Commit Graph

7228 Commits

Author SHA1 Message Date
pthatcher@webrtc.org
02c21dbef1 Make one OWNERS files for all of webrtc/libjingle so we don't need approval from webrtc/OWNERS every time we want to add a directory.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7919 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:04:41 +00:00
andrew@webrtc.org
08df9b2841 Add a manageable command-line tool for AudioProcessing.
This is the start of a replacement for the venerable and unwieldly
process_test.cc (aka audioproc). It will be limited to:
- Reading WAV or aecdebug protobuf files.
- Calling the float AudioProcessing interface.
- Requiring aecdebug files for running bi-directional stream
components (e.g. AEC).

This initial version only handles WAV files.

R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7918 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:57:15 +00:00
aluebs@webrtc.org
cf6d0b64ef Add 48kHz support to AGC
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.

Originally reviewed here:
https://webrtc-codereview.appspot.com/26339004/

BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:56:09 +00:00
andrew@webrtc.org
2510d11c0f Add (safe) uint32_t cast to fix Win64 build.
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:47:42 +00:00
andrew@webrtc.org
048c5029f5 Handle all permissible PCM fields with WavReader.
I discovered the hard way that Adobe Audition writes an 18 byte format
header with an extra (zero) extension size field. Although:
https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
indicates this field shouldn't exist for PCM, the documentation here:
http://www-mmsp.ece.mcgill.ca/documents/AudioFormats/WAVE/WAVE.html
doesn't list it as strictly forbidden, only that it _must_ exist for
non-PCM formats.

Audition can write metadata to the file after the audio data, which is
also not forbidden. We now ensure to read only up to the audio payload
length to avoid reading the metadata.

R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7915 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:17:21 +00:00
pbos@webrtc.org
451a133f44 Add AGC manager tests.
R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/35539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 14:48:47 +00:00
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
kjellander@webrtc.org
0198933b3d Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate().
This should fix the following error I'm seeing in Win8 GN trybot:

e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(78)
: error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(30)
: warning C4373:
'webrtc::BitrateControllerImpl::RtcpBandwidthObserverImpl::OnReceivedEstimatedBitrate':
virtual function overrides 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate',
previous versions of the compiler did not override when parameters only differed by const/volatile qualifiers
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\rtp_rtcp\interface\rtp_rtcp_defines.h(286)
: see declaration of 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate'

http://build.chromium.org/p/tryserver.chromium.win/builders/win8_chromium_gn_dbg/builds/23/steps/compile/logs/stdio

The above was triggered in CL https://codereview.chromium.org/802113002/

BUG=None
R=kjellander@google.com, kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37409004

Patch from Thiago Farina <tfarina@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7911 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:29:59 +00:00
asapersson@webrtc.org
d08d389ce8 Add field to counters for when first rtp/rtcp packet is sent/received.
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:03:11 +00:00
bjornv@webrtc.org
b395a5ea65 audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/
include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.

BUG=
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 10:38:10 +00:00
guoweis@webrtc.org
55360ae402 Revert "Add adapter_type into Candidate object."
This reverts commit aaf02cc2d4.

BUG=
TBR=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 05:28:10 +00:00
marpan@webrtc.org
d021bbbc9e Fix vp9 setting in vie loopback test.
If vp9 codec was selected then videoCodec.codecSpecific.VP8.numberOfTemporalLayers was being set.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/37389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 00:21:47 +00:00
guoweis@webrtc.org
aaf02cc2d4 Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 23:03:10 +00:00
pkasting@chromium.org
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
aluebs@webrtc.org
96a626262a Remove 20ms support in AGC
Today, 10ms is the standard chunk length used in whole AudioProcessing, so this was only adding unnecessary complexity and maintainance.
Removing it doesn't change the bahavior in any use case of today.

R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7904 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 21:54:50 +00:00
guoweis@webrtc.org
1f05c45976 Reenable test case P2PTransportChannelTest.TestIPv6Connections
BUG=3317
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 21:25:54 +00:00
pbos@webrtc.org
a7f77720cb Merge in AGC manager and AGC tools.
R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/37379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7902 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 16:33:16 +00:00
bjornv@webrtc.org
903b4ae603 Removes unused test files by audio_processing/transient
BUG=
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 16:13:05 +00:00
bjornv@webrtc.org
dd322136fe resources/audio_processing: Removed unused test files
Two files not used by any tests are removed.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7900 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 15:57:11 +00:00
minyue@webrtc.org
6fd9308420 Suppressing warnings in GetRTT() in VoE.
GetRTT() was separated from GetRTPStatistics() but the warnings were not updated.

Now GetRTT() is only only used by GetRTPStatistics() and the warning pops up pointlessly and too often.

This CL is to suppress these warnings and maintain a proper warning for GetRTPStatistics().

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7899 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 14:56:44 +00:00
tommi@webrtc.org
e2e199b894 Clean up StatsObserver's OnComplete methods (address TODOs).
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 13:22:54 +00:00
pbos@webrtc.org
3440fe1bc5 Use webrtc_root instead of DEPTH for iSAC.
Un-breaks chromium.webrtc.fyi. Broken as Chromium doesn't have webrtc/
checked out in root.

TBR=bjornv@webrtc.org,tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7897 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 10:56:50 +00:00
buildbot@webrtc.org
032b802a8c (Auto)update libjingle 82121498-> 82126219
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:48:07 +00:00
tommi@webrtc.org
dd0601fbcf Remove unneeded ctor and add a more practical one
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:47:49 +00:00
tommi@webrtc.org
69bc5a300f Add thread asserts to StatsCollector.
Also adding a "ForTest" postfix to a test-only method.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:44:48 +00:00
pbos@webrtc.org
788acd17ad Merge audio_processing changes.
R=aluebs@webrtc.org, bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
pbos@webrtc.org
fb108b5a28 Revert r7885.
Breaks compile step of other code where network name of
cricket::Candidate is used.

TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/31229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 08:04:50 +00:00
andrew@webrtc.org
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
pbos@webrtc.org
18a3896bd2 Revert r7886:7887.
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.

TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:03:04 +00:00
andrew@webrtc.org
40e4767f2b Add NEON intrinsics version for min_max_operations_neon.c
WebRtcSpl_MinValueW32Neon, WebRtcSpl_MaxValueW32Neon, WebRtcSpl_MaxValueW16Neon
and WebRtcSpl_MaxAbsValueW32Neon are added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Id461d64c3313f56147edadd2231e8845574ead2a

Review URL: https://webrtc-codereview.appspot.com/28259004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7889 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 06:07:47 +00:00
magjed@webrtc.org
e575e9c40f Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-14 11:09:23 +00:00
pthatcher@webrtc.org
e9db7fe80c Put pseudotcp back because remoting uses it.
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-13 01:56:39 +00:00
pthatcher@webrtc.org
dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 21:04:42 +00:00
guoweis@webrtc.org
8c9d79a29d Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 19:21:14 +00:00
tommi@webrtc.org
c57310b982 Switch kStatsValueName* constants to be enums instead of char*.
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 17:41:28 +00:00
henrik.lundin@webrtc.org
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
kjellander@webrtc.org
c8bc717905 Fix webrtc gn windows build.
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7882 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 12:10:46 +00:00
jansson@webrtc.org
f68faa542a Removing manual test pages because they have been moved to github.
BUG=https://github.com/GoogleChrome/webrtc/issues/203
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7881 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 09:30:41 +00:00
pthatcher@webrtc.org
40b276ea7b Cleanup little things found when refactoring.
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 02:44:30 +00:00
aluebs@webrtc.org
27d106bcf7 Move the downmixing out of AudioBuffer
This provides more flexibility if some component in AudioProcessing wants to operate before downmixing.
Now the AudioProcessing does only track the processing rate, but not the processing number of channels. This is tracked by the AudioBuffer itself and can be changed at any time to one smaller or equal the input number of channels. For each chunk it is reset to input number of channels and the end it should be equal to the output number of channels.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7879 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 17:09:21 +00:00
minyue@webrtc.org
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
pbos@webrtc.org
5f162c8509 Merge AEC changes.
R=bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/34459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7877 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:46:59 +00:00
pbos@webrtc.org
2b19f06312 Wire up RTT statistics to webrtc::Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:26:09 +00:00
pbos@webrtc.org
13518951e3 Remove old_factory from WebRtcVideoEngine.
Minor pending cleanup.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7875 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:14:30 +00:00
perkj@webrtc.org
128fabaf7b Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""
Original cl description:

Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 12:25:57 +00:00
kjellander@webrtc.org
626c09f6a3 Move isolate path into webrtc/build/android/test_runner.py
This will make it easier to execute tests and allows
for more cleanup in the buildbot recipes.
Now tests can be listed using:
webrtc/build/android/test_runner.py gtest --help
and executed like
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests

TESTED=
Ran:
webrtc/build/android/test_runner.py gtest --help
and verified the tests were listed.
I wiped /sdcard/resources on my device, executed:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests
and verified it passed and that resources/audio_coding/testfile32kHz.pcm
was copied to the device.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7873 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 11:59:46 +00:00
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
minyue@webrtc.org
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