cab1291745
Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder
...
Re-enable the test and explicitly call delete on red, even though the
test should die in the AudioEncoderCopyRed cunstructor. Apparently,
things work a little differently under memcheck.
BUG=4108, 3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7942 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 06:58:42 +00:00
eb544460e4
Rename _t struct types in audio_coding.
...
_t names are reserved in POSIX.
R=henrik.lundin@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/34509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 15:23:29 +00:00
e728ee03ba
Remove or rename typedefs with _t prefixes.
...
_t prefixes are reserved for additional typenames in POSIX.
R=henrik.lundin@webrtc.org , hta@webrtc.org , stefan@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/36559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 13:43:55 +00:00
e102e8147b
Enable the iSACfix AudioDecoder test (and make it work again)
...
As far as I can tell, the test should have been enabled again once
https://code.google.com/p/webrtc/issues/detail?id=1353 was fixed, but
it wasn't, and has rotted a bit as a result. I'm not sure why the
number of encoded bytes have changed, but the output seems to be
correct (EncodeDecodeTest encodes, decodes, and compares the result
with the original).
The DecodePlc change is necessary because r7912 added support for that
to the iSACfix AudioDecoder.
BUG=1353, 3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 07:30:23 +00:00
a32487f97b
Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder
...
Fails linux memcheck.
BUG=4108
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:04:55 +00:00
c1c9291e9b
Make an AudioEncoder subclass for RED
...
This class only supports the simple case of payload duplication. That
is, one single encoder is used, and the redundant payload is a one-step
delayed payload.
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
88bdec8c3a
AudioEncoder subclass for iSACfix
...
This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:49:37 +00:00
0b1534c52e
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
...
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
b413a30097
Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.
...
This intrinsics version gives bit-exact result as the current assembly
neon code. And the performance is 38% better than current assembly
neon version, 5.92 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I257e33ef6d634a519fd71adc4f52b06dd655bd9d
Review URL: https://webrtc-codereview.appspot.com/32749004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:23:49 +00:00
3b79daff14
Moving encoded_bytes into EncodedInfo
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 13:31:24 +00:00
0ca768b131
Adding DTX to WebRTC Opus wrapper (relanding).
...
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.
See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/
Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 16:09:35 +00:00
817e50dd7d
Make an AudioEncoder subclass for PCM16B
...
The implementation depends on AudioEncoderPcm to reduce code
duplication.
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
b3ad8cf6ca
Make an AudioEncoder subclass for iSAC
...
BUG=3926
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
abe3f1879c
Checking whether ACM uses codec internal or WebRTC DTX.
...
It was not clear how one could know if ACM is using DTX from WebRTC or codec internal DTX.
This CL makes better use of IsInternalDTXReplacedWithWebRtc() which was designed for G.729 to export such information.
Before
IsInternalDTXReplacedWithWebRtc() gives true only if codec == G729 and G729's internal DTX is replaced with WebRTC DTX.
Now
IsInternalDTXReplacedWithWebRtc() gives true also when codec does not have internal DTX, i.e., must use WebRTC DTX, which is much more logical.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:53:21 +00:00
55d42c32a4
DCHECK: Reference condition parameter in release builds
...
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as
int x = ...
DCHECK_EQ(x, 17);
R=andrew@webrtc.org
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7858
and reverted: https://code.google.com/p/webrtc/source/detail?r=7859
Review URL: https://webrtc-codereview.appspot.com/31169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:32:30 +00:00
d8ca723de7
Remove CELT support from audio_coding.
...
R=henrik.lundin@webrtc.org , juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 11:49:13 +00:00
3cd26b677a
Revert r7858 ("DCHECK: Reference condition parameter in release builds")
...
Apparently Visual Studio is cleverer than I am at figuring out what
local variables are actually unused.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7859 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:57:14 +00:00
3148060e61
DCHECK: Reference condition parameter in release builds
...
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as
int x = ...
DCHECK_EQ(x, 17);
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7858 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:45:47 +00:00
ff1a3e36bd
Make an AudioEncoder subclass for comfort noise
...
BUG=3926
R=bjornv@webrtc.org , kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 07:29:08 +00:00
19dd129c69
Revert 7846 "Adding DTX to WebRTC Opus wrapper"
...
> Adding DTX to WebRTC Opus wrapper
>
> This is a step toward adding Opus DTX support in WebRTC.
>
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
>
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
>
> We transmit the first 1-byte packet to let decoder be in-sync
>
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13219004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 15:11:15 +00:00
4321f175f1
Adding DTX to WebRTC Opus wrapper
...
This is a step toward adding Opus DTX support in WebRTC.
Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
We transmit the first 1-byte packet to let decoder be in-sync
BUG=webrtc:1014
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 13:27:39 +00:00
1784d7cfad
Adding an codec interal CNG test in NetEq.
...
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7843 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:46:39 +00:00
e04a93bcf5
Move the AudioDecoder interface out of NetEq
...
It belongs with the codecs, next to the AudioEncoder interface.
R=andrew@webrtc.org , henrik.lundin@webrtc.org , kjellander@webrtc.org
Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
8911bc52f1
Add AudioEncoder::Max10MsFramesInAPacket
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7834 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:15:55 +00:00
130fef89dd
Bugfix in AudioDecoderTest
...
When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:07:59 +00:00
fcbe36a1d9
Add const qualifier to WebRtcPcm16b_Encode
...
BUG=909
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 18:26:49 +00:00
a1ef7bfa15
ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable.
...
Ideally, this is a stopgap fix until ATTRIBUTE_UNUSED can be given a
proper definition.
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7830 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:53:10 +00:00
cb858ba397
Make an AudioEncoder subclass for iLBC
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@google.com
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:11:44 +00:00
33ccdfa1f5
Relanding r7807.
...
r7807 was reverted to be excluded from the cause of a failure.
It has been verified and can reland now.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 12:14:12 +00:00
52bc4f4797
Revert 7807 "Removing unused opus wrapper APIs."
...
> Removing unused opus wrapper APIs.
>
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
>
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
>
> BUG=
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28139004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 11:00:50 +00:00
e54a6342dd
Removing unused opus wrapper APIs.
...
WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
WebRtcOpus_DecodePlcMaster/Slave() are also removed.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 08:47:25 +00:00
3a52458237
add WebRtcIsacfix_AutocorrNeon's intrinsics version
...
The modification only uses the unique part of the
WebRtcIsacfix_AutocorrC function. Pass FiltersTest.AutocorrFixTest test
on both ARMv7 and ARM64, and the single function performance is similar
with original assembly version on different platforms. If not
specified, the code is compiled by GCC 4.6. The result is the "X
version / C version" ratio, and the less is better.
| run 100k times | cortex-a7 | cortex-a15 |
| use C as the base on each | (1.2Ghz) | (1.7Ghz) |
| CPU target | | |
|----------------------------+-----------+------------|
| Neon asm | 24% | 23% |
| Neon intrinsics (GCC 4.6) | 33% | 32% |
| Neon intrinsics (GCC 4.8) | 27% | 27% |
BUG=3850
R=andrew@webrtc.org , jridges@masque.com
Change-Id: Id6cd0671502fadbebd10b1f5493f5b16c988286f
Review URL: https://webrtc-codereview.appspot.com/27999004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7802 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 21:58:18 +00:00
8dc21dc238
Rename internal AudioEncoder::Encode method to EncodeInternal
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 20:36:03 +00:00
3800e13a3a
Revert r7798 ("Move the AudioDecoder interface out of NetEq")
...
Apparently, it caused all sorts of problems I don't have time to
straighten out right now.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 16:28:17 +00:00
00ba1a7dfd
Move the AudioDecoder interface out of NetEq
...
It belongs with the codecs, next to the AudioEncoder interface.
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 14:23:23 +00:00
fa914e283c
Adding a duration printout to neteq_rtpplay
...
BUG=2692
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7796 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 13:28:53 +00:00
1751ee7d32
Remove -flax-vector-conversions flag for ARM NEON building.
...
Pass compilation on both ARMv7 and ARM64. The generated
binary (audioproc) is byte to byte (with symbol striped) same as
before. The output of audioproc -aecm is also byte to byte same between
C and NEON version on ARMv7 and ARM64.
Change-Id: Ibdf40fe085f6bad1311f59bf9318bbcf37dd7ce5
BUG=3850
R=andrew@webrtc.org , jridges@masque.com
Review URL: https://webrtc-codereview.appspot.com/30239004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7783 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 19:36:14 +00:00
7f1dfa5b61
Adding a payload type to AudioEncoder objects
...
The type is set in the Config struct and is provided in the EncodedInfo
output struct from each Encode() call. The audio_decoder_unittest is
updated to verify correct propagation of the payload type.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7780 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 12:08:39 +00:00
0cd5558f2b
AudioEncoder subclass for G722
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 11:45:51 +00:00
1db20a4180
Adding EncodedInfo struct to AudioEncoder::Encode
...
This struct will be expanded in future changes.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 14:44:50 +00:00
20446e7e56
Move and rename neteq/test/RTPcat to neteq/tools/rtpcat
...
BUG=2692
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7770 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 14:23:01 +00:00
c93437ef96
Add test NetEqDecodingTest.CngFirst
...
This CL adds a test to verify that it is ok to start the stream with
a comfort noise packet.
BUG=4021
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7769 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 11:42:42 +00:00
83317146ba
Adding a new test helper RtpFileWriter and use it in RTPcat
...
This new helper class writes RTP packets to file in rtpdump format.
A unit test is also included.
The new test class is used while re-writing the test tool RTPcat.
BUG=2692
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 11:25:04 +00:00
91d928e737
Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
...
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 15:50:30 +00:00
03499a0e95
Add wav output capability to neteq_rtpplay
...
This CL makes neteq_rtpplay capable of writing to wav files as well as
pcm files. This is done through the new class OutputWavFile, which
wraps a WavWriter object in an AudioSink interface.
BUG=2692
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7740 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 14:50:53 +00:00
1153322cf8
Build fix for MIPS Android Webview build.
...
Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580 ).
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 16:28:32 +00:00
4591fbd09f
Use size_t more consistently for packet/payload lengths.
...
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
6ff3ac1db8
Fix problems if first packet into NetEq is rejected
...
This CL fixes the problem described in issue 4021. In summary, of the
very first packet coming in to NetEq gets rejected because the RTP
payload type is unknown, subsequent GetAudio calls would trigger asserts
(in debug builds). The problem was that the first_packet_ variable was
reset and new_codec_ was set, even though the packet was discarded
further down the line. Now, these variables are modified after the
packet has been verified.
BUG=4021
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7724 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 14:14:49 +00:00
ed91068bf1
Create a NetEq test for when the first incoming payload type is unknown
...
This test is currently disabled. It triggers an issue where the NetEq
will trigger asserts on subsequent GetAudio calls if the first inserted
packet is rejected, for instance since the payload type is unknown to
NetEq.
BUG=4021
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 11:01:02 +00:00
40af3a56e4
Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty"
...
This reverts r7719. It failed to compile in Chromium.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7720 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-19 13:46:52 +00:00
