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sample times are now contiguous. DTMF tone length changed to 80ms

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to be multiple of 20ms as used by most codecs.
This commit is contained in:
gardners 2012-04-25 05:38:45 +09:30
parent 12be1e326e
commit 6c044d5e8d
2 changed files with 31 additions and 13 deletions
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1
serval.h
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@ -1275,6 +1275,7 @@ typedef struct vomp_call_state {
int ringing;
unsigned long long create_time;
unsigned long long last_activity;
unsigned long long audio_clock;
int audio_started;
int last_sent_status;
int next_status_time;

43
vomp.c
View File

@ -221,23 +221,38 @@ int vomp_send_status(vomp_call_state *call,int flags,overlay_mdp_frame *arg)
unsigned short *len=&mdp.out.payload_length;
unsigned char *p=&mdp.out.payload[0];
/* write the sample end-time in milliseconds since call establishment */
unsigned long long now=overlay_gettime_ms();
unsigned long long now_rel_call=now-call->create_time;
p[(*len)++]=(now_rel_call>>24)&0xff;
p[(*len)++]=(now_rel_call>>16)&0xff;
p[(*len)++]=(now_rel_call>>8)&0xff;
p[(*len)++]=(now_rel_call>>0)&0xff;
/* record sample in recent list.
XXX - What timestamp to attach to the sample?
Two obvious choices:
1. The sample is for the most recent n milliseconds; or
2. The sample is for the next n milliseconds following the
last sample.
/* record sample in recent list */
Option 1 introduces all sorts of problems with sample production
jitter, where as option 2 has no such problems, but simply requires the
producer of audio to ensure that they provide exactly the right amount
of audio, or risk the call getting out of sync. This is a fairly
reasonable expectation, or else things go to pot.
Note that in-call slew is the responsibility of the player, not the
recorder of audio. Basically if the audio queue starts to bank up,
then the player needs to drop samples.
*/
vomp_sample_block *sb=call->recent_samples;
int rotor=call->recent_sample_rotor%VOMP_MAX_RECENT_SAMPLES;
sb[rotor].codec=arg->vompevent.audio_sample_codec;
sb[rotor].endtime=now_rel_call;
sb[rotor].starttime=now_rel_call-vomp_codec_timespan(sb[rotor].codec);
sb[rotor].endtime=call->audio_clock+vomp_codec_timespan(sb[rotor].codec)-1;
sb[rotor].starttime=call->audio_clock;
call->audio_clock=sb[rotor].endtime+1;
bcopy(&arg->vompevent.audio_bytes[0],&sb[rotor].bytes[0],
vomp_sample_size(sb[rotor].codec));
/* write the sample end-time in milliseconds since call establishment */
p[(*len)++]=(call->audio_clock>>24)&0xff;
p[(*len)++]=(call->audio_clock>>16)&0xff;
p[(*len)++]=(call->audio_clock>>8)&0xff;
p[(*len)++]=(call->audio_clock>>0)&0xff;
/* stuff frame with most recent sample blocks as a form of preemptive
retransmission. But don't make the packets too large. */
while ((*len)<256) {
@ -248,7 +263,7 @@ int vomp_send_status(vomp_call_state *call,int flags,overlay_mdp_frame *arg)
rotor--; if (rotor<0) rotor+=VOMP_MAX_RECENT_SAMPLES;
rotor%=VOMP_MAX_RECENT_SAMPLES;
if ((!sb[rotor].endtime)||(sb[rotor].endtime==now_rel_call)) break;
if ((!sb[rotor].endtime)||(sb[rotor].endtime==call->audio_clock+1)) break;
}
call->recent_sample_rotor++;
call->recent_sample_rotor%=VOMP_MAX_RECENT_SAMPLES;
@ -1080,7 +1095,7 @@ int vomp_codec_timespan(int c)
case VOMP_CODEC_16SIGNED: return 20;
case VOMP_CODEC_8ULAW: return 20;
case VOMP_CODEC_8ALAW: return 20;
case VOMP_CODEC_DTMF: return 70;
case VOMP_CODEC_DTMF: return 80;
case VOMP_CODEC_ENGAGED: return 20;
case VOMP_CODEC_ONHOLD: return 20;
case VOMP_CODEC_CALLERID: return 0;
@ -1236,7 +1251,9 @@ int app_vomp_dtmf(int argc, const char *const *argv, struct command_line_option
int digit=vomp_parse_dtmf_digit(digits[i]);
if (digit<0) return WHYF("'%c' is not a DTMF digit.",digits[i]);
mdp.vompevent.audio_bytes[mdp.vompevent.audio_sample_bytes]
=(digit<<4)+7; /* 70ms standard tone duration */
=(digit<<4); /* 80ms standard tone duration, so that it is a multiple
of the majority of codec time units (70ms is the nominal
DTMF tone length for most systems). */
if (overlay_mdp_send(&mdp,0,0)) WHY("Send DTMF failed.");
}