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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 -- 2019
* 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 <string.h>
#include <sys/stat.h>
#include <sys/syslog.h>
#include <sys/types.h>
#include <unistd.h>
#include "config.h"
#ifdef CONFIG_MBEDTLS
#include <mbedtls/aes.h>
#include <mbedtls/havege.h>
#endif
#ifdef CONFIG_POLARSSL
#include <polarssl/aes.h>
#include <polarssl/havege.h>
#endif
#ifdef CONFIG_OPENSSL
#include <openssl/aes.h>
#endif
#ifdef CONFIG_SOXR
#include <soxr.h>
#endif
#ifdef CONFIG_CONVOLUTION
#include <FFTConvolver/convolver.h>
#endif
#ifdef CONFIG_METADATA_HUB
#include "metadata_hub.h"
#endif
#ifdef CONFIG_DACP_CLIENT
#include "dacp.h"
#endif
#include "common.h"
#include "mdns.h"
#include "player.h"
#include "rtp.h"
#include "rtsp.h"
#include "alac.h"
#ifdef CONFIG_APPLE_ALAC
#include "apple_alac.h"
#endif
#include "loudness.h"
#include "activity_monitor.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 2500
// static abuf_t audio_buffer[BUFFER_FRAMES];
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
uint32_t modulo_32_offset(uint32_t from, uint32_t to) {
if (from <= to)
return to - from;
else
return UINT32_MAX - from + to + 1;
}
uint64_t modulo_64_offset(uint64_t from, uint64_t to) {
if (from <= to)
return to - from;
else
return UINT64_MAX - from + to + 1;
}
void do_flush(uint32_t timestamp, rtsp_conn_info *conn);
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].resend_level = 0;
conn->audio_buffer[i].sequence_number = 0;
}
conn->ab_synced = 0;
conn->last_seqno_read = -1;
conn->ab_buffering = 1;
}
// given starting and ending points as unsigned 32-bit integers running modulo 2^32, returns the
// position of x in the interval in *pos
// returns true if x is actually within the buffer
int position_in_modulo_uint32_t_buffer(uint32_t x, uint32_t start, uint32_t end, uint32_t *pos) {
int response = 0; // not in the buffer
if (start <= end) {
if (x < start) {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
} else {
if (pos)
*pos = x - start;
if (x < end)
response = 1;
}
} else if ((x >= start) && (x <= UINT32_MAX)) {
response = 1;
if (pos)
*pos = x - start;
} else {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
if (x < end) {
response = 1;
}
}
return response;
}
// this is used.
static inline seq_t SUCCESSOR(seq_t x) {
uint32_t p = x & 0xffff;
p += 1;
p = p & 0xffff;
return p;
}
// used in seq_diff and seq_order
// 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;
}
void reset_input_flow_metrics(rtsp_conn_info *conn) {
conn->play_number_after_flush = 0;
conn->packet_count_since_flush = 0;
conn->input_frame_rate_starting_point_is_valid = 0;
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
}
void unencrypted_packet_decode(unsigned char *packet, int length, short *dest, int *outsize,
int size_limit, rtsp_conn_info *conn) {
if (conn->stream.type == ast_apple_lossless) {
#ifdef CONFIG_APPLE_ALAC
if (config.use_apple_decoder) {
if (conn->decoder_in_use != 1 << decoder_apple_alac) {
debug(2, "Apple ALAC Decoder used on encrypted audio.");
conn->decoder_in_use = 1 << decoder_apple_alac;
}
apple_alac_decode_frame(packet, length, (unsigned char *)dest, outsize);
*outsize = *outsize * 4; // bring the size to bytes
} else
#endif
{
if (conn->decoder_in_use != 1 << decoder_hammerton) {
debug(2, "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 if (conn->stream.type == ast_uncompressed) {
int length_to_use = length;
if (length_to_use > size_limit) {
warn("unencrypted_packet_decode: uncompressed audio packet too long (size: %d bytes) to "
"process -- truncated",
length);
length_to_use = size_limit;
}
int i;
short *source = (short *)packet;
for (i = 0; i < (length_to_use / 2); i++) {
*dest = ntohs(*source);
dest++;
source++;
}
*outsize = length_to_use;
}
}
int audio_packet_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 maximum_possible_outsize = outsize;
if (conn->stream.encrypted) {
unsigned char iv[16];
int aeslen = len & ~0xf;
memcpy(iv, conn->stream.aesiv, sizeof(iv));
#ifdef CONFIG_MBEDTLS
mbedtls_aes_crypt_cbc(&conn->dctx, MBEDTLS_AES_DECRYPT, aeslen, iv, buf, packet);
#endif
#ifdef CONFIG_POLARSSL
aes_crypt_cbc(&conn->dctx, AES_DECRYPT, aeslen, iv, buf, packet);
#endif
#ifdef CONFIG_OPENSSL
AES_cbc_encrypt(buf, packet, aeslen, &conn->aes, iv, AES_DECRYPT);
#endif
memcpy(packet + aeslen, buf + aeslen, len - aeslen);
unencrypted_packet_decode(packet, len, dest, &outsize, maximum_possible_outsize, conn);
} else {
// not encrypted
unencrypted_packet_decode(buf, len, dest, &outsize, maximum_possible_outsize, conn);
}
if (outsize > maximum_possible_outsize) {
debug(2, "Output from alac_decode larger (%d bytes, not frames) than expected (%d bytes) -- "
"truncated, but buffer overflow possible! Encrypted = %d.",
outsize, maximum_possible_outsize, 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_alac_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;
alac = alac_create(conn->input_bit_depth,
conn->input_num_channels); // no pthread cancellation point in here
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); // no pthread cancellation point in here
#ifdef CONFIG_APPLE_ALAC
apple_alac_init(fmtp); // no pthread cancellation point in here
#endif
return 0;
}
static void terminate_decoders(rtsp_conn_info *conn) {
alac_free(conn->decoder_info);
#ifdef CONFIG_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);
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, uint32_t actual_timestamp, uint8_t *data, int len,
rtsp_conn_info *conn) {
// ignore a request to flush that has been made before the first packet...
if (conn->packet_count == 0) {
debug_mutex_lock(&conn->flush_mutex, 1000, 1);
conn->flush_requested = 0;
conn->flush_rtp_timestamp = 0;
debug_mutex_unlock(&conn->flush_mutex, 3);
}
debug_mutex_lock(&conn->ab_mutex, 30000, 0);
conn->packet_count++;
conn->packet_count_since_flush++;
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 ((conn->flush_rtp_timestamp != 0) && (actual_timestamp != conn->flush_rtp_timestamp) &&
(modulo_32_offset(actual_timestamp, conn->flush_rtp_timestamp) <
conn->input_rate * 10)) { // if it's less than 10 seconds
debug(3, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
seqno, actual_timestamp, conn->flush_rtp_timestamp);
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
} else {
/*
if ((conn->flush_rtp_timestamp != 0) &&
(modulo_32_offset(conn->flush_rtp_timestamp, actual_timestamp) > conn->input_rate / 5)
&&
(modulo_32_offset(conn->flush_rtp_timestamp, actual_timestamp) < conn->input_rate)) {
// between 0.2 and 1 second
debug(2, "Dropping flush request in player_put_packet");
conn->flush_rtp_timestamp = 0;
}
*/
abuf_t *abuf = 0;
if (!conn->ab_synced) {
// if this is the first packet...
debug(3, "syncing to seqno %u.", seqno);
conn->ab_write = seqno;
conn->ab_read = seqno;
conn->ab_synced = 1;
}
// here, we should check for missing frames
int resend_interval = (((250 * 44100) / 352) / 1000); // approximately 250 ms intervals
const int number_of_resend_attempts = 8;
int latency_based_resend_interval =
(conn->latency) / (number_of_resend_attempts * conn->max_frames_per_packet);
if (latency_based_resend_interval > resend_interval)
resend_interval = latency_based_resend_interval;
if (conn->resend_interval != resend_interval) {
debug(2, "Resend interval for latency of %u frames is %d frames.", conn->latency,
resend_interval);
conn->resend_interval = resend_interval;
}
if (conn->ab_write ==
seqno) { // if this is the expected packet (which could be the first packet...)
uint64_t reception_time = get_absolute_time_in_fp();
if (conn->input_frame_rate_starting_point_is_valid == 0) {
if ((conn->packet_count_since_flush >= 500) && (conn->packet_count_since_flush <= 510)) {
conn->frames_inward_measurement_start_time = reception_time;
conn->frames_inward_frames_received_at_measurement_start_time = actual_timestamp;
conn->input_frame_rate_starting_point_is_valid = 1; // valid now
}
}
conn->frames_inward_measurement_time = reception_time;
conn->frames_inward_frames_received_at_measurement_time = actual_timestamp;
abuf = conn->audio_buffer + BUFIDX(seqno);
conn->ab_write = SUCCESSOR(seqno); // move the write pointer to the next free space
} 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->resend_level = 0;
// abuf->timestamp = 0;
abuf->given_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);
/*
if (abuf->ready)
debug(1,"Late apparently duplicate packet received that is %d packets
late.",seq_diff(seqno, conn->ab_write, conn->ab_read));
else
debug(1,"Late packet received that is %d packets late.",seq_diff(seqno,
conn->ab_write, conn->ab_read));
*/
} else { // too late.
// debug(1,"Too late packet received that is %d packets late.",seq_diff(seqno,
// conn->ab_write, conn->ab_read));
conn->too_late_packets++;
}
// pthread_mutex_unlock(&ab_mutex);
if (abuf) {
int datalen = conn->max_frames_per_packet;
if (audio_packet_decode(abuf->data, &datalen, data, len, conn) == 0) {
abuf->ready = 1;
abuf->length = datalen;
// abuf->timestamp = ltimestamp;
abuf->given_timestamp = actual_timestamp;
abuf->sequence_number = seqno;
} else {
debug(1, "Bad audio packet detected and discarded.");
abuf->ready = 0;
abuf->resend_level = 0;
// abuf->timestamp = 0;
abuf->given_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.");
// if it's at the expected time, do a look back for missing packets
// but release the ab_mutex when doing a resend
if (!conn->ab_buffering) {
int j;
for (j = 1; j <= number_of_resend_attempts; j++) {
// check j times, after a short period of has elapsed, assuming 352 frames per packet
// the higher the step_exponent, the less it will try. 1 means it will try very
// hard. 2.0 seems good.
float step_exponent = 2.0;
int back_step = (int)(resend_interval * pow(j, step_exponent));
int k;
for (k = -1; k <= 1; k++) {
if ((back_step + k) <
seq_diff(conn->ab_read, conn->ab_write,
conn->ab_read)) { // if it's within the range of frames in use...
int item_to_check = (conn->ab_write - (back_step + k)) & 0xffff;
seq_t next = item_to_check;
abuf_t *check_buf = conn->audio_buffer + BUFIDX(next);
if ((!check_buf->ready) &&
(check_buf->resend_level <
j)) { // prevent multiple requests from the same level of lookback
check_buf->resend_level = j;
if (config.disable_resend_requests == 0) {
debug_mutex_unlock(&conn->ab_mutex, 3);
rtp_request_resend(next, 1, conn);
conn->resend_requests++;
debug_mutex_lock(&conn->ab_mutex, 20000, 1);
}
}
}
}
}
}
}
}
debug_mutex_unlock(&conn->ab_mutex, 0);
}
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. Interesting part is 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) {
/*
{
static int old_volume = 0;
if (volume != old_volume) {
debug(1,"Volume is now %d.",volume);
old_volume = volume;
}
}
*/
int64_t hyper_sample = sample;
int result = 0;
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: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.
int64_t dither_mask = 0;
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.");
break;
}
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");
break;
}
*outp += result;
}
void buffer_get_frame_cleanup_handler(void *arg) {
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
debug_mutex_unlock(&conn->ab_mutex, 0);
}
// 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 *curframe = NULL;
int notified_buffer_empty = 0; // diagnostic only
debug_mutex_lock(&conn->ab_mutex, 30000, 0);
int wait;
long dac_delay = 0; // long because alsa returns a long
pthread_cleanup_push(buffer_get_frame_cleanup_handler,
(void *)conn); // undo what's been done so far
do {
// get the time
local_time_now = get_absolute_time_in_fp(); // type okay
// debug(3, "buffer_get_frame is iterating");
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
debug_mutex_lock(&conn->flush_mutex, 1000, 1);
conn->flush_requested = 1;
debug_mutex_unlock(&conn->flush_mutex, 3);
}
}
if (config.output->is_running)
if (config.output->is_running() != 0) { // if the back end isn't running for any reason
debug(3, "not running");
debug_mutex_lock(&conn->flush_mutex, 1000, 0);
conn->flush_requested = 1;
debug_mutex_unlock(&conn->flush_mutex, 0);
}
debug_mutex_lock(&conn->flush_mutex, 1000, 0);
if (conn->flush_requested == 1) {
if (config.output->flush)
config.output->flush(); // no cancellation points
ab_resync(conn); // no cancellation points
conn->first_packet_timestamp = 0;
conn->first_packet_time_to_play = 0;
conn->time_since_play_started = 0;
conn->flush_requested = 0;
}
debug_mutex_unlock(&conn->flush_mutex, 0);
if (conn->ab_synced) {
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)
// debug(2,"flush_rtp_timestamp is %" PRIx32 " and curframe->given_timestamp is %" PRIx32
// ".", conn->flush_rtp_timestamp , curframe->given_timestamp);
if ((conn->flush_rtp_timestamp != 0) &&
(curframe->given_timestamp != conn->flush_rtp_timestamp) &&
(modulo_32_offset(curframe->given_timestamp, conn->flush_rtp_timestamp) <
conn->input_rate * 10)) { // if it's less than ten seconds
debug(3, "Dropping flushed packet in buffer_get_frame, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
curframe->sequence_number, curframe->given_timestamp, conn->flush_rtp_timestamp);
curframe->ready = 0;
curframe->resend_level = 0;
curframe = NULL; // this will be returned and will cause the loop to go around again
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
} else if ((conn->flush_rtp_timestamp != 0) &&
(modulo_32_offset(conn->flush_rtp_timestamp, curframe->given_timestamp) >
conn->input_rate / 5) &&
(modulo_32_offset(conn->flush_rtp_timestamp, curframe->given_timestamp) <
conn->input_rate * 10)) {
debug(3, "Dropping flush request in buffer_get_frame");
conn->flush_rtp_timestamp = 0;
}
}
if ((curframe) && (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 = 1; // 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));
if (have_timestamp_timing_information(conn)) { // if we have a reference time
// debug(1,"First frame seen with timestamp...");
conn->first_packet_timestamp =
curframe->given_timestamp; // we will keep buffering until we are
// supposed to start playing this
have_sent_prefiller_silence = 0;
// debug(1, "First packet timestamp is %" PRId64 ".", conn->first_packet_timestamp);
// say we have started playing here
#ifdef CONFIG_METADATA
debug(2, "pffr");
send_ssnc_metadata(
'pffr', NULL, 0,
0); // "first frame received", but don't wait if the queue is locked
#endif
// 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);
// what we are asking for here is "what is the local time at which time the calculated
// frame should be played"
uint64_t should_be_time;
frame_to_local_time(conn->first_packet_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset *
conn->input_rate), // this will go modulo 2^32
&should_be_time,
conn);
conn->first_packet_time_to_play = should_be_time;
if (local_time_now > conn->first_packet_time_to_play) {
uint64_t lateness = local_time_now - conn->first_packet_time_to_play;
lateness = (lateness * 1000000) >> 32; // microseconds
debug(3, "First packet is %" PRIu64 " microseconds late! Flushing 0.5 seconds",
lateness);
do_flush(conn->first_packet_timestamp + 5 * 4410, conn);
}
}
}
if (conn->first_packet_time_to_play != 0) {
// recalculate conn->first_packet_time_to_play -- the latency might change
uint64_t should_be_time;
frame_to_local_time(conn->first_packet_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset *
conn->input_rate), // this should go modulo 2^32
&should_be_time,
conn);
conn->first_packet_time_to_play = should_be_time;
// we want the frames of silence sent at the start to be fairly large in case the output
// device's minimum buffer size is large. But they can't be greater than the silent lead_in time
// which is either the agreed latency or the silent lead-in time specified by the setting
// In fact, if should be some fraction of them, to allow for adjustment.
int64_t max_dac_delay = conn->latency;
if (config.audio_backend_silent_lead_in_time >= 0)
max_dac_delay =
(int64_t)(config.audio_backend_silent_lead_in_time * conn->input_rate);
max_dac_delay = max_dac_delay / 4;
// debug(1,"max_dac_delay is %" PRIu64 " frames.", max_dac_delay);
int64_t filler_size = max_dac_delay;
if (local_time_now > conn->first_packet_time_to_play) {
uint64_t lateness = local_time_now - conn->first_packet_time_to_play;
lateness = (lateness * 1000000) >> 32; // microseconds
debug(3, "Gone past starting time by %" PRIu64 " microseconds.", lateness);
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 {