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Some bug fixes and hacks. Not working.

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This commit is contained in:
Mike Brady
2016-09-09 12:34:21 +01:00
parent 28a9bd9ac9
commit d1c6478035
1 changed files with 32 additions and 21 deletions
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player.c
+32 -21
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@@ -587,10 +587,10 @@ static inline void process_sample(int32_t sample,char**outp,enum sps_format_t fo
int64_t hyper_sample = sample;
int result;
// first, modify volume, if necessary
if (volume==0x10000)
//if (volume==0x10000)
hyper_sample<<32;
else
hyper_sample*=(volume<<16);
//else
// hyper_sample*=(volume<<16);
// next, do dither, if necessary
if (dither) {
@@ -629,24 +629,25 @@ static inline void process_sample(int32_t sample,char**outp,enum sps_format_t fo
}
// move the result to the desired position in the int64_t
char *op = *outp;
short ss;
switch (format) {
case SPS_FORMAT_S32_LE:
hyper_sample>>64-32;
hyper_sample>>=(64-32);
*op++ = (int32_t)hyper_sample;
result=4;
break;
case SPS_FORMAT_S24_LE:
hyper_sample>>64-24;
hyper_sample>>=(64-24);
*op = (int32_t)hyper_sample;
result=4;
break;
case SPS_FORMAT_S16_LE:
hyper_sample>>64-16;
hyper_sample>>=(64-16);
*op = (int16_t)hyper_sample;
result=2;
break;
case SPS_FORMAT_S8:
hyper_sample>>64-8;
hyper_sample>>=(64-8);
*op = (int8_t)hyper_sample;
result=1;
break;
@@ -858,14 +859,14 @@ 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_rate;
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency*output_sample_ratio+config.audio_backend_latency_offset*config.output_rate;
if (delta>=0) {
int64_t delta_fp_sec = (delta << 32) / output_rate; // int64_t which is positive
int64_t delta_fp_sec = (delta << 32) / config.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_rate; // int64_t which is positive
int64_t delta_fp_sec = (abs_delta << 32) / config.output_rate; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
@@ -880,25 +881,25 @@ 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_rate;
int64_t delta = (first_packet_timestamp - reference_timestamp)+config.latency*output_sample_ratio+config.audio_backend_latency_offset*config.output_rate;
if (delta>=0) {
int64_t delta_fp_sec = (delta << 32) / output_rate; // int64_t which is positive
int64_t delta_fp_sec = (delta << 32) / config.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_rate; // int64_t which is positive
int64_t delta_fp_sec = (abs_delta << 32) / config.output_rate; // int64_t which is positive
first_packet_time_to_play=reference_timestamp_time-delta_fp_sec;
}
int64_t max_dac_delay = output_rate/10;
int64_t max_dac_delay = config.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_rate)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read,
// flush.",(((tn-first_packet_time_to_play)*config.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 +919,7 @@ 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_rate) >> 32;
((first_packet_time_to_play - local_time_now) * config.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 +1003,17 @@ 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_rate -
config.audio_backend_buffer_desired_length*output_rate; // all arguments are int32_t, so expression promotion okay
int64_t offset = config.latency*output_sample_ratio + config.audio_backend_latency_offset*config.output_rate -
config.audio_backend_buffer_desired_length*config.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_rate; // int64_t which is positive
uint64_t net_offset_fp_sec = (net_offset << 32) / config.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_rate; // int64_t which is positive
uint64_t net_offset_fp_sec = (abs_net_offset << 32) / config.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);
}
@@ -1490,7 +1491,8 @@ static void *player_thread_func(void *arg) {
}
} else {
int enable_dither = !((fix_volume==0x10000)&&(input_bit_depth==output_bit_depth)); // avoid dither on audio being sent through without alteration
int enable_dither = 0;
// 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
@@ -1535,7 +1537,15 @@ static void *player_thread_func(void *arg) {
}
// here, replicate the samples if you're upsampling
for (j=0;j<output_sample_ratio;j++) {
// raise the 16-bit sample to 32 bits.
int32_t t = ls<<16;
*outpl++=t;
int32_t u = rs<<16;
*outpl++=u;
/*
switch (output_bit_depth) {
case 16:
*outps++=ls;
@@ -1554,6 +1564,7 @@ static void *player_thread_func(void *arg) {
*outpl++=u;
} break;
}
*/
}
}