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
97d0489058
Add video send bitrates to histogram stats:
...
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")
Add retransmitted bytes to StreamDataCounters.
Change in UpdateRtpStats to also update counters for retransmitted packet.
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 09:47:53 +00:00
7ba9f27f2b
Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper.
...
This makes it possible to clean up the recipes a lot.
BUG=
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7837 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 06:46:13 +00:00
86b6d65ef1
Remove no longer used video codec test framework.
...
Moves one test to the vp8 unittests which might still be good to have.
Also does a bit of clean up in vp8 unittests.
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7835 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 00:02:45 +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
edeea91803
Change all system clock types to int64_t in bitrate_controller.
...
They are both compared to int64_t types inside the class, and is being called
with int64_t types. Could possibly cause bugs.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7832 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 19:46:23 +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
ee43263a50
Cleaned up real_fft APIs due to non-existing NEON code
...
There are NEON APIs that are not used. Cleaning that up for better overview.
BUG=3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7827 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 16:36:22 +00:00
ba8138ba38
Change type of nack_last_time_sent_full_ from uint32_t to int64_t.
...
Could cause nack requests to be sent too frequently.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7825 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 13:29:02 +00:00
fb01376eca
Adjust some parameters for VP9 tests.
...
Needed for the next/upcoming libvpx roll.
BUG=
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7821 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 06:25:51 +00:00
5af8cd77e2
Add codereview.settings to the /webrtc subdirectory
...
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/webrtc
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7818 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 13:43:35 +00:00
0b38478885
Add support for parsing header only RTP dumps with bwe_rtp_play.
...
Also adds support for printing the original_length in rtp_to_text.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 15:43:49 +00:00
9f79fe684a
Merge remote bitrate estimator changes.
...
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7811 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 15:34:06 +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
8c9ff203c5
Redo the change of https://webrtc-codereview.appspot.com/30949004/
...
The previous change causes a build issue as there is subclass of TransportChannel in chromium. To break the circular dependency, a stub of implementation for GetState() is provided and will be removed once the jingle_glue::MockTransportChannel has the function defined.
TBR=pthatcher@webrtc.org
BUG=411086
Review URL: https://webrtc-codereview.appspot.com/34369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7806 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 07:56:02 +00:00
fd8422938c
Revert "Implement GetState() for channel's connectivity check state."
...
This reverts commit ff72f9e692d0918b32646dadaf382aa4355d8437.
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7805 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 00:51:59 +00:00
ff72f9e692
Implement GetState() for channel's connectivity check state.
...
Previously, IceState is considered completed when there is only one connection (and the rest was trimmed). However, since the trimming logic is only done within the scope of network, when IPv6 and IPv4 both exist, the completion event is never fired.
This change adds the GetState() to each channel and it could decide what Completion means. The transport object then aggregates all channels before determining it's completed.
Each channel's IceState will be aggregrated at Transport level for overall Ice state
BUG=411086
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 00:14:49 +00:00
fd4acf6d55
Adding WebRtcSpl_MaxAbsValueW16 intrinsics version
...
The modification only uses the unique part of the WebRtcSpl_MaxAbsValue
function. Pass Spltest.MinMaxOperationTest conformance 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 | 32% | 15% |
| Neon intrinsics (GCC 4.6) | 36% | 37% |
| Neon intrinsics (GCC 4.8) | 35% | 18% |
BUG=3580
R=andrew@webrtc.org , jridges@masque.com
Change-Id: Ia2f6822ec58774b401cc440b6751a97e540b5048
Review URL: https://webrtc-codereview.appspot.com/30109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7803 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 21:59:02 +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
d1fac61e8f
Remove need for assembly offset generation in aecm and ns module.
...
All *neon.S files in aecm and ns modules have been removed. We need no
assembly offset generation now.
Pass byte to byte conformance test for aecm and ns test in audioproc
between new NEON (written in intrinsics) version and C version on both
ARMv7 and ARM64.
BUG=3580
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I05d43d0c04d00bead65ca8c8fda25f0a42394b2b
Review URL: https://webrtc-codereview.appspot.com/32229004
Patch from Zhongwei Yai <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7800 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 17:54:38 +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
c3e097cdc5
Add Android test runner script for WebRTC.
...
The Android test execution toolchain scripts in Chromium
has been causing headaches for us several times. Mostly
because they're tailored at running Chrome tests only.
Wrapping their script in our own avoids the pain of
upstreaming new test names to Chromium and rolling them
in to get them running on our bots.
TESTED=Ran a test on a local device using:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests --verbose --isolate-file-path webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate --release
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7794 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 09:57:08 +00:00
511f8a8ef2
TurnPort should ignore STUN binding reponses when using shared socket.
...
BUG=4043
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7792 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 02:17:07 +00:00
001f3b9818
Adjust parameter in videoprocessor_integration_test for vp9.
...
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7791 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 02:00:12 +00:00
a7384a1126
Simplify audio_buffer APIs
...
Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.
BUG=webrtc:3146
R=andrew@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 01:06:35 +00:00
ceca014b8b
Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9.
...
BUG=4059
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7789 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 01:05:43 +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
ac68ef9ad4
Clear 2 unused functions in audio processing aecm module.
...
unused functions:
WebRtcAecm_WindowAndFFTNeon
WebRtcAecm_InverseFFTAndWindowNeon
BUG=3580
R=andrew@webrtc.org
Change-Id: I12c50a8706d40f9ea98208b5733c00ede7b1f435
Review URL: https://webrtc-codereview.appspot.com/30269004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7782 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 18:33:52 +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
84515f841d
Roll chromium_revision 309cf65..24b4c73
...
Two VP9 tests needed to be disabled (see webrtc:4059) to make all tests pass.
Relevant changes:
* src/third_party/android_tools: ea50ccc..4c47ef6
* src/third_party/icu: dd72764..866ff69
* src/third_party/libvpx: 2e5ced5..429874c
* src/third_party/nss: 258342e..bb4e75a
* src/third_party/yasm/source/patched-yasm: c960eb1..4671120
* src/tools/gyp: 0a381c0..fe00999
* src/tools/swarming_client: 5b827c9..1d4965c
Details: 309cf65..24b4c73TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7778 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 08:48:08 +00:00
5950b645b9
Use c++11 features in webrtc/base/network.cc as a test to see if we can use them.
...
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25209005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7777 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 23:18:27 +00:00
7f722492f1
Set simulcastIdx field to zero even if it has no meaning.
...
Helps to be able to memcmp between 2 parses of the same packet.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7773 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 15:29:29 +00:00
273a414b0e
Report encoded frame size in VideoSendStream.
...
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 15:23:21 +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
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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
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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
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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
83b5200f95
Add framerate for complete received frames to histogram stats:
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"WebRTC.Video.CompleteFramesReceivedPerSecond".
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7762 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 10:17:13 +00:00
cc144deaab
Make bands vector in SplittingFilter Analysis const
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BUG=webrtc:3146
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7761 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 00:26:27 +00:00
8789376cd3
Move ChannelBuffer class to channel_buffer file
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No change in functionallity.
BUG=webrtc:3146
R=andrew@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 23:40:25 +00:00
d87213af49
Remove unused RtpStatistics struct.
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This unused struct is basically a copy of RtcpStatistics in
webrtc/common_types.h. I expect this wasn't properly removed when that
one was added.
R=tommi@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/25239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7758 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 13:48:35 +00:00
