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Working reliably with all 16-bit speeds

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This commit is contained in:
Mike Brady
2016-09-04 16:34:56 +01:00
parent 46267f7574
commit eca364691e
1 changed files with 47 additions and 43 deletions
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player.c
+47 -43
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@@ -387,7 +387,7 @@ static int init_decoder(int32_t fmtp[12]) {
field bitDepth uint8_t describes the bit depth of the source PCM data (maximum value = 32)
field pb uint8_t currently unused tuning parameter.
field pb uint8_t currently unused tuning parametetbugr.
value should be set to 40
field mb uint8_t currently unused tuning parameter.
@@ -746,11 +746,11 @@ static abuf_t *buffer_get_frame(void) {
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency+config.audio_backend_latency_offset;
if (delta>=0) {
int64_t delta_fp_sec = (delta << 32) / 44100; // int64_t which is positive
int64_t delta_fp_sec = (delta << 32) / output_sample_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) / 44100; // int64_t which is positive
int64_t delta_fp_sec = (abs_delta << 32) / output_sample_rate; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
@@ -768,22 +768,22 @@ static abuf_t *buffer_get_frame(void) {
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency+config.audio_backend_latency_offset;
if (delta>=0) {
int64_t delta_fp_sec = (delta << 32) / 44100; // int64_t which is positive
int64_t delta_fp_sec = (delta << 32) / output_sample_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) / 44100; // int64_t which is positive
int64_t delta_fp_sec = (abs_delta << 32) / output_sample_rate; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
int64_t max_dac_delay = 4410;
int64_t max_dac_delay = output_sample_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)*44100)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read,
// 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,
// ab_write));
if (config.output->flush)
@@ -803,7 +803,7 @@ static abuf_t *buffer_get_frame(void) {
} else
dac_delay = 0;
int64_t gross_frame_gap =
((first_packet_time_to_play - local_time_now) * 44100) >> 32;
((first_packet_time_to_play - local_time_now) * output_sample_rate) >> 32;
int64_t exact_frame_gap = gross_frame_gap - dac_delay;
if (exact_frame_gap < 0) {
// we've gone past the time...
@@ -890,12 +890,12 @@ static abuf_t *buffer_get_frame(void) {
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) / 44100; // int64_t which is positive
uint64_t net_offset_fp_sec = (net_offset << 32) / output_sample_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) / 44100; // int64_t which is positive
uint64_t net_offset_fp_sec = (abs_net_offset << 32) / output_sample_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);
}
@@ -917,7 +917,7 @@ static abuf_t *buffer_get_frame(void) {
if (wait) {
uint64_t time_to_wait_for_wakeup_fp =
((uint64_t)1 << 32) / 44100; // this is time period of one frame
((uint64_t)1 << 32) / input_sample_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
@@ -993,13 +993,14 @@ static inline short shortmean(short a, short b) {
// stuff: 1 means add 1; 0 means do nothing; -1 means remove 1
static int stuff_buffer_basic(short *inptr, int length, short *outptr, int stuff) {
int tstuff = stuff;
if ((stuff > 1) || (stuff < -1) || (length <100)) {
// debug(1, "Stuff argument to stuff_buffer must be from -1 to +1 and length >100.");
return length;
tstuff = 0; // if any of these conditions hold, don't stuff anything/
}
int i;
int stuffsamp = length;
if (stuff)
if (tstuff)
// stuffsamp = rand() % (length - 1);
stuffsamp =
(rand() % (length - 2)) + 1; // ensure there's always a sample before and after the item
@@ -1009,8 +1010,8 @@ static int stuff_buffer_basic(short *inptr, int length, short *outptr, int stuff
*outptr++ = dithered_vol(*inptr++);
*outptr++ = dithered_vol(*inptr++);
};
if (stuff) {
if (stuff == 1) {
if (tstuff) {
if (tstuff == 1) {
// debug(3, "+++++++++");
// interpolate one sample
//*outptr++ = dithered_vol(((long)inptr[-2] + (long)inptr[0]) >> 1);
@@ -1025,8 +1026,8 @@ static int stuff_buffer_basic(short *inptr, int length, short *outptr, int stuff
// if you're removing, i.e. stuff < 0, copy that much less over. If you're adding, do all the rest.
int remainder = length;
if (stuff<0)
remainder = remainder+stuff; // don't run over the correct end of the output buffer
if (tstuff<0)
remainder = remainder+tstuff; // don't run over the correct end of the output buffer
for (i = stuffsamp; i < remainder; i++) {
*outptr++ = dithered_vol(*inptr++);
@@ -1035,22 +1036,23 @@ static int stuff_buffer_basic(short *inptr, int length, short *outptr, int stuff
}
pthread_mutex_unlock(&vol_mutex);
return length + stuff;
return length + tstuff;
}
#ifdef HAVE_LIBSOXR
// stuff: 1 means add 1; 0 means do nothing; -1 means remove 1
static int stuff_buffer_soxr(short *inptr, int length, short *outptr, int stuff) {
if ((stuff > 1) || (stuff < -1) || (length < 100)) {
// debug(1, "Stuff argument to sox_stuff_buffer must be from -1 to +1 and length must be > 100.");
return length;
int tstuff = stuff;
if ((stuff > 1) || (stuff < -1) || (length <100)) {
// debug(1, "Stuff argument to stuff_buffer must be from -1 to +1 and length >100.");
tstuff = 0; // if any of these conditions hold, don't stuff anything/
}
int i;
short *ip, *op;
ip = inptr;
op = outptr;
if (stuff) {
if (tstuff) {
// debug(1,"Stuff %d.",stuff);
soxr_io_spec_t io_spec;
io_spec.itype = SOXR_INT16_I;
@@ -1061,9 +1063,9 @@ static int stuff_buffer_soxr(short *inptr, int length, short *outptr, int stuff)
size_t odone;
soxr_error_t error = soxr_oneshot(length, length + stuff, 2, /* Rates and # of chans. */
soxr_error_t error = soxr_oneshot(length, length + tstuff, 2, /* Rates and # of chans. */
inptr, length, NULL, /* Input. */
outptr, length + stuff, &odone, /* Output. */
outptr, length + tstuff, &odone, /* Output. */
&io_spec, /* Input, output and transfer spec. */
NULL, NULL); /* Default configuration.*/
@@ -1082,7 +1084,7 @@ static int stuff_buffer_soxr(short *inptr, int length, short *outptr, int stuff)
}
// keep the last (dpm) samples, to mitigate the Gibbs phenomenon
op = outptr + (length + stuff - gpm) * sizeof(short);
op = outptr + (length + tstuff - gpm) * sizeof(short);
ip = inptr + (length - gpm) * sizeof(short);
for (i = 0; i < gpm; i++) {
*op++ = *ip++;
@@ -1093,7 +1095,7 @@ static int stuff_buffer_soxr(short *inptr, int length, short *outptr, int stuff)
if (fix_volume != 65536.0) {
// pthread_mutex_lock(&vol_mutex);
op = outptr;
for (i = 0; i < length + stuff; i++) {
for (i = 0; i < length + tstuff; i++) {
*op = dithered_vol(*op);
op++;
*op = dithered_vol(*op);
@@ -1111,7 +1113,7 @@ static int stuff_buffer_soxr(short *inptr, int length, short *outptr, int stuff)
};
// pthread_mutex_unlock(&vol_mutex);
}
return length + stuff;
return length + tstuff;
}
#endif
@@ -1366,7 +1368,9 @@ static void *player_thread_func(void *arg) {
default:
die("Shairport Sync only supports 16 bit input");
}
inbuf = tbuf;
inbuflength*=output_sample_ratio;
}
// We have a frame of data. We need to see if we want to add or remove a frame from it to
@@ -1397,10 +1401,10 @@ 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 * 44100) >> 32;
td_in_frames = (td * output_sample_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 * 44100) >> 32; //use the absolute td value for the present. Types okay
td_in_frames = (td * output_sample_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
}
@@ -1511,13 +1515,13 @@ static void *player_thread_func(void *arg) {
// 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 (inframe->data==NULL)
if (inbuf==NULL)
debug(1,"NULL inframe->data to play -- skipping it.");
else {
if (inframe->length==0)
if (inbuflength==0)
debug(1,"empty frame to play -- skipping it (2).");
else
config.output->play(inframe->data, inframe->length);
config.output->play(inbuf, inbuflength);
}
} else {
@@ -1526,16 +1530,16 @@ static void *player_thread_func(void *arg) {
switch (config.packet_stuffing) {
case ST_basic:
// if (amount_to_stuff) debug(1,"Basic stuff...");
play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, amount_to_stuff);
play_samples = stuff_buffer_basic(inbuf, inbuflength, outbuf, amount_to_stuff);
break;
case ST_soxr:
// if (amount_to_stuff) debug(1,"Soxr stuff...");
play_samples = stuff_buffer_soxr(inframe->data, inframe->length, outbuf, amount_to_stuff);
play_samples = stuff_buffer_soxr(inbuf, inbuflength, outbuf, amount_to_stuff);
break;
}
#else
// if (amount_to_stuff) debug(1,"Standard stuff...");
play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, amount_to_stuff);
play_samples = stuff_buffer_basic(inbuf, inbuflength, outbuf, amount_to_stuff);
#endif
/*
@@ -1593,21 +1597,21 @@ static void *player_thread_func(void *arg) {
if (fix_volume == 0x10000)
if (inframe->data==NULL)
debug(1,"NULL inframe->data to play -- skipping it.");
if (inbuf==NULL)
debug(1,"NULL inbuf to play -- skipping it.");
else {
if (inframe->length==0)
if (inbuflength==0)
debug(1,"empty frame to play -- skipping it (3).");
else
config.output->play(inframe->data, inframe->length);
config.output->play(inbuf, inbuflength);
}
else {
play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, 0); // no stuffing, but volume adjustment
play_samples = stuff_buffer_basic(inbuf, inbuflength, outbuf, 0); // no stuffing, but volume adjustment
if (outbuf==NULL)
debug(1,"NULL outbuf to play -- skipping it.");
else {
if (inframe->length==0)
if (inbuf==0)
debug(1,"empty frame to play -- skipping it (4).");
else
config.output->play(outbuf, play_samples);
@@ -1692,8 +1696,8 @@ static void *player_thread_func(void *arg) {
"%*d," /* min buffer occupancy */
"%*d", /* max buffer occupancy */
10, moving_average_sync_error,
10, moving_average_correction * 1000000 / 352,
10, moving_average_insertions_plus_deletions * 1000000 / 352,
10, moving_average_correction * 1000000 / (352*output_sample_ratio),
10, moving_average_insertions_plus_deletions * 1000000 / (352*output_sample_ratio),
12, play_number,
7, missing_packets,
7, late_packets,