nils/shairport-sync
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

New backend for converting 32-bit "native" samples to output format ready for testing

Browse Source
This commit is contained in:
Mike Brady
2016-09-08 21:53:19 +01:00
parent 8d008fa2ec
commit 3469a2ab02
2 changed files with 89 additions and 103 deletions
Download Patch File Download Diff File Expand all files Collapse all files
player.c
+86 -102
View File
@@ -74,7 +74,7 @@ static unsigned char *aesiv;
#ifdef HAVE_LIBSSL
static AES_KEY aes;
#endif
static int input_sample_rate, input_bit_depth, input_num_channels, max_frames_per_packet;
static int input_rate, input_bit_depth, input_num_channels, max_frames_per_packet;
static uint32_t timestamp_epoch, last_timestamp, maximum_timestamp_interval;// timestamp_epoch of zero means not initialised, could start at 2 or 1.
@@ -83,7 +83,7 @@ static uint32_t timestamp_epoch, last_timestamp, maximum_timestamp_interval;// t
static int input_bytes_per_frame = 4;
static int output_bytes_per_frame;
static int output_sample_ratio;
static int output_sample_rate;
static int output_rate;
static int64_t previous_random_number = 0;
@@ -421,7 +421,7 @@ static int init_decoder(int32_t fmtp[12]) {
max_frames_per_packet = fmtp[1]; // number of audio frames per packet.
input_sample_rate = fmtp[11];
input_rate = fmtp[11];
input_num_channels = fmtp[7];
input_bit_depth = fmtp[3];
@@ -583,7 +583,7 @@ int32_t rand_in_range(int32_t exclusive_range_limit) {
return sp >> 32;
}
inline void process_sample(int32_t sample,char**outp,enum sps_format_t format,int volume,int dither) {
static inline void process_sample(int32_t sample,char**outp,enum sps_format_t format,int volume,int dither) {
int64_t hyper_sample = sample;
int result;
// first, modify volume, if necessary
@@ -858,14 +858,14 @@ static abuf_t *buffer_get_frame(void) {
debug(1,"Output sample ratio is %d",output_sample_ratio);
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency*output_sample_ratio+config.audio_backend_latency_offset*output_sample_rate;
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency*output_sample_ratio+config.audio_backend_latency_offset*output_rate;
if (delta>=0) {
int64_t delta_fp_sec = (delta << 32) / output_sample_rate; // int64_t which is positive
int64_t delta_fp_sec = (delta << 32) / output_rate; // int64_t which is positive
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) / output_sample_rate; // int64_t which is positive
int64_t delta_fp_sec = (abs_delta << 32) / output_rate; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
@@ -880,25 +880,25 @@ debug(1,"Output sample ratio is %d",output_sample_ratio);
if (first_packet_time_to_play != 0) {
// recalculate first_packet_time_to_play -- the latency might change
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency*output_sample_ratio+config.audio_backend_latency_offset*output_sample_rate;
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency*output_sample_ratio+config.audio_backend_latency_offset*output_rate;
if (delta>=0) {
int64_t delta_fp_sec = (delta << 32) / output_sample_rate; // int64_t which is positive
int64_t delta_fp_sec = (delta << 32) / output_rate; // int64_t which is positive
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) / output_sample_rate; // int64_t which is positive
int64_t delta_fp_sec = (abs_delta << 32) / output_rate; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
int64_t max_dac_delay = output_sample_rate/10;
int64_t max_dac_delay = output_rate/10;
int64_t filler_size = max_dac_delay; // 0.1 second -- the maximum we'll add to the DAC
if (local_time_now >= first_packet_time_to_play) {
// we've gone past the time...
// debug(1,"Run past the exact start time by %llu frames, with time now of %llx, fpttp
// of %llx and dac_delay of %d and %d packets;
// flush.",(((tn-first_packet_time_to_play)*output_sample_rate)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read,
// flush.",(((tn-first_packet_time_to_play)*output_rate)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read,
// ab_write));
if (config.output->flush)
@@ -918,7 +918,7 @@ debug(1,"Output sample ratio is %d",output_sample_ratio);
} else
dac_delay = 0;
int64_t gross_frame_gap =
((first_packet_time_to_play - local_time_now) * output_sample_rate) >> 32;
((first_packet_time_to_play - local_time_now) * output_rate) >> 32;
int64_t exact_frame_gap = gross_frame_gap - dac_delay;
if (exact_frame_gap < 0) {
// we've gone past the time...
@@ -1002,17 +1002,17 @@ debug(1,"Output sample ratio is %d",output_sample_ratio);
if (reference_timestamp) { // if we have a reference time
int64_t packet_timestamp = curframe->timestamp; // types okay
int64_t delta = packet_timestamp - reference_timestamp;
int64_t offset = config.latency*output_sample_ratio + config.audio_backend_latency_offset*output_sample_rate -
config.audio_backend_buffer_desired_length*output_sample_rate; // all arguments are int32_t, so expression promotion okay
int64_t offset = config.latency*output_sample_ratio + config.audio_backend_latency_offset*output_rate -
config.audio_backend_buffer_desired_length*output_rate; // all arguments are int32_t, so expression promotion okay
int64_t net_offset = delta + offset; // okay
uint64_t time_to_play = reference_timestamp_time; // type okay
if (net_offset >= 0) {
uint64_t net_offset_fp_sec = (net_offset << 32) / output_sample_rate; // int64_t which is positive
uint64_t net_offset_fp_sec = (net_offset << 32) / output_rate; // int64_t which is positive
time_to_play += net_offset_fp_sec; // using the latency requested...
// debug(2,"Net Offset: %lld, adjusted: %lld.",net_offset,net_offset_fp_sec);
} else {
int64_t abs_net_offset = -net_offset;
uint64_t net_offset_fp_sec = (abs_net_offset << 32) / output_sample_rate; // int64_t which is positive
uint64_t net_offset_fp_sec = (abs_net_offset << 32) / output_rate; // int64_t which is positive
time_to_play -= net_offset_fp_sec;
// debug(2,"Net Offset: %lld, adjusted: -%lld.",net_offset,net_offset_fp_sec);
}
@@ -1034,7 +1034,7 @@ debug(1,"Output sample ratio is %d",output_sample_ratio);
if (wait) {
uint64_t time_to_wait_for_wakeup_fp =
((uint64_t)1 << 32) / input_sample_rate; // this is time period of one frame
((uint64_t)1 << 32) / input_rate; // this is time period of one frame
time_to_wait_for_wakeup_fp *= 4 * 352; // four full 352-frame packets
time_to_wait_for_wakeup_fp /= 3; // four thirds of a packet time
@@ -1125,7 +1125,7 @@ static inline int32_t mean_32(int32_t a, int32_t b) {
// formats accepted so far include S16_LE, S24_LE and S32_LE on a little endinan machine.
// stuff: 1 means add 1; 0 means do nothing; -1 means remove 1
static int stuff_buffer_basic_32(int32_t *inptr, int length, enum sps_format_t l_output_format, char *outptr, int* outlength, int stuff, int dither) {
static int stuff_buffer_basic_32(int32_t *inptr, int length, enum sps_format_t l_output_format, char *outptr, int stuff, int dither) {
int tstuff = stuff;
char *l_outptr = outptr;
if ((stuff > 1) || (stuff < -1) || (length <100)) {
@@ -1174,7 +1174,6 @@ static int stuff_buffer_basic_32(int32_t *inptr, int length, enum sps_format_t l
}
}
pthread_mutex_unlock(&vol_mutex);
*outlength = l_outptr-outptr; // in bytes
return length + tstuff;
}
@@ -1314,14 +1313,9 @@ static void *player_thread_func(void *arg) {
struct inter_threads_record itr;
itr.please_stop = 0;
timestamp_epoch = 0; // indicate that the next timestamp will be the first one.
maximum_timestamp_interval = input_sample_rate * 60; // actually there shouldn't be more than about 13v seconds of a gap between successive rtptimes, at worst
output_sample_rate = input_sample_rate;
if (config.output_rate!=0)
output_sample_rate = config.output_rate;
output_sample_ratio = output_sample_rate/input_sample_rate;
maximum_timestamp_interval = input_rate * 60; // actually there shouldn't be more than about 13v seconds of a gap between successive rtptimes, at worst
output_sample_ratio = config.output_rate/input_rate;
debug(1,"Output sample ratio is %d.",output_sample_ratio);
@@ -1351,7 +1345,7 @@ static void *player_thread_func(void *arg) {
// check that there are enough buffers to accommodate the desired latency and the latency offset
int maximum_latency = config.latency+config.audio_backend_latency_offset*input_sample_rate;
int maximum_latency = config.latency+config.audio_backend_latency_offset*input_rate;
if ((maximum_latency+(352-1))/352 + 10 > BUFFER_FRAMES)
die("Not enough buffers available for a total latency of %d frames. A maximum of %d 352-frame packets may be accommodated.",maximum_latency,BUFFER_FRAMES);
connection_state_to_output = get_requested_connection_state_to_output();
@@ -1386,7 +1380,10 @@ static void *player_thread_func(void *arg) {
char rnstate[256];
initstate(time(NULL), rnstate, 256);
signed short *inbuf, *outbuf, *tbuf, *silence;
signed short *inbuf, *tbuf, *silence;
char *outbuf;
int inbuflength;
int output_bit_depth = 16; // default;
@@ -1409,7 +1406,7 @@ static void *player_thread_func(void *arg) {
// if we are changing any of the parameters of the input, like sample rate or sample depth, then we
// need an intermediate "transition" buffer
// if ((config.output_rate!=0 && (input_sample_rate!=config.output_rate)) || (input_bit_depth!=output_bit_depth)) {
// if ((config.output_rate!=0 && (input_rate!=config.output_rate)) || (input_bit_depth!=output_bit_depth)) {
tbuf = malloc(output_bytes_per_frame*(max_frames_per_packet*output_sample_ratio+max_frame_size_change));
if (tbuf==NULL)
debug(1,"Failed to allocate memory for the transition buffer.");
@@ -1492,6 +1489,9 @@ static void *player_thread_func(void *arg) {
config.output->play(inbuf, inbuflength);
}
} else {
int enable_dither = !((fix_volume==0x10000)&&(input_bit_depth==output_bit_depth)); // avoid dither on audio being sent through without alteration
// here, let's transform the frame of data, if necessary
if (tbuf!=NULL) { //this will be null if no changes are needed
@@ -1595,10 +1595,10 @@ static void *player_thread_func(void *arg) {
if (local_time_now >= reference_timestamp_time) {
// debug(1,"td is %lld.",td);
td = local_time_now - reference_timestamp_time; // this is the positive value. Conversion is positive uint64_t to int64_t, thus okay
td_in_frames = (td * output_sample_rate) >> 32;
td_in_frames = (td * config.output_rate) >> 32;
} else {
td = reference_timestamp_time - local_time_now; // this is the absolute value, which should be negated. Conversion is positive uint64_t to int64_t, thus okay.
td_in_frames = (td * output_sample_rate) >> 32; //use the absolute td value for the present. Types okay
td_in_frames = (td * config.output_rate) >> 32; //use the absolute td value for the present. Types okay
td_in_frames = -td_in_frames;
td = -td; // should be okay, as the range of values should be very small w.r.t 64 bits
}
@@ -1671,10 +1671,10 @@ static void *player_thread_func(void *arg) {
// before we finally commit to this frame, check its sequencing and timing
// require a certain error before bothering to fix it...
if (sync_error > config.tolerance*output_sample_rate) { // int64_t > int, okay
if (sync_error > config.tolerance*config.output_rate) { // int64_t > int, okay
amount_to_stuff = -1;
}
if (sync_error < -config.tolerance*output_sample_rate) {
if (sync_error < -config.tolerance*config.output_rate) {
amount_to_stuff = 1;
}
@@ -1704,74 +1704,58 @@ static void *player_thread_func(void *arg) {
if (config.no_sync!=0)
amount_to_stuff = 0 ; // no stuffing if it's been disabled
if ((amount_to_stuff == 0) && (fix_volume == 0x10000) && ((config.output_rate==0) || (config.output_rate==44100)) && ((config.output_format==0) || (config.output_format==SPS_FORMAT_S16_LE))) {
// if no stuffing needed and no volume adjustment, then
// don't send to stuff_buffer_* and don't copy to outbuf; just send directly to the
// output device...
if (inbuf==NULL)
debug(1,"NULL inframe->data to play -- skipping it.");
else {
if (inbuflength==0)
debug(1,"empty frame to play -- skipping it (2).");
else
config.output->play(inbuf, inbuflength);
}
} else {
#ifdef HAVE_LIBSOXR
switch (config.packet_stuffing) {
case ST_basic:
// if (amount_to_stuff) debug(1,"Basic stuff...");
if (tbuf)
play_samples = stuff_buffer_basic_32((int32_t*)inbuf, inbuflength, (int32_t*)outbuf, amount_to_stuff,output_bit_depth);
else
play_samples = stuff_buffer_basic(inbuf, inbuflength, outbuf, amount_to_stuff);
break;
case ST_soxr:
// if (amount_to_stuff) debug(1,"Soxr stuff...");
if (tbuf)
play_samples = stuff_buffer_soxr_32((int32_t*)inbuf, inbuflength, (int32_t*)outbuf, amount_to_stuff,output_bit_depth);
else
play_samples = stuff_buffer_soxr(inbuf, inbuflength, outbuf, amount_to_stuff);
break;
}
#else
// if (amount_to_stuff) debug(1,"Standard stuff...");
switch (config.packet_stuffing) {
case ST_basic:
// if (amount_to_stuff) debug(1,"Basic stuff...");
if (tbuf)
play_samples = stuff_buffer_basic_32((int32_t*)inbuf, inbuflength, (int32_t*)outbuf, amount_to_stuff,output_bit_depth);
play_samples = stuff_buffer_basic_32((int32_t*)inbuf, inbuflength, config.output_format, outbuf, amount_to_stuff, enable_dither);
else
play_samples = stuff_buffer_basic(inbuf, inbuflength, outbuf, amount_to_stuff);
play_samples = stuff_buffer_basic(inbuf, inbuflength, (short *)outbuf, amount_to_stuff);
break;
case ST_soxr:
// if (amount_to_stuff) debug(1,"Soxr stuff...");
if (tbuf)
play_samples = stuff_buffer_soxr_32((int32_t*)inbuf, inbuflength, (int32_t*)outbuf, amount_to_stuff,output_bit_depth);
else
play_samples = stuff_buffer_soxr(inbuf, inbuflength, (short *)outbuf, amount_to_stuff);
break;
}
#else
// if (amount_to_stuff) debug(1,"Standard stuff...");
if (tbuf)
play_samples = stuff_buffer_basic_32((int32_t*)inbuf, inbuflength, config.output_format, outbuf, amount_to_stuff, enable_dither);
else
play_samples = stuff_buffer_basic(inbuf, inbuflength, (short *)outbuf, amount_to_stuff);
#endif
/*
{
int co;
int is_silent=1;
short *p = outbuf;
for (co=0;co<play_samples;co++) {
if (*p!=0)
is_silent=0;
p++;
}
if (is_silent)
debug(1,"Silence!");
}
*/
if (outbuf==NULL)
debug(1,"NULL outbuf to play -- skipping it.");
else {
if (play_samples==0)
debug(1,"play_samples==0 skipping it (1).");
else
config.output->play(outbuf, play_samples);
/*
{
int co;
int is_silent=1;
short *p = outbuf;
for (co=0;co<play_samples;co++) {
if (*p!=0)
is_silent=0;
p++;
}
if (is_silent)
debug(1,"Silence!");
}
*/
if (outbuf==NULL)
debug(1,"NULL outbuf to play -- skipping it.");
else {
if (play_samples==0)
debug(1,"play_samples==0 skipping it (1).");
else
config.output->play((short*)outbuf, play_samples); // remove the (short*)!
}
// check for loss of sync
// timestamp of zero means an inserted silent frame in place of a missing frame
@@ -1781,7 +1765,7 @@ static void *player_thread_func(void *arg) {
abs_sync_error = -abs_sync_error;
if ((config.no_sync==0) && (inframe->timestamp != 0) && (!please_stop) && (config.resyncthreshold > 0.0) &&
(abs_sync_error > config.resyncthreshold*output_sample_rate)) {
(abs_sync_error > config.resyncthreshold*config.output_rate)) {
sync_error_out_of_bounds++;
// debug(1,"Sync error out of bounds: Error: %lld; previous error: %lld; DAC: %lld;
// timestamp: %llx, time now
@@ -1811,9 +1795,9 @@ static void *player_thread_func(void *arg) {
}
else {
if (tbuf)
play_samples = stuff_buffer_basic_32((int32_t*)inbuf, inbuflength, (int32_t*)outbuf, 0, output_bit_depth); // no stuffing, but volume adjustment
play_samples = stuff_buffer_basic_32((int32_t*)inbuf, inbuflength, config.output_format, outbuf, 0, enable_dither);
else
play_samples = stuff_buffer_basic(inbuf, inbuflength, outbuf, 0); // no stuffing, but volume adjustment
play_samples = stuff_buffer_basic(inbuf, inbuflength, (short *)outbuf, 0); // no stuffing, but volume adjustment
if (outbuf==NULL)
debug(1,"NULL outbuf to play -- skipping it.");
@@ -1821,7 +1805,7 @@ static void *player_thread_func(void *arg) {
if (inbuf==0)
debug(1,"empty frame to play -- skipping it (4).");
else
config.output->play(outbuf, play_samples);
config.output->play((short*)outbuf, play_samples); // remove the (short*)!
}
}
}
@@ -1902,7 +1886,7 @@ static void *player_thread_func(void *arg) {
"%*lli," /* min DAC queue size */
"%*d," /* min buffer occupancy */
"%*d", /* max buffer occupancy */
10, 1000*moving_average_sync_error/output_sample_rate,
10, 1000*moving_average_sync_error/config.output_rate,
10, moving_average_correction * 1000000 / (352*output_sample_ratio),
10, moving_average_insertions_plus_deletions * 1000000 / (352*output_sample_ratio),
12, play_number,
@@ -1923,7 +1907,7 @@ static void *player_thread_func(void *arg) {
"%*lli," /* min DAC queue size */
"%*d," /* min buffer occupancy */
"%*d", /* max buffer occupancy */
10, 1000*moving_average_sync_error/output_sample_rate,
10, 1000*moving_average_sync_error/config.output_rate,
12, play_number,
7, missing_packets,
7, late_packets,
@@ -1942,7 +1926,7 @@ static void *player_thread_func(void *arg) {
"%*llu," /* resend requests */
"%*d," /* min buffer occupancy */
"%*d", /* max buffer occupancy */
10, 1000*moving_average_sync_error/output_sample_rate,
10, 1000*moving_average_sync_error/config.output_rate,
12, play_number,
7, missing_packets,
7, late_packets,
shairport.c
+3 -1
View File
@@ -774,6 +774,8 @@ int main(int argc, char **argv) {
config.audio_backend_buffer_desired_length = 6615; // 0.15 seconds.
config.udp_port_base = 6001;
config.udp_port_range = 100;
config.output_format = SPS_FORMAT_S16_LE; // default
config.output_rate = 44100; // default
config.decoders_supported = 1<<decoder_hammerton; // David Hammerton's decoder supported by default
#ifdef HAVE_APPLE_ALAC
config.decoders_supported += 1<<decoder_apple_alac;
@@ -1039,7 +1041,7 @@ int main(int argc, char **argv) {
debug(1, "disable_synchronization is %d.", config.no_sync);
debug(1, "use_mmap_if_available is %d.", config.no_mmap ? 0 : 1);
debug(1, "output_rate is %d (0 means 44,100).", config.output_rate);
debug(1, "output_format is %d (0 means S16_LE).", config.output_format);
debug(1, "output_format is %d (2 is the default SPS_FORMAT_S16_LE).", config.output_format);
debug(1, "audio backend desired buffer length is %f seconds.",
config.audio_backend_buffer_desired_length);
debug(1, "audio backend latency offset is %f seconds.", config.audio_backend_latency_offset);