/*

* HairTunes - RAOP packet handler and slave-clocked replay engine

* Copyright (c) James Laird 2011

* 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 <string.h>

#include <sys/socket.h>

#include <sys/select.h>

#include <arpa/inet.h>

#include <netinet/in.h>

#include <pthread.h>

#include <openssl/aes.h>

#include <ao/ao.h>

#include <math.h>

#ifdef FANCY_RESAMPLING

#include <samplerate.h>

#endif

#include <assert.h>

int debug = 0;

#include "alac.h"

// default buffer - about half a second

#define BUFFER_FRAMES 64

#define START_FILL 55

#define MAX_PACKET 2048

typedef unsigned short seq_t;

// global options (constant after init)

unsigned char aeskey[16], aesiv[16];

AES_KEY aes;

char *rtphost = 0;

int dataport = 0, controlport = 0, timingport = 0;

int fmtp[32];

int sampling_rate;

int frame_size;

#define FRAME_BYTES (4*frame_size)

// maximal resampling shift - conservative

#define OUTFRAME_BYTES (4*(frame_size+3))

alac_file *decoder_info;

#ifdef FANCY_RESAMPLING

int fancy_resampling = 1;

SRC_STATE *src;

#endif

void rtp_request_resend(seq_t first, seq_t last);

void init_buffer(void);

void ab_resync(void);

// interthread variables

// stdin->decoder

volatile double volume = 1.0;

volatile long fix_volume = 0x10000;

typedef struct audio_buffer_entry { // decoded audio packets

int ready;

signed short *data;

} abuf_t;

volatile abuf_t audio_buffer[BUFFER_FRAMES];

#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)

// mutex-protected variables

volatile seq_t ab_read, ab_write;

int ab_buffering = 1, ab_synced = 0;

pthread_mutex_t ab_mutex = PTHREAD_MUTEX_INITIALIZER;

pthread_cond_t ab_buffer_ready = PTHREAD_COND_INITIALIZER;

void die(char *why) {

fprintf(stderr, "FATAL: %s\n", why);

exit(1);

}

int hex2bin(unsigned char *buf, char *hex) {

int i, j;

if (strlen(hex) != 0x20)

return 1;

for (i=0; i<0x10; i++) {

if (!sscanf(hex, "%2X", &j))

return 1;

hex += 2;

*buf++ = j;

}

return 0;

}

int init_decoder(void) {

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 = create_alac(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];

allocate_buffers(alac);

return 0;

}

int main(int argc, char **argv) {

char *hexaeskey = 0, *hexaesiv = 0;

char *fmtpstr = 0;

char *arg;

int i;

assert(RAND_MAX >= 0x10000); // XXX move this to compile time

while (arg = *++argv) {

if (!strcasecmp(arg, "iv")) {

hexaesiv = *++argv;

argc--;

} else

if (!strcasecmp(arg, "key")) {

hexaeskey = *++argv;

argc--;

} else

if (!strcasecmp(arg, "fmtp")) {

fmtpstr = *++argv;

} else

if (!strcasecmp(arg, "cport")) {

controlport = atoi(*++argv);

} else

if (!strcasecmp(arg, "tport")) {

timingport = atoi(*++argv);

} else

if (!strcasecmp(arg, "dport")) {

dataport = atoi(*++argv);

} else

if (!strcasecmp(arg, "host")) {

rtphost = *++argv;

}

#ifdef FANCY_RESAMPLING

else

if (!strcasecmp(arg, "resamp")) {

fancy_resampling = atoi(*++argv);

}

#endif

}

if (!hexaeskey || !hexaesiv)

die("Must supply AES key and IV!");

if (hex2bin(aesiv, hexaesiv))

die("can't understand IV");

if (hex2bin(aeskey, hexaeskey))

die("can't understand key");

AES_set_decrypt_key(aeskey, 128, &aes);

memset(fmtp, 0, sizeof(fmtp));

i = 0;

while (arg = strsep(&fmtpstr, " \t"))

fmtp[i++] = atoi(arg);

init_decoder();

init_buffer();

init_rtp(); // open a UDP listen port and start a listener; decode into ring buffer

fflush(stdout);

init_output(); // resample and output from ring buffer

char line[128];

int in_line = 0;

int n;

double f;

while (fgets(line + in_line, sizeof(line) - in_line, stdin)) {

n = strlen(line);

if (line[n-1] != '\n') {

in_line = strlen(line) - 1;

if (n == sizeof(line)-1)

in_line = 0;

continue;

}

if (sscanf(line, "vol: %lf\n", &f)) {

assert(f<=0);

if (debug)

fprintf(stderr, "VOL: %lf\n", f);

volume = pow(10.0,0.1*f);

fix_volume = 65536.0 * volume;

continue;

}

if (!strcmp(line, "exit\n")) {

exit(0);

}

if (!strcmp(line, "flush\n")) {

pthread_mutex_lock(&ab_mutex);

ab_resync();

pthread_mutex_unlock(&ab_mutex);

if (debug)

fprintf(stderr, "FLUSH\n");

}

}

fprintf(stderr, "bye!\n");

fflush(stderr);

}

void init_buffer(void) {

int i;

for (i=0; i<BUFFER_FRAMES; i++)

audio_buffer[i].data = malloc(OUTFRAME_BYTES);

ab_resync();

}

void ab_resync(void) {

int i;

for (i=0; i<BUFFER_FRAMES; i++)

audio_buffer[i].ready = 0;

ab_synced = 0;

}

// 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;

}

void alac_decode(short *dest, char *buf, int len) {

char packet[MAX_PACKET];

assert(len<=MAX_PACKET);

char iv[16];

int i;

memcpy(iv, aesiv, sizeof(iv));

for (i=0; i+16<=len; i += 16)

AES_cbc_encrypt(buf+i, packet+i, 0x10, &aes, iv, AES_DECRYPT);

if (len & 0xf)

memcpy(packet+i, buf+i, len & 0xf);

int outsize;

decode_frame(decoder_info, packet, dest, &outsize);

assert(outsize == FRAME_BYTES);

}

void buffer_put_packet(seq_t seqno, char *data, int len) {

volatile abuf_t *abuf = 0;

short read;

short buf_fill;

pthread_mutex_lock(&ab_mutex);

if (!ab_synced) {

ab_write = seqno;

ab_read = seqno-1;

ab_synced = 1;

}

if (seqno == ab_write+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, seqno-1);

abuf = audio_buffer + BUFIDX(seqno);

ab_write = seqno;

} else if (seq_order(ab_read, seqno)) { // late but not yet played

abuf = audio_buffer + BUFIDX(seqno);

} else { // too late.

fprintf(stderr, "\nlate packet %04X (%04X:%04X)\n", seqno, ab_read, ab_write);

}

buf_fill = ab_write - ab_read;

pthread_mutex_unlock(&ab_mutex);

if (abuf) {

alac_decode(abuf->data, data, len);

abuf->ready = 1;

}

if (ab_buffering && buf_fill >= START_FILL)

pthread_cond_signal(&ab_buffer_ready);

if (!ab_buffering) {

// check if the t+10th packet has arrived... last-chance resend

read = ab_read + 10;

abuf = audio_buffer + BUFIDX(read);

if (!abuf->ready)

rtp_request_resend(read, read);

}

}

static int rtp_sockets[2]; // data, control

struct sockaddr_in rtp_client;

void *rtp_thread_func(void *arg) {

socklen_t si_len = sizeof(rtp_client);

char packet[MAX_PACKET];

char *pktp;

seq_t seqno;

ssize_t plen;

int sock = rtp_sockets[0], csock = rtp_sockets[1];

int readsock;

char type;

fd_set fds;

FD_ZERO(&fds);

FD_SET(sock, &fds);

FD_SET(csock, &fds);

while (select(csock>sock ? csock+1 : sock+1, &fds, 0, 0, 0)!=-1) {

if (FD_ISSET(sock, &fds)) {

readsock = sock;

} else {

readsock = csock;

}

FD_SET(sock, &fds);

FD_SET(csock, &fds);

plen = recvfrom(readsock, packet, sizeof(packet), 0, (struct sockaddr*)&rtp_client, &si_len);

if (plen < 0)

continue;

assert(plen<=MAX_PACKET);

type = packet[1] & ~0x80;

if (type == 0x60 || type == 0x56) { // audio data / resend

pktp = packet;

if (type==0x56) {

pktp += 4;

plen -= 4;

}

seqno = ntohs(*(unsigned short *)(pktp+2));

buffer_put_packet(seqno, pktp+12, plen-12);

}

}

}

void rtp_request_resend(seq_t first, seq_t last) {

if (seq_order(last, first))

return;

fprintf(stderr, "requesting resend on %d packets (port %d)\n", last-first+1, controlport);

char req[8]; // *not* a standard RTCP NACK

req[0] = 0x80;

req[1] = 0x55|0x80; // Apple 'resend'

*(unsigned short *)(req+2) = htons(1); // our seqnum

*(unsigned short *)(req+4) = htons(first); // missed seqnum

*(unsigned short *)(req+6) = htons(last-first+1); // count

rtp_client.sin_port = htons(controlport);

sendto(rtp_sockets[1], req, sizeof(req), 0, (struct sockaddr *)&rtp_client, sizeof(struct sockaddr_in));

}

int init_rtp(void) {

struct sockaddr_in si;

int sock, csock; // data and control (we treat the streams the same here)

sock = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);

if (sock==-1)

die("Can't create socket!");

memset(&si, 0, sizeof(si));

si.sin_family = AF_INET;

si.sin_addr.s_addr = htonl(INADDR_ANY);

unsigned short port = 6000 - 3;

do {

port += 3;

si.sin_port = htons(port);

} while (bind(sock, (struct sockaddr*)&si, sizeof(si))==-1);

csock = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);

if (csock==-1)

die("Can't create socket!");

si.sin_port = htons(port + 1);

if (bind(csock, (struct sockaddr*)&si, sizeof(si))==-1)

die("can't bind control socket");

printf("port: %d\n", port); // let our handler know where we end up listening

printf("cport: %d\n", port+1);

pthread_t rtp_thread;

rtp_sockets[0] = sock;

rtp_sockets[1] = csock;

pthread_create(&rtp_thread, NULL, rtp_thread_func, (void *)rtp_sockets);

return port;

}

static inline short dithered_vol(short sample) {

static short rand_a, rand_b;

long out;

rand_b = rand_a;

rand_a = rand() & 0xffff;

out = (long)sample * fix_volume;

if (fix_volume < 0x10000) {

out += rand_a;

out -= rand_b;

}

return out>>16;

}

typedef struct {

double hist[2];

double a[2];

double b[3];

} biquad_t;

static void biquad_init(biquad_t *bq, double a[], double b[]) {

bq->hist[0] = bq->hist[1] = 0.0;

memcpy(bq->a, a, 2*sizeof(double));

memcpy(bq->b, b, 3*sizeof(double));

}

static void biquad_lpf(biquad_t *bq, double freq, double Q) {

double w0 = 2*M_PI*freq/((float)sampling_rate/(float)frame_size);

double alpha = sin(w0)/(2.0*Q);

double a_0 = 1.0 + alpha;

double b[3], a[2];

b[0] = (1.0-cos(w0))/(2.0*a_0);

b[1] = (1.0-cos(w0))/a_0;

b[2] = b[0];

a[0] = -2.0*cos(w0)/a_0;

a[1] = (1-alpha)/a_0;

biquad_init(bq, a, b);

}

static double biquad_filt(biquad_t *bq, double in) {

double w = in - bq->a[0]*bq->hist[0] - bq->a[1]*bq->hist[1];

double out = bq->b[1]*bq->hist[0] + bq->b[2]*bq->hist[1] + bq->b[0]*w;

bq->hist[1] = bq->hist[0];

bq->hist[0] = w;

}

double bf_playback_rate = 1.0;

static double bf_est_drift = 0.0; // local clock is slower by

static biquad_t bf_drift_lpf;

static double bf_est_err = 0.0, bf_last_err;

static biquad_t bf_err_lpf, bf_err_deriv_lpf;

static double desired_fill;

static int fill_count;

void bf_est_reset(short fill) {

biquad_lpf(&bf_drift_lpf, 1.0/180.0, 0.3);

biquad_lpf(&bf_err_lpf, 1.0/10.0, 0.25);

biquad_lpf(&bf_err_deriv_lpf, 1.0/2.0, 0.2);

fill_count = 0;

bf_playback_rate = 1.0;

bf_est_err = bf_last_err = 0;

desired_fill = fill_count = 0;

}

void bf_est_update(short fill) {

if (fill_count < 1000) {

desired_fill += (double)fill/1000.0;

fill_count++;

return;

}

#define CONTROL_A (1e-4)

#define CONTROL_B (1e-1)

double buf_delta = fill - desired_fill;

bf_est_err = biquad_filt(&bf_err_lpf, buf_delta);

double err_deriv = biquad_filt(&bf_err_deriv_lpf, bf_est_err - bf_last_err);

bf_est_drift = biquad_filt(&bf_drift_lpf, CONTROL_B*(bf_est_err*CONTROL_A + err_deriv) + bf_est_drift);

if (debug)

fprintf(stderr, "bf %d err %f drift %f desiring %f ed %f estd %f\r", fill, bf_est_err, bf_est_drift, desired_fill, err_deriv, err_deriv + CONTROL_A*bf_est_err);

bf_playback_rate = 1.0 + CONTROL_A*bf_est_err + bf_est_drift;

bf_last_err = bf_est_err;

}

// get the next frame, when available. return 0 if underrun/stream reset.

short *buffer_get_frame(void) {

short buf_fill;

seq_t read;

pthread_mutex_lock(&ab_mutex);

buf_fill = ab_write - ab_read;

if (buf_fill < 1 || !ab_synced) { // init or underrun. stop and wait

if (ab_synced)

fprintf(stderr, "\nunderrun.\n");

ab_buffering = 1;

pthread_cond_wait(&ab_buffer_ready, &ab_mutex);

ab_read++;

buf_fill = ab_write - ab_read;

pthread_mutex_unlock(&ab_mutex);

bf_est_reset(buf_fill);

return 0;

}

if (buf_fill >= BUFFER_FRAMES) { // overrunning! uh-oh. restart at a sane distance

fprintf(stderr, "\noverrun.\n");

ab_read = ab_write - START_FILL;

}

read = ab_read;

ab_read++;

pthread_mutex_unlock(&ab_mutex);

buf_fill = ab_write - ab_read;

bf_est_update(buf_fill);

volatile abuf_t *curframe = audio_buffer + BUFIDX(read);

if (!curframe->ready) {

fprintf(stderr, "\nmissing frame.\n");

memset(curframe->data, 0, FRAME_BYTES);

}

curframe->ready = 0;

return curframe->data;

}

int stuff_buffer(double playback_rate, short *inptr, short *outptr) {

int i;

int stuffsamp = frame_size;

int stuff = 0;

double p_stuff;

p_stuff = 1.0 - pow(1.0 - fabs(playback_rate-1.0), frame_size);

if ((float)rand()/((float)RAND_MAX) < p_stuff) {

stuff = playback_rate > 1.0 ? -1 : 1;

stuffsamp = rand() % (frame_size - 1);

}

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) {

if (debug)

fprintf(stderr, "+++++++++\n");

// 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) {

if (debug)

fprintf(stderr, "---------\n");

inptr++;

inptr++;

}

for (i=stuffsamp; i<frame_size + stuff; i++) {

*outptr++ = dithered_vol(*inptr++);

*outptr++ = dithered_vol(*inptr++);

}

}

}

void *audio_thread_func(void *arg) {

ao_device *dev = arg;

int i, play_samples;

signed short buf_fill;

signed short *inbuf, *outbuf;

outbuf = malloc(OUTFRAME_BYTES);

#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;

srcdat.output_frames = 2*FRAME_BYTES;

srcdat.src_ratio = 1.0;

srcdat.end_of_input = 0;

}

#endif

while (1) {

do {

inbuf = buffer_get_frame();

} while (!inbuf);

#ifdef FANCY_RESAMPLING

if (fancy_resampling) {

for (i=0; i<2*FRAME_BYTES; i++) {

frame[i] = (float)inbuf[i] / 32768.0;

frame[i] *= volume;

}

srcdat.src_ratio = bf_playback_rate;

src_process(src, &srcdat);

assert(srcdat.input_frames_used == FRAME_BYTES);

src_float_to_short_array(outframe, outbuf, FRAME_BYTES*2);

play_samples = srcdat.output_frames_gen;

} else

#endif

play_samples = stuff_buffer(bf_playback_rate, inbuf, outbuf);

ao_play(dev, (char *)outbuf, play_samples*4);

}

}

int init_output(void) {

ao_initialize();

int driver = ao_default_driver_id();

ao_sample_format fmt;

fmt.bits = 16;

fmt.rate = sampling_rate;

fmt.channels = 2;

fmt.byte_format = AO_FMT_LITTLE;

fmt.matrix = 0;

ao_device *dev = ao_open_live(driver, &fmt, 0);

int err;

#ifdef FANCY_RESAMPLING

if (fancy_resampling)

src = src_new(SRC_SINC_MEDIUM_QUALITY, 2, &err);

else

src = 0;

#endif

pthread_t audio_thread;

pthread_create(&audio_thread, NULL, audio_thread_func, dev);

}