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savoirfairelinux
jami-daemon
Commits
f1b0f902
Commit
f1b0f902
authored
18 years ago
by
yanmorin
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Adding timestamp just before sending microphone sound (rtp)
parent
85019ee5
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Changes
2
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2 changed files
src/audio/audiortp.cpp
+22
-1
22 additions, 1 deletion
src/audio/audiortp.cpp
src/audio/ringbuffer.cpp
+4
-2
4 additions, 2 deletions
src/audio/ringbuffer.cpp
with
26 additions
and
3 deletions
src/audio/audiortp.cpp
+
22
−
1
View file @
f1b0f902
...
...
@@ -260,6 +260,23 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
SRC_DATA
src_data
;
#ifdef DATAFORMAT_IS_FLOAT
src_data
.
data_in
=
_dataAudioLayer
;
/* _debug("inb: %d %f %f %f %f %f %f %f %f %f %f %f %f\n", nbSample, _dataAudioLayer[0],
_dataAudioLayer[1 ],
_dataAudioLayer[2 ],
_dataAudioLayer[3 ],
_dataAudioLayer[4 ],
_dataAudioLayer[5 ],
_dataAudioLayer[6 ],
_dataAudioLayer[7 ],
_dataAudioLayer[8 ],
_dataAudioLayer[9 ],
_dataAudioLayer[10],
_dataAudioLayer[11],
_dataAudioLayer[12],
_dataAudioLayer[13],
_dataAudioLayer[14],
_dataAudioLayer[15]
);*/
#else
src_short_to_float_array
(
_dataAudioLayer
,
_floatBuffer48000
,
nbSample
);
src_data
.
data_in
=
_floatBuffer48000
;
...
...
@@ -272,6 +289,9 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
src_simple
(
&
src_data
,
SRC_SINC_BEST_QUALITY
/*SRC_SINC_MEDIUM_QUALITY*/
,
1
);
// 1 = channel
nbSample
=
src_data
.
output_frames_gen
;
//if (nbSample > RTP_20S_8KHZ_MAX) { _debug("Alert from mic, nbSample %d is bigger than expected %d\n", nbSample, RTP_20S_8KHZ_MAX); }
/*_debug("ina: %d %f %f\n", nbSample, _floatBuffer8000[0],
_floatBuffer8000[1 ]
);*/
src_float_to_short_array
(
_floatBuffer8000
,
_intBuffer8000
,
nbSample
);
toSIP
=
_intBuffer8000
;
}
else
{
...
...
@@ -285,7 +305,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
#endif
}
if
(
nbSample
<
RTP_20S_8KHZ_MAX
)
{
if
(
nbSample
<
(
RTP_20S_8KHZ_MAX
-
10
)
)
{
// if only 10 is missing, it's ok
// fill end with 0...
//_debug("begin: %p, nbSample: %d\n", toSIP, nbSample);
//_debug("has to fill: %d chars at %p\n", (RTP_20S_8KHZ_MAX-nbSample)*sizeof(int16), toSIP + nbSample);
...
...
@@ -301,6 +321,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
// encode divise by two
// Send encoded audio sample over the network
if
(
compSize
>
RTP_20S_8KHZ_MAX
)
{
_debug
(
"! ARTP: %d should be %d
\n
"
,
compSize
,
RTP_20S_8KHZ_MAX
);}
timestamp
+=
time
->
getSecond
();
if
(
!
_sym
)
{
_sessionSend
->
putData
(
timestamp
,
_sendDataEncoded
,
compSize
);
}
else
{
...
...
This diff is collapsed.
Click to expand it.
src/audio/ringbuffer.cpp
+
4
−
2
View file @
f1b0f902
...
...
@@ -87,8 +87,9 @@ RingBuffer::Put(void* buffer, int toCopy, unsigned short volume) {
//fprintf(stderr, "P");
while
(
toCopy
)
{
block
=
toCopy
;
if
(
block
>
(
mBufferSize
-
pos
))
// from current pos. to end of buffer
if
(
block
>
(
mBufferSize
-
pos
))
{
// from current pos. to end of buffer
block
=
mBufferSize
-
pos
;
}
// put the data inside the buffer.
if
(
volume
!=
100
)
{
...
...
@@ -138,8 +139,9 @@ RingBuffer::Get(void *buffer, int toCopy, unsigned short volume) {
//fprintf(stderr, "G");
while
(
toCopy
)
{
block
=
toCopy
;
if
(
block
>
(
mBufferSize
-
mStart
))
if
(
block
>
(
mBufferSize
-
mStart
))
{
block
=
mBufferSize
-
mStart
;
}
if
(
volume
!=
100
)
{
SFLDataFormat
*
start
=
(
SFLDataFormat
*
)(
mBuffer
+
mStart
);
...
...
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