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BC
public
mediastreamer2
Commits
fc7273d1
Commit
fc7273d1
authored
Jun 04, 2014
by
Gautier Pelloux-Prayer
Browse files
fix audio stream unit tests: do not set rtcp interval but use the default setting
parent
f813759e
Changes
1
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2 additions
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6 deletions
+2
-6
tester/mediastreamer2_audio_stream_tester.c
tester/mediastreamer2_audio_stream_tester.c
+2
-6
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tester/mediastreamer2_audio_stream_tester.c
View file @
fc7273d1
...
...
@@ -213,9 +213,6 @@ static float adaptive_audio_stream(int codec_payload, int initial_bitrate,int ta
params
.
max_bandwidth
=
target_bw
;
params
.
max_buffer_size
=
initial_bitrate
;
float
bw_usage_ratio
;
// this variable should not be changed, since algorithm results rely on this value
// (the bigger it is, the more accurate is bandwidth estimation)
int
rtcp_interval
=
RTCP_DEFAULT_REPORT_INTERVAL
;
float
marielle_send_bw
;
media_stream_enable_adaptive_bitrate_control
(
&
marielle
->
stream
->
ms
,
TRUE
);
...
...
@@ -226,8 +223,7 @@ static float adaptive_audio_stream(int codec_payload, int initial_bitrate,int ta
stream_manager_start
(
margaux
,
codec_payload
,
marielle
->
local_rtp
,
-
1
,
NULL
,
RECORDED_16K_1S_FILE
);
rtp_session_enable_network_simulation
(
margaux
->
stream
->
ms
.
sessions
.
rtp_session
,
&
params
);
rtp_session_set_rtcp_report_interval
(
margaux
->
stream
->
ms
.
sessions
.
rtp_session
,
rtcp_interval
);
wait_for_until
(
&
marielle
->
stream
->
ms
,
&
margaux
->
stream
->
ms
,
&
marielle
->
stats
.
number_of_EndOfFile
,
10
,
rtcp_interval
*
max_recv_rtcp_packet
);
wait_for_until
(
&
marielle
->
stream
->
ms
,
&
margaux
->
stream
->
ms
,
&
marielle
->
stats
.
number_of_EndOfFile
,
10
,
2500
*
max_recv_rtcp_packet
);
marielle_send_bw
=
media_stream_get_up_bw
(
&
marielle
->
stream
->
ms
);
bw_usage_ratio
=
marielle_send_bw
/
params
.
max_bandwidth
;
...
...
@@ -252,7 +248,7 @@ static void adaptive_opus_audio_stream() {
// on EDGEBW, both should be overconsumming
bw_usage
=
adaptive_audio_stream
(
OPUS_PAYLOAD_TYPE
,
8000
,
EDGE_BW
,
14
);
CU_ASSERT_IN_RANGE
(
bw_usage
,
2
.
f
,
3
.
f
);
// bad! since this codec cant change its ptime and it is the lower bitrate, no improvement can occur
CU_ASSERT_IN_RANGE
(
bw_usage
,
1
.
f
,
3
.
f
);
// bad! since this codec cant change its ptime and it is the lower bitrate, no improvement can occur
bw_usage
=
adaptive_audio_stream
(
OPUS_PAYLOAD_TYPE
,
48000
,
EDGE_BW
,
11
);
CU_ASSERT_IN_RANGE
(
bw_usage
,
1
.
f
,
1
.
2
f
);
// bad!
...
...
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