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Fix silence bug and hack code to avoid the stuff procedures

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This commit is contained in:
Mike Brady
2016-09-05 17:11:30 +01:00
parent e6e1acd91e
commit cb7c12d636
1 changed files with 14 additions and 9 deletions
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player.c
+14 -9
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@@ -845,11 +845,11 @@ debug(1,"Output sample ratio is %d",output_sample_ratio);
// debug(2,"Zero length silence buffer needed with gross_frame_gap of %lld and dac_delay of %lld.",gross_frame_gap,dac_delay);
// the fs (number of frames of silence to play) can be zero in the DAC doesn't start ouotputting frames for a while -- it could get loaded up but not start responding for many milliseconds.
if (fs!=0) {
silence = malloc(input_bytes_per_frame*fs);
silence = malloc(output_bytes_per_frame*fs);
if (silence==NULL)
debug(1,"Failed to allocate %d byte silence buffer.",fs);
else {
memset(silence, 0, input_bytes_per_frame*fs);
memset(silence, 0, output_bytes_per_frame*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);
@@ -1158,6 +1158,8 @@ static void *player_thread_func(void *arg) {
break;
}
debug(1,"Output frame bytes is %d.",output_bytes_per_frame);
// create and start the timing, control and audio receiver threads
pthread_t rtp_audio_thread, rtp_control_thread, rtp_timing_thread;
pthread_create(&rtp_audio_thread, NULL, &rtp_audio_receiver, (void *)&itr);
@@ -1223,6 +1225,8 @@ static void *player_thread_func(void *arg) {
}
debug(1,"Output bit depth is %d.",output_bit_depth);
// if we are changing any of the parameters of the input, like sample rate or sample depth, then we
// need an intermediate "transition" buffer
@@ -1334,12 +1338,12 @@ static void *player_thread_func(void *arg) {
} else if (both < INT16_MIN) {
both = INT16_MIN;
}
uint16_t sboth = (uint16_t)both;
int16_t sboth = (int16_t)both;
ls = sboth;
rs = sboth;
} break;
case ST_reverse_stereo: {
uint16_t t = ls;
int16_t t = ls;
ls = rs;
rs = t;
} break;
@@ -1359,15 +1363,15 @@ static void *player_thread_func(void *arg) {
*outps++=rs;
break;
case 24: {
uint32_t t = ls<<8;
int32_t t = ls<<8;
*outpl++=t;
uint32_t u = rs<<8;
int32_t u = rs<<8;
*outpl++=u;
} break;
case 32: {
uint32_t t = ls<<16;
int32_t t = ls<<16;
*outpl++=t;
uint32_t u = rs<<16;
int32_t u = rs<<16;
*outpl++=u;
} break;
}
@@ -1522,7 +1526,8 @@ 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 ((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 (1) {
// 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...