a8cc3440b1
Allowing RED decoding for Opus.
...
BUG=4247
TEST=reproduced and fixed the bug
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41809004
Cr-Commit-Position: refs/heads/master@{#8364}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8364 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 14:02:17 +00:00
648f5d6dc7
pcm16b: Make input arrays const and use uint8_t[] for byte arrays
...
There were both uint8 and uint16 versions of the pcm16b encode and
decode functions; this patch removes the latter.
BUG=909
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34139004
Cr-Commit-Position: refs/heads/master@{#8309}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8309 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 09:19:09 +00:00
c11348b5d7
Fixing a bug in expand_rate calculation for stereo signal.
...
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41849004
Cr-Commit-Position: refs/heads/master@{#8307}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8307 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 08:36:07 +00:00
1c6239a3b6
G711: Make input arrays const and use uint8_t[] for byte arrays
...
BUG=909
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39809004
Cr-Commit-Position: refs/heads/master@{#8294}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8294 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 12:56:16 +00:00
2b69eab077
Restructure GYP for vp9, opus and direct trace
...
This is needed to make the build more flexible for some use cases.
BUG=4185
R=andresp@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34099004
Cr-Commit-Position: refs/heads/master@{#8290}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8290 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:01:40 +00:00
74d27884af
Remove defined(__cplusplus) tests in C++ code.
...
This header is a C++ header (it contains keywords such as 'class'
and 'public'). It is not necessary to test defined(__cplusplus).
That test is appropriate in a C header that may be included by C++
code.
R=henrik.lundin@webrtc.org , jan.skoglund@webrtc.org , sprang@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/38899004
Cr-Commit-Position: refs/heads/master@{#8256}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8256 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 19:18:21 +00:00
0e81fdf5d2
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
...
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40569004
Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 23:54:40 +00:00
a1dfbf1e5c
WebRtcG722_Decode: Input array should be const uint8_t[]
...
BUG=909
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38799004
Cr-Commit-Position: refs/heads/master@{#8224}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8224 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 08:58:39 +00:00
026b892e72
Using << on an int8_t or uint8_t will output a character rather than a number.
...
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:54:19 +00:00
7d2b6a9346
Enable Clang warning implicit-fallthrough and annotate the code.
...
BUG=4242
R=henrik.lundin@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34899004
Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 18:38:13 +00:00
4dba2e98a2
Consolidate anonymous namespace content and file-static methods to all be in the
...
anonymous namespace, in preparation for refactoring a few of the functions a
little.
No code change.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8155 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 19:59:32 +00:00
7dba7860c7
Setting Opus target application.
...
This CL is to allow to set Opus target application at the creation of an encoder.
According to Opus spec, there are three applications:
OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY
BUG=
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 16:01:50 +00:00
a32d15448d
Disable tests failing on Android ARM64 (Nexus9).
...
BUG=4198,4199,4200
TBR=andrew@webrtc.org
TESTED=Printed using #pragma message to check that the define was properly used.
Review URL: https://webrtc-codereview.appspot.com/33919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8090 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 12:46:01 +00:00
2ebfac5649
Remove COMPILE_ASSERT and use static_assert everywhere
...
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org , andrew@webrtc.org , hellner@chromium.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
86e1e487e7
Move system_wrappers.gyp files to the proper directory.
...
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
3df38b442f
Unify the two copies of compile_assert.h
...
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org , andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
c14e3572c6
common_audio: Made input signal const in WebRtcSplFilterMAFastQ12()
...
BUG=3353, 1133
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8037 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 05:50:52 +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
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
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
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
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
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
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
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
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
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
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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
