_use != 1 << decoder_hammerton) {
debug(1, "Hammerton Decoder used on encrypted audio.");
conn->decoder_in_use = 1 << decoder_hammerton;
}
alac_decode_frame(conn->decoder_info, packet, (unsigned char *)dest, &outsize);
}
} else {
// not encrypted
#ifdef HAVE_APPLE_ALAC
if (config.use_apple_decoder) {
if (conn->decoder_in_use != 1 << decoder_apple_alac) {
debug(1, "Apple ALAC Decoder used on unencrypted audio.");
conn->decoder_in_use = 1 << decoder_apple_alac;
}
apple_alac_decode_frame(buf, len, (unsigned char *)dest, &outsize);
outsize = outsize * 4; // bring the size to bytes
} else
#endif
{
if (conn->decoder_in_use != 1 << decoder_hammerton) {
debug(1, "Hammerton Decoder used on unencrypted audio.");
conn->decoder_in_use = 1 << decoder_hammerton;
}
alac_decode_frame(conn->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, conn->stream.encrypted);
reply = -1; // output packet is the wrong size
}
*destlen = outsize / conn->input_bytes_per_frame;
if ((outsize % conn->input_bytes_per_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, conn->input_bytes_per_frame);
return reply;
}
static int init_decoder(int32_t fmtp[12], rtsp_conn_info *conn) {
// This is a guess, but the format of the fmtp looks identical to the format of an
// ALACSpecificCOnfig
// which is detailed in the file ALACMagicCookieDescription.txt in the Apple ALAC sample
// implementation
// Here it is:
/*
struct ALACSpecificConfig (defined in ALACAudioTypes.h)
abstract This struct is used to describe codec provided information about the encoded
Apple Lossless bitstream.
It must accompany the encoded stream in the containing audio file and be provided
to the decoder.
field frameLength uint32_t indicating the frames per packet
when
no
explicit
frames per packet setting is
present in the packet header. The
encoder frames per packet can be explicitly set
but for maximum compatibility, the
default encoder setting of 4096 should be used.
field compatibleVersion uint8_t indicating compatible version,
value must be set to 0
field bitDepth uint8_t describes the bit depth of the
source
PCM
data
(maximum
value = 32)
field pb uint8_t currently unused tuning
parametetbugr.
value should be set to 40
field mb uint8_t currently unused tuning parameter.
value should be set to 14
field kb uint8_t currently unused tuning parameter.
value should be set to 10
field numChannels uint8_t describes the channel count (1 =
mono,
2
=
stereo,
etc...)
when channel layout info is not provided
in the 'magic cookie', a channel count > 2
describes a set of discreet channels
with no specific ordering
field maxRun uint16_t currently unused.
value should be set to 255
field maxFrameBytes uint32_t the maximum size of an Apple
Lossless
packet
within
the encoded stream.
value of 0 indicates unknown
field avgBitRate uint32_t the average bit rate in bits per
second
of
the
Apple
Lossless stream.
value of 0 indicates unknown
field sampleRate uint32_t sample rate of the encoded stream
*/
// We are going to go on that basis
alac_file *alac;
conn->max_frames_per_packet = fmtp[1]; // number of audio frames per packet.
conn->input_rate = fmtp[11];
conn->input_num_channels = fmtp[7];
conn->input_bit_depth = fmtp[3];
conn->input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
alac = alac_create(conn->input_bit_depth, conn->input_num_channels);
if (!alac)
return 1;
conn->decoder_info = alac;
alac->setinfo_max_samples_per_frame = conn->max_frames_per_packet;
alac->setinfo_7a = fmtp[2];
alac->setinfo_sample_size = conn->input_bit_depth;
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);
#ifdef HAVE_APPLE_ALAC
apple_alac_init(fmtp);
#endif
return 0;
}
static void terminate_decoders(rtsp_conn_info *conn) {
alac_free(conn->decoder_info);
#ifdef HAVE_APPLE_ALAC
apple_alac_terminate();
#endif
}
static void init_buffer(rtsp_conn_info *conn) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++)
conn->audio_buffer[i].data = malloc(
conn->input_bytes_per_frame * (conn->max_frames_per_packet + conn->max_frame_size_change));
ab_resync(conn);
}
static void free_audio_buffers(rtsp_conn_info *conn) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++)
free(conn->audio_buffer[i].data);
}
void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
rtsp_conn_info *conn) {
// all timestamps are done at the output rate
int64_t ltimestamp = timestamp * conn->output_sample_ratio;
// ignore a request to flush that has been made before the first packet...
if (conn->packet_count == 0) {
pthread_mutex_lock(&conn->flush_mutex);
conn->flush_requested = 0;
conn->flush_rtp_timestamp = 0;
pthread_mutex_unlock(&conn->flush_mutex);
}
pthread_mutex_lock(&conn->ab_mutex);
conn->packet_count++;
conn->time_of_last_audio_packet = get_absolute_time_in_fp();
if (conn->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 ((conn->flush_rtp_timestamp != 0) && (ltimestamp <= conn->flush_rtp_timestamp)) {
debug(3,
"Dropping flushed packet in player_put_packet, seqno %u, timestamp %lld, flushing to "
"timestamp: %lld.",
seqno, ltimestamp, conn->flush_rtp_timestamp);
} else {
if ((conn->flush_rtp_timestamp != 0x0) &&
(ltimestamp > conn->flush_rtp_timestamp)) // if we have gone past the flush boundary time
conn->flush_rtp_timestamp = 0x0;
abuf_t *abuf = 0;
if (!conn->ab_synced) {
debug(2, "syncing to seqno %u.", seqno);
conn->ab_write = seqno;
conn->ab_read = seqno;
conn->ab_synced = 1;
}
if (conn->ab_write == seqno) { // expected packet
abuf = conn->audio_buffer + BUFIDX(seqno);
conn->ab_write = SUCCESSOR(seqno);
} else if (seq_order(conn->ab_write, seqno, conn->ab_read)) { // 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(conn->ab_write, seqno, conn->ab_read);
if (gap <= 0)
debug(1, "Unexpected gap size: %d.", gap);
int i;
for (i = 0; i < gap; i++) {
abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->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 = conn->audio_buffer + BUFIDX(seqno);
// rtp_request_resend(ab_write, gap);
// resend_requests++;
conn->ab_write = SUCCESSOR(seqno);
} else if (seq_order(conn->ab_read, seqno, conn->ab_read)) { // late but not yet played
conn->late_packets++;
abuf = conn->audio_buffer + BUFIDX(seqno);
} else { // too late.
conn->too_late_packets++;
}
// pthread_mutex_unlock(&ab_mutex);
if (abuf) {
int datalen = conn->max_frames_per_packet;
if (alac_decode(abuf->data, &datalen, data, len, conn) == 0) {
abuf->ready = 1;
abuf->length = datalen;
abuf->timestamp = ltimestamp;
abuf->sequence_number = seqno;
} 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(&conn->flowcontrol);
if (rc)
debug(1, "Error signalling flowcontrol.");
}
pthread_mutex_unlock(&conn->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 void process_sample(int32_t sample, char **outp, enum sps_format_t format, int volume,
int dither, rtsp_conn_info *conn) {
int64_t hyper_sample = sample;
int result;
if (config.loudness) {
hyper_sample <<=
32; // Do not apply volume as it has already been done with the Loudness DSP filter
} else {
int64_t hyper_volume = (int64_t)volume << 16;
hyper_sample =
hyper_sample * hyper_volume; // this is 64 bit bit multiplication -- we may need to
// dither it down to its
// target resolution
}
// next, do dither, if necessary
if (dither) {
// add a TPDF dither -- see
// http://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://www.ece.rochester.edu/courses/ECE472/resources/Papers/Lipshitz_1992.pdf
// by Lipshitz, Wannamaker and Vanderkooy, 1992.
int64_t dither_mask;
switch (format) {
case SPS_FORMAT_S32:
dither_mask = (int64_t)1 << (64 + 1 - 32);
break;
case SPS_FORMAT_S24:
case SPS_FORMAT_S24_3LE:
case SPS_FORMAT_S24_3BE:
dither_mask = (int64_t)1 << (64 + 1 - 24);
break;
case SPS_FORMAT_S16:
dither_mask = (int64_t)1 << (64 + 1 - 16);
break;
case SPS_FORMAT_S8:
case SPS_FORMAT_U8:
dither_mask = (int64_t)1 << (64 + 1 - 8);
break;
case SPS_FORMAT_UNKNOWN:
die("Unexpected SPS_FORMAT_UNKNOWN while calculating dither mask.");
}
dither_mask -= 1;
// int64_t r = r64i();
int64_t r = ranarray64i(); // use an array of precalculated pseudorandom numbers rather than
// calculating them on the fly. Should be easier on low-powered
// processors
int64_t tpdf = (r & dither_mask) - (conn->previous_random_number & dither_mask);
conn->previous_random_number = r;
// add dither, allowing for clipping
if (tpdf >= 0) {
if (INT64_MAX - tpdf >= hyper_sample)
hyper_sample += tpdf;
else
hyper_sample = INT64_MAX;
} else {
if (INT64_MIN - tpdf <= hyper_sample)
hyper_sample += tpdf;
else
hyper_sample = INT64_MIN;
}
// dither is complete here
}
// move the result to the desired position in the int64_t
char *op = *outp;
uint8_t byt;
switch (format) {
case SPS_FORMAT_S32:
hyper_sample >>= (64 - 32);
*(int32_t *)op = hyper_sample;
result = 4;
break;
case SPS_FORMAT_S24_3LE:
hyper_sample >>= (64 - 24);
byt = (uint8_t)hyper_sample;
*op++ = byt;
byt = (uint8_t)(hyper_sample >> 8);
*op++ = byt;
byt = (uint8_t)(hyper_sample >> 16);
*op++ = byt;
result = 3;
break;
case SPS_FORMAT_S24_3BE:
hyper_sample >>= (64 - 24);
byt = (uint8_t)(hyper_sample >> 16);
*op++ = byt;
byt = (uint8_t)(hyper_sample >> 8);
*op++ = byt;
byt = (uint8_t)hyper_sample;
*op++ = byt;
result = 3;
break;
case SPS_FORMAT_S24:
hyper_sample >>= (64 - 24);
*(int32_t *)op = hyper_sample;
result = 4;
break;
case SPS_FORMAT_S16:
hyper_sample >>= (64 - 16);
*(int16_t *)op = (int16_t)hyper_sample;
result = 2;
break;
case SPS_FORMAT_S8:
hyper_sample >>= (int8_t)(64 - 8);
*op = hyper_sample;
result = 1;
break;
case SPS_FORMAT_U8:
hyper_sample >>= (uint8_t)(64 - 8);
hyper_sample += 128;
*op = hyper_sample;
result = 1;
break;
case SPS_FORMAT_UNKNOWN:
die("Unexpected SPS_FORMAT_UNKNOWN while outputting samples");
}
*outp += result;
}
// get the next frame, when available. return 0 if underrun/stream reset.
static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
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(&conn->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 ((conn->time_of_last_audio_packet != 0) && (conn->stop == 0) &&
(config.dont_check_timeout == 0)) {
uint64_t ct = config.timeout; // go from int to 64-bit int
// if (conn->packet_count>500) { //for testing -- about 4 seconds of play first
if ((local_time_now > conn->time_of_last_audio_packet) &&
(local_time_now - conn->time_of_last_audio_packet >= ct << 32)) {
debug(1, "As Yeats almost said, \"Too long a silence / can make a stone of the heart\" "
"from RTSP conversation %d.",
conn->connection_number);
conn->stop = 1;
pthread_kill(conn->thread, SIGUSR1);
}
}
int rco = get_requested_connection_state_to_output();
if (conn->connection_state_to_output != rco) {
conn->connection_state_to_output = rco;
// change happening
if (conn->connection_state_to_output == 0) { // going off
pthread_mutex_lock(&conn->flush_mutex);
conn->flush_requested = 1;
pthread_mutex_unlock(&conn->flush_mutex);
}
}
pthread_mutex_lock(&conn->flush_mutex);
if (conn->flush_requested == 1) {
if (config.output->flush)
config.output->flush();
ab_resync(conn);
conn->first_packet_timestamp = 0;
conn->first_packet_time_to_play = 0;
conn->time_since_play_started = 0;
conn->flush_requested = 0;
}
pthread_mutex_unlock(&conn->flush_mutex);
uint32_t flush_limit = 0;
if (conn->ab_synced) {
do {
curframe = conn->audio_buffer + BUFIDX(conn->ab_read);
if ((conn->ab_read != conn->ab_write) &&
(curframe->ready)) { // it could be synced and empty, under
// exceptional circumstances, with the
// frame unused, thus apparently ready
if (curframe->sequence_number != conn->ab_read) {
// some kind of sync problem has occurred.
if (BUFIDX(curframe->sequence_number) == BUFIDX(conn->ab_read)) {
// it looks like some kind of aliasing has happened
if (seq_order(conn->ab_read, curframe->sequence_number, conn->ab_read)) {
conn->ab_read = curframe->sequence_number;
debug(1, "Aliasing of buffer index -- reset.");
}
} else {
debug(1, "Inconsistent sequence numbers detected");
}
}
if ((conn->flush_rtp_timestamp != 0) &&
(curframe->timestamp <= conn->flush_rtp_timestamp)) {
debug(1, "Dropping flushed packet seqno %u, timestamp %lld", curframe->sequence_number,
curframe->timestamp);
curframe->ready = 0;
flush_limit++;
conn->ab_read = SUCCESSOR(conn->ab_read);
}
if (curframe->timestamp > conn->flush_rtp_timestamp)
conn->flush_rtp_timestamp = 0;
}
} while ((conn->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 = conn->audio_buffer + BUFIDX(conn->ab_read);
if (curframe->ready) {
notified_buffer_empty = 0; // at least one buffer now -- diagnostic only.
if (conn->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 sent some silent frames to the
// DAC
int64_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, conn);
reference_timestamp *= conn->output_sample_ratio;
if (conn->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));
// say we have started playing here
#if defined(HAVE_MPRIS)
if ((conn->play_state != SST_stopped) && (conn->play_state != SST_playing)) {
conn->play_state = SST_playing;
debug(1, "MPRIS Playing");
media_player2_player_set_playback_status(mprisPlayerPlayerSkeleton, "Playing");
}
#endif
if (reference_timestamp) { // if we have a reference time
// debug(1,"First frame seen with timestamp...");
conn->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.
// debug(1, "Output sample ratio is %d", conn->output_sample_ratio);
int64_t delta = (conn->first_packet_timestamp - reference_timestamp) +
config.latency * conn->output_sample_ratio +
(int64_t)(config.audio_backend_latency_offset * config.output_rate);
if (delta >= 0) {
int64_t delta_fp_sec =
(delta << 32) / config.output_rate; // int64_t which is positive
conn->first_packet_time_to_play = reference_timestamp_time + delta_fp_sec;
} else {
int64_t abs_delta = -delta;
int64_t delta_fp_sec =
(abs_delta << 32) / config.output_rate; // int64_t which is positive
conn->first_packet_time_to_play = reference_timestamp_time - delta_fp_sec;
}
if (local_time_now >= conn->first_packet_time_to_play) {
debug(
1,
"First packet is late! It should have played before now. Flushing 0.1 seconds");
player_flush(conn->first_packet_timestamp + 4410 * conn->output_sample_ratio, conn);
}
}
}
if (conn->first_packet_time_to_play != 0) {
// recalculate conn->first_packet_time_to_play -- the latency might change
int64_t delta = (conn->first_packet_timestamp - reference_timestamp) +
config.latency * conn->output_sample_ratio +
(int64_t)(config.audio_backend_latency_offset * config.output_rate);
if (delta >= 0) {
int64_t delta_fp_sec =
(delta << 32) / config.output_rate; // int64_t which is positive
conn->first_packet_time_to_play = reference_timestamp_time + delta_fp_sec;
} else {
int64_t abs_delta = -delta;
int64_t delta_fp_sec =
(abs_delta << 32) / config.output_rate; // int64_t which is positive
conn->first_packet_time_to_play = reference_timestamp_time - delta_fp_sec;
}
// now, the size of the initial silence must be affected by the lead-in time.
// it must be somewhat less than the lead-in time so that dynamic adjustments can be
// made
// to compensate for delays due to paging, etc.
// The suggestion is that it should be at least 100 ms less than the lead-in time.
int64_t max_dac_delay = config.output_rate / 10; // so the lead-in time must be greater
// than this, say 0.2 sec, to allow for
// dynamic adjustment
int64_t filler_size = max_dac_delay; // 0.1 second -- the maximum we'll add to the DAC
if (local_time_now >= conn->first_packet_time_to_play) {
// debug(1,"Gone past starting time");
have_sent_prefiller_silence = 1;
conn->ab_buffering = 0;
// 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-conn->first_packet_time_to_play)*config.output_rate)>>32)+dac_delay,tn,conn->first_packet_time_to_play,dac_delay,seq_diff(ab_read,
// ab_write));
/*
if (config.output->flush)
config.output->flush();
ab_resync(conn);
conn->first_packet_timestamp = 0;
conn->first_packet_time_to_play = 0;
conn->time_since_play_started = 0;
*/
} else {
// do some calculations
int64_t lead_time = conn->first_packet_time_to_play - local_time_now;
int64_t lead_in_time =
(int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)0x100000000);
// debug(1,"Lead time is %llx at fpttp
// %llx.",lead_time,conn->first_packet_time_to_play);
// an audio_backend_silent_lead_in_time of less than zero means start filling ASAP
if ((lead_in_time < 0) || (lead_time <= lead_in_time)) {
// debug(1,"Checking");
if (config.output->delay) {
// conn->first_packet_time_to_play is definitely later than local_time_now
if (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 =
((conn->first_packet_time_to_play - local_time_now) * config.output_rate) >>
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 past 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,conn->first_packet_time_to_play,dac_delay,seq_diff(ab_read,
// ab_write));
if (config.output->flush)
config.output->flush();
ab_resync(conn);
conn->first_packet_timestamp = 0;
conn->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 <= conn->max_frames_per_packet * 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);
conn->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(conn->output_bytes_per_frame * fs);
if (silence == NULL)
debug(1, "Failed to allocate %d byte silence buffer.", fs);