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shairport-sync/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 <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <sys/types.h>
#include <pthread.h>
#include <openssl/aes.h>
#include <math.h>
#include <sys/stat.h>
#include <sys/signal.h>
#include <assert.h>
#include <fcntl.h>
#include <stdlib.h>
#include <errno.h>
#include "common.h"
#include "player.h"
#include "rtp.h"
#ifdef FANCY_RESAMPLING
#include <samplerate.h>
#endif
#include "alac.h"
// parameters from the source
static unsigned char *aesiv;
static AES_KEY aes;
static int sampling_rate, frame_size;
#define FRAME_BYTES(frame_size) (4*frame_size)
// maximal resampling shift - conservative
#define OUTFRAME_BYTES(frame_size) (4*(frame_size+3))
static pthread_t player_thread;
static int please_stop;
static alac_file *decoder_info;
#ifdef FANCY_RESAMPLING
static int fancy_resampling = 1;
static SRC_STATE *src;
#endif
// debug variables
static int late_packet_message_sent;
static uint64_t packet_count = 0;
// interthread variables
static double volume = 1.0;
static int fix_volume = 0x10000;
static pthread_mutex_t vol_mutex = PTHREAD_MUTEX_INITIALIZER;
// 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
typedef struct audio_buffer_entry { // decoded audio packets
int ready;
uint32_t timestamp;
signed short *data;
} abuf_t;
static abuf_t audio_buffer[BUFFER_FRAMES];
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
// mutex-protected variables
static seq_t ab_read, ab_write;
static int ab_buffering = 1, ab_synced = 0;
static uint32_t first_packet_timestamp = 0;
static int flush_requested = 0;
static uint32_t flush_rtp_timestamp;
// mutexes and condition variables
static pthread_mutex_t ab_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t flush_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_cond_t flowcontrol;
static int64_t first_packet_time_to_play; // nanoseconds
// stats
static uint64_t missing_packets,late_packets,too_late_packets,resend_requests;
static void bf_est_reset(short fill);
static void ab_resync(void) {
int i;
for (i=0; i<BUFFER_FRAMES; i++)
audio_buffer[i].ready = 0;
ab_synced = 0;
ab_buffering = 1;
}
// 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) {
signed short d = b - a;
return d > 0;
}
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 int init_decoder(int32_t fmtp[12]) {
alac_file *alac;
frame_size = fmtp[1]; // stereo samples
sampling_rate = fmtp[11];
int sample_size = fmtp[3];
if (sample_size != 16)
die("only 16-bit samples supported!");
alac = alac_create(sample_size, 2);
if (!alac)
return 1;
decoder_info = alac;
alac->setinfo_max_samples_per_frame = frame_size;
alac->setinfo_7a = fmtp[2];
alac->setinfo_sample_size = sample_size;
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);
return 0;
}
static void free_decoder(void) {
alac_free(decoder_info);
}
#ifdef FANCY_RESAMPLING
static int init_src(void) {
int err;
if (fancy_resampling)
src = src_new(SRC_SINC_MEDIUM_QUALITY, 2, &err);
else
src = NULL;
return err;
}
static void free_src(void) {
src_delete(src);
src = NULL;
}
#endif
static void init_buffer(void) {
int i;
for (i=0; i<BUFFER_FRAMES; i++)
audio_buffer[i].data = malloc(OUTFRAME_BYTES(frame_size));
ab_resync();
}
static void free_buffer(void) {
int i;
for (i=0; i<BUFFER_FRAMES; i++)
free(audio_buffer[i].data);
}
void player_put_packet(seq_t seqno,uint32_t timestamp, uint8_t *data, int len) {
packet_count++;
//if (packet_count<10)
// debug(1,"Packet %llu received.",packet_count);
abuf_t *abuf = 0;
int16_t buf_fill;
pthread_mutex_lock(&ab_mutex);
if (!ab_synced) {
debug(2, "syncing to first seqno %04X.", seqno);
ab_write = seqno-1;
ab_read = seqno;
ab_synced = 1;
}
if (seq_diff(ab_write, seqno) == 1) { // expected packet
abuf = audio_buffer + BUFIDX(seqno);
ab_write = seqno;
} else if (seq_order(ab_write, seqno)) { // newer than expected
rtp_request_resend(ab_write+1, seqno-1);
resend_requests++;
abuf = audio_buffer + BUFIDX(seqno);
ab_write = 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, "late packet %04X (%04X:%04X).", seqno, ab_read, ab_write);
late_packet_message_sent=1;
}
}
buf_fill = seq_diff(ab_read, ab_write);
// pthread_mutex_unlock(&ab_mutex);
if (abuf) {
alac_decode(abuf->data, data, len);
abuf->ready = 1;
abuf->timestamp = timestamp;
}
// pthread_mutex_lock(&ab_mutex);
int rc = pthread_cond_signal(&flowcontrol);
if (rc)
debug(1,"Error signalling flowcontrol.");
pthread_mutex_unlock(&ab_mutex);
}
static short lcg_rand(void) {
static unsigned long lcg_prev = 12345;
lcg_prev = lcg_prev * 69069 + 3;
return lcg_prev & 0xffff;
}
static inline short dithered_vol(short sample) {
static short rand_a, rand_b;
long out;
out = (long)sample * fix_volume;
if (fix_volume < 0x10000) {
rand_b = rand_a;
rand_a = lcg_rand();
out += rand_a;
out -= rand_b;
}
return out>>16;
}
// 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;
seq_t read, next;
abuf_t *abuf = 0;
int i;
abuf_t *curframe;
pthread_mutex_lock(&ab_mutex);
int wait;
int32_t dac_delay = 0;
do {
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;
flush_requested=0;
}
pthread_mutex_unlock(&flush_mutex);
curframe = audio_buffer + BUFIDX(ab_read);
if (config.output->delay) {
dac_delay = config.output->delay();
if (dac_delay==-1) {
debug(1,"Error getting dac_delay at start of loop.");
dac_delay=0;
}
}
if (curframe->ready) {
if ((flush_rtp_timestamp) && (flush_rtp_timestamp>=curframe->timestamp)) {
// debug(1,"Dropping flushed packet %u.",curframe->timestamp);
curframe->ready=0;
ab_read++;
} else 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;
get_reference_timestamp_stuff(&reference_timestamp,&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, see if we should start playing. We need to know when to allow the packets to be sent to the player
// 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.
// it might not be the timestamp of our first packet, however, so we might have to do some calculations.
struct timespec tn;
clock_gettime(CLOCK_MONOTONIC,&tn);
local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000;
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)<<32)/44100; // using the latency requested...
if (local_time_now>=first_packet_time_to_play)
debug(1,"First packet is late! It should have played before now...");
}
}
if (first_packet_time_to_play!=0) {
uint32_t filler_size = frame_size;
if (config.output->delay) {
dac_delay = config.output->delay();
if (dac_delay==-1) {
debug(1,"Delay error when setting a filler size.");
dac_delay=0;
}
if (dac_delay==0)
filler_size = 11025; // 0.25 second
}
struct timespec time_now;
clock_gettime(CLOCK_MONOTONIC,&time_now);
uint64_t tn = ((uint64_t)time_now.tv_sec<<32)+((uint64_t)time_now.tv_nsec<<32)/1000000000;
if (tn>=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;
} else {
uint64_t gross_frame_gap = ((first_packet_time_to_play-tn)*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 ((exact_frame_gap<=filler_size) || (exact_frame_gap<=frame_size*2)) {
fs=exact_frame_gap;
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);
}
}
}
}
}
// allow to DAC to have a quarter of a second -- seems necessary for VMWare Fusion and a WiFi connection.
wait = (ab_buffering || (dac_delay>=4410) || (!ab_synced)) && (!please_stop);
// wait = (ab_buffering || (seq_diff(ab_read, ab_write) < (config.latency-22000)/(352)) || (!ab_synced)) && (!please_stop);
if (wait) {
struct timespec to;
clock_gettime(CLOCK_MONOTONIC, &to);
// Watch out: should be 1^9/44100, but this causes integer overflow
uint64_t calc = (uint64_t)(4*352*1000000)/(44*3); //four thirds of a packet time
uint32_t calcs = calc&0xFFFFFFFF;
to.tv_nsec+=calcs;
if (to.tv_nsec>1000000000) {
to.tv_nsec-=1000000000;
to.tv_sec+=1;
}
pthread_cond_timedwait(&flowcontrol,&ab_mutex,&to);
}
} while (wait);
if (!please_stop) {
// buf_fill = seq_diff(ab_read, ab_write);
// if (buf_fill < 1 || !ab_synced) {
// if (buf_fill < 1)
if (dac_delay==0) {// we just got underrun
debug(1,"Underrun!");
ab_resync(); // starting over
}
}
if (please_stop) {
pthread_mutex_unlock(&ab_mutex);
debug(1,"Exiting from buffer_get_frame.");
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++;
// 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 < (seq_diff(ab_read,ab_write) / 2); i = (i * 2)) {
next = ab_read + i;
abuf = audio_buffer + BUFIDX(next);
if (!abuf->ready) {
rtp_request_resend(next, next);
resend_requests++;
}
}
}
if (!curframe->ready) {
// debug(1, "missing frame %04X. Supplying a silent frame.", read);
missing_packets++;
memset(curframe->data, 0, FRAME_BYTES(frame_size));
}
curframe->ready = 0;
pthread_mutex_unlock(&ab_mutex);
return curframe;
}
static int stuff_buffer(short *inptr, short *outptr, int32_t stuff) {
int i;
int stuffsamp = frame_size;
if (stuff)
stuffsamp = rand() % (frame_size - 1);
pthread_mutex_lock(&vol_mutex);
for (i=0; i<stuffsamp; i++) { // the whole frame, if no stuffing
*outptr++ = dithered_vol(*inptr++);
*outptr++ = dithered_vol(*inptr++);
};
if (stuff) {
if (stuff==1) {
debug(3, "+++++++++");
// interpolate one sample
*outptr++ = dithered_vol(((long)inptr[-2] + (long)inptr[0]) >> 1);
*outptr++ = dithered_vol(((long)inptr[-1] + (long)inptr[1]) >> 1);
} else if (stuff==-1) {
debug(3, "---------");
inptr++;
inptr++;
}
for (i=stuffsamp; i<frame_size + stuff; i++) {
*outptr++ = dithered_vol(*inptr++);
*outptr++ = dithered_vol(*inptr++);
}
}
pthread_mutex_unlock(&vol_mutex);
return frame_size + stuff;
}
static void *player_thread_func(void *arg) {
#define averaging_interval 71
#define trend_interval 44100
int64_t corrections[trend_interval];
int64_t sum_of_corrections, sum_of_insertions_and_deletions;
double moving_average_correction, moving_average_insertions_and_deletions;
int oldest_correction, newest_correction, number_of_corrections;
int64_t delays[averaging_interval];
int64_t sum_of_delays,moving_average_delay;
int oldest_delay,newest_delay,number_of_delays;
int64_t additions,deletions = 0;
additions = deletions = 0;
int64_t frames = 0;
int play_samples;
int64_t initial_latency = 0;
int64_t previous_latency,current_latency,change_in_latency;
int64_t current_delay;
int play_number = 0;
uint64_t accumulated_buffers_in_use = 0;
int64_t accumulated_delay = 0;
int64_t accumulated_da_delay = 0;
const int print_interval = 71;
// I think it's useful to keep this prime to prevent it from falling into a pattern with some other process.
struct drand48_data randBuffer;
srand48_r(time(NULL), &randBuffer);
signed short *inbuf, *outbuf, *silence;
outbuf = malloc(OUTFRAME_BYTES(frame_size));
silence = malloc(OUTFRAME_BYTES(frame_size));
memset(silence, 0, OUTFRAME_BYTES(frame_size));
#ifdef FANCY_RESAMPLING
float *frame, *outframe;
SRC_DATA srcdat;
if (fancy_resampling) {
frame = malloc(frame_size*2*sizeof(float));
outframe = malloc(2*frame_size*2*sizeof(float));
srcdat.data_in = frame;
srcdat.data_out = outframe;
srcdat.input_frames = FRAME_BYTES(frame_size);
srcdat.output_frames = 2*FRAME_BYTES(frame_size);
srcdat.src_ratio = 1.0;
srcdat.end_of_input = 0;
}
#endif
change_in_latency = 0;
late_packet_message_sent=0;
oldest_delay=newest_delay=number_of_delays=0;
sum_of_delays=0;
oldest_correction=newest_correction=number_of_corrections=0;
sum_of_corrections=0;
sum_of_insertions_and_deletions=0;
missing_packets=late_packets=too_late_packets=resend_requests=0;
flush_rtp_timestamp=0;
while (!please_stop) {
abuf_t *inframe = buffer_get_frame();
if (inframe) {
inbuf = inframe->data;
if (inbuf) {
#ifdef FANCY_RESAMPLING
if (fancy_resampling) {
int i;
pthread_mutex_lock(&vol_mutex);
for (i=0; i<2*FRAME_BYTES(frame_size); i++) {
frame[i] = (float)inbuf[i] / 32768.0;
frame[i] *= volume;
}
pthread_mutex_unlock(&vol_mutex);
srcdat.src_ratio = bf_playback_rate;
src_process(src, &srcdat);
assert(srcdat.input_frames_used == FRAME_BYTES(frame_size));
src_float_to_short_array(outframe, outbuf, FRAME_BYTES(frame_size)*2);
play_samples = srcdat.output_frames_gen;
} else
#endif
// we need to see whether the player is a bit slow or a bit fast, so we track the overall latency
struct timespec tn;
clock_gettime(CLOCK_MONOTONIC,&tn);
uint64_t local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000;
double x;
drand48_r(&randBuffer, &x);
play_number++;
int32_t abs_change_in_latency = change_in_latency;
if (abs_change_in_latency<1)
abs_change_in_latency=-abs_change_in_latency;
int32_t amount_to_stuff=0;
if (x<=(double)1.0*abs_change_in_latency/print_interval) {
amount_to_stuff= -abs_change_in_latency/change_in_latency;
}
// prevent dithering by +/- 1...
if (abs_change_in_latency<20)
amount_to_stuff=0;
if (number_of_corrections==trend_interval) { // the array of corrections is full
sum_of_corrections-=corrections[oldest_correction];
int64_t oldest_correction_value = corrections[oldest_correction];
// remove the correction from the tally of insertions and deletions
if (oldest_correction_value>=0)
sum_of_insertions_and_deletions-=oldest_correction_value;
else
sum_of_insertions_and_deletions+=oldest_correction_value;
oldest_correction=(oldest_correction+1)%trend_interval;
number_of_corrections--;
}
corrections[newest_correction]=amount_to_stuff;
sum_of_corrections+=amount_to_stuff;
// add the correction to the tally of insertions and deletions
if (amount_to_stuff>=0)
sum_of_insertions_and_deletions+=amount_to_stuff;
else
sum_of_insertions_and_deletions-=amount_to_stuff;
newest_correction=(newest_correction+1)%trend_interval;
number_of_corrections++;
moving_average_correction = (1.0*sum_of_corrections)/number_of_corrections;
moving_average_insertions_and_deletions = (1.0*sum_of_insertions_and_deletions)/number_of_corrections;
play_samples = stuff_buffer(inbuf, outbuf,amount_to_stuff);
if (amount_to_stuff > 0)
additions += amount_to_stuff;
else
deletions -= amount_to_stuff;
frames += play_samples;
if (config.output->delay) {
current_delay = config.output->delay();
if (current_delay==-1)
debug(1,"Delay error when checking running latency.");
current_delay=0;
}
config.output->play(outbuf, play_samples);
// now see if we can calculate how early or late the first frame would be
uint32_t reference_timestamp;
uint64_t reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp,&reference_timestamp_time);
int64_t rt,nt;
rt = reference_timestamp;
nt = inframe->timestamp;
int64_t td_in_frames;
int64_t td = local_time_now-reference_timestamp_time;
// debug(1,"td is %lld.",td);
if (td>=0) {
td_in_frames = (td*44100)>>32;
} else {
td_in_frames = -((-td*44100)>>32);
}
// this is the actual delay, which will fluctuate a good bit about a potentially rising or falling trend.
int64_t delay = td_in_frames+rt-(nt-current_delay);
// if ((play_number<(10)) || (play_number%500==0))
// if ((play_number<(10)))
// debug(1,"Latency error for packet %d: %d frames.",play_number,delay-config.latency);
if (number_of_delays==averaging_interval) { // the array of delays is full
sum_of_delays-=delays[oldest_delay];
oldest_delay=(oldest_delay+1)%averaging_interval;
number_of_delays--;
}
delays[newest_delay]=delay;
sum_of_delays+=delay;
newest_delay=(newest_delay+1)%averaging_interval;
number_of_delays++;
moving_average_delay = sum_of_delays/number_of_delays; // this doesn't seem to be right
change_in_latency=moving_average_delay-config.latency;
accumulated_da_delay+=current_delay;
accumulated_delay+=delay;
accumulated_buffers_in_use += seq_diff(ab_read, ab_write);
if (play_number%print_interval==0) {
current_latency=accumulated_delay/print_interval;
if ((play_number/print_interval)%100==0)
{ // only print every hundredth one, in verbose mode
//debug(1,"Valid frames: %lld; overall frames added/subtracted %lld; frames added + frames deleted %lld; average D/A delay, average latency (frames): %llu, %llu; average buffers in use: %llu, moving average delay (number of delays): %llu (%lu).",
// frames-(additions-deletions), additions-deletions, additions+deletions, accumulated_da_delay/print_interval,current_latency,accumulated_buffers_in_use/print_interval,moving_average_delay,number_of_delays);
//debug(1,"Frames %lld, correction %lld, mods %lld, dac_buffer %llu, latency %llu, missing_packets %llu, late_packets %llu, too_late_packets %llu resend_requests %llu.",
//frames-(additions-deletions), additions-deletions, additions+deletions,accumulated_da_delay/print_interval,moving_average_delay,missing_packets,late_packets,too_late_packets,resend_requests);
debug(1,"Drift: %.1f (ppm); Corrections: %.1f (ppm); missing_packets %llu; late_packets %llu; too_late_packets %llu; resend_requests %llu.", -moving_average_correction*1000000/352, moving_average_insertions_and_deletions*1000000/352,missing_packets,late_packets,too_late_packets,resend_requests);
}
if (previous_latency==0)
previous_latency=current_latency;
else {
previous_latency=current_latency;
}
accumulated_delay=0;
accumulated_da_delay=0;
accumulated_buffers_in_use=0;
}
}
}
}
return 0;
}
// takes the volume as specified by the airplay protocol
void player_volume(double f) {
// The volume ranges -144.0 (mute) or -30 -- 0. See http://git.zx2c4.com/Airtunes2/about/#setting-volume
// By examination, the -30 -- 0 range is linear on the slider; i.e. the slider is calibrated in 30 equal increments
// So, we will pass this on without any weighting if we have a hardware mixer, as we expect the mixer to be calibrated in dB.
// Here, we ask for an attenuation we will apply in software. The dB range of a value from 1 to 65536 is about 48.1 dB (log10 of 65536 is 4.8164).
// Thus, we ask out vol2attn function for an appropriate dB between -48.1 and 0 dB and translate it back to a number.
double linear_volume = pow(10,vol2attn(f,0,-4810)/1000);
if(f == -144.0)
linear_volume = 0.0;
if (config.output->volume) {
config.output->volume(f);
linear_volume=1.0; // no attenuation needed
}
pthread_mutex_lock(&vol_mutex);
volume = linear_volume;
fix_volume = 65536.0 * volume;
pthread_mutex_unlock(&vol_mutex);
}
void player_flush(uint32_t timestamp) {
pthread_mutex_lock(&flush_mutex);
flush_requested=1;
flush_rtp_timestamp=timestamp; // flush all packets up to (and including?) this
pthread_mutex_unlock(&flush_mutex);
}
/*
if (config.output->flush)
config.output->flush();
pthread_mutex_lock(&ab_mutex);
ab_resync();
pthread_mutex_unlock(&ab_mutex);
*/
int player_play(stream_cfg *stream) {
debug(1,"player_play called...");
packet_count = 0;
if (config.buffer_start_fill > BUFFER_FRAMES)
die("specified buffer starting fill %d > buffer size %d",
config.buffer_start_fill, BUFFER_FRAMES);
AES_set_decrypt_key(stream->aeskey, 128, &aes);
aesiv = stream->aesiv;
init_decoder(stream->fmtp);
// must be after decoder init
init_buffer();
#ifdef FANCY_RESAMPLING
init_src();
#endif
please_stop = 0;
if (config.cmd_start && !fork()) {
if (system(config.cmd_start))
warn("exec of external start command failed");
exit(0);
}
// set the flowcontrol condition variable to wait on a monotonic clock
pthread_condattr_t attr;
pthread_condattr_init(&attr);
pthread_condattr_setclock( &attr, CLOCK_MONOTONIC);
int rc = pthread_cond_init(&flowcontrol,&attr);
if (rc)
debug(1,"Error initialising condition variable.");
config.output->start(sampling_rate);
pthread_create(&player_thread, NULL, player_thread_func, NULL);
return 0;
}
void player_stop(void) {
debug(1,"player_stop called...");
please_stop = 1;
pthread_cond_signal(&flowcontrol); // tell it to give up
pthread_join(player_thread, NULL);
config.output->stop();
if (config.cmd_stop && !fork()) {
if (system(config.cmd_stop))
warn("exec of external stop command failed");
exit(0);
}
free_buffer();
free_decoder();
#ifdef FANCY_RESAMPLING
free_src();
#endif
int rc = pthread_cond_destroy(&flowcontrol);
if (rc)
debug(1,"Error destroying condition variable.");
debug(1,"player_stop finished...");
}
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