admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
7,075 Commits 1 Branch 0 Tags
Commit Graph

1830 Commits

Author SHA1 Message Date
stefan@webrtc.org
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
pbos@webrtc.org
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
minyue@webrtc.org
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
minyue@webrtc.org
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
minyue@webrtc.org
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
andrew@webrtc.org
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
henrik.lundin@webrtc.org
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
andrew@webrtc.org
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
kwiberg@webrtc.org
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
kwiberg@webrtc.org
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
henrik.lundin@webrtc.org
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
marpan@webrtc.org
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
aluebs@webrtc.org
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
marpan@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
kjellander@webrtc.org
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..24b4c73/DEPS

Clang version was not updated in this roll.

BUG=4059
TBR=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
andresp@webrtc.org
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
pbos@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
asapersson@webrtc.org
83b5200f95 Add framerate for complete received frames to histogram stats: 
"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
aluebs@webrtc.org
cc144deaab Make bands vector in SplittingFilter Analysis const 
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
aluebs@webrtc.org
8789376cd3 Move ChannelBuffer class to channel_buffer file 
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
asapersson@webrtc.org
d952c40c7e Add receive bitrates to histogram stats: 
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")

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

Review URL: https://webrtc-codereview.appspot.com/27189005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 07:38:56 +00:00
aluebs@webrtc.org
79b9eba3ab Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands 
Implemented the 3 bands splitting filter bank by:
1. Upsample by 4/3.
2. Split twice into 2 bands.
3. Discard upper most band, because it is empty anyway.

A unittest was also implemented:
1. Generate a signal from presence or absence of sine waves of different frequencies.
2. Split into 3 bands and check their presence or absence.
3. Recombine the bands.
4. Calculate delay (as it is an IIR it depends on frequency).
5. Check that the cross correlation of input and output is high enough at that delay.

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

Review URL: https://webrtc-codereview.appspot.com/31029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 20:21:38 +00:00
andrew@webrtc.org
a56a2c57cf Enabling building with NEON on ARM64 
This patch enables NEON on ARM64 platform. Passed building both on
Android ARMv7 and Android ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/25069004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 17:01:40 +00:00
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
aff1751c96 Add new test for VP8 packetizer to test tight partitions 
It was discovered that if remaining_bytes is an exact multiple of
max_payload_len in RtpPacketizerVp8::CalcNextSize, then the packetizer
will produce too many packets (i.e., split the payload into more
packets than needed).

This CL adds a test to trigger the problem.

BUG=4019
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 12:36:58 +00:00
kjellander@webrtc.org
8562f23acb OWNERS: Remove tomasl@ and mallinath@ 
mallinath@ has left the team and tomasl@ says he doesn't
know why he's owner in webrtc/test/channel_transport

R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 10:05:05 +00:00
pbos@webrtc.org
4f16c874c6 Simplifying VideoReceiver and JitterBuffer. 
Removing frame_buffers_ array and dual-receiver mechanism. Also adding
some thread annotations to VCMJitterBuffer.

R=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/27239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 09:06:48 +00:00
pbos@webrtc.org
9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28029004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 08:25:50 +00:00
andrew@webrtc.org
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
kjellander@webrtc.org
ad0e71c9a3 Update mock_frame_dropper.h to use size_t 
This mock was missed in the work of
https://webrtc-codereview.appspot.com/23129004 since the file
is not currently used by any test in this repo.

BUG=chromium:81439
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-21 09:40:57 +00:00
pkasting@chromium.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
6f6ef72950 Add DCHECK to ensure that NetEq's packet buffer is not empty 
This DCHECK ensures that one packet was inserted after the buffer was
flushed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-19 13:02:24 +00:00
aluebs@webrtc.org
087da13fe8 Add empty 3 band splitting filter API 
This is only an empty API that will never be used. For now is 48kHz not supported in AudioProcessing. For that it needs to be added in InitializeLocked. But before the 3 band filter bank needs to be populated.

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

Review URL: https://webrtc-codereview.appspot.com/30139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 23:01:23 +00:00
pkasting@chromium.org
2656bf813f Fix ExpectedQueueTimeMs() to avoid truncation or overflow. 
BUG=none
TEST=none
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 22:21:14 +00:00
pbos@webrtc.org
f5b56fbc41 Annotate COMPILE_ASSERT with __attribute__((unused)). 
Also renames UNUSED -> ATTRIBUTE_UNUSED to be able to use this when
building peerconnection_jni.cc which apparently has this defined to
something else.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=4018

Review URL: https://webrtc-codereview.appspot.com/28039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 13:47:38 +00:00
henrik.lundin@webrtc.org
966a708b93 Use RtpFileSource in NetEqDecodingTest 
This CL removes the dependency on the old NETEQTEST_RTPpacket class
from the NetEqDecodingTest code, and also removes the dependency
from the modules_unittests target to neteq_test_tools.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 09:08:38 +00:00
aluebs@webrtc.org
be05c74ec8 Wrap the splitting filter in its own class 
This doesn't change the behavior at all.
The logic behind this is having one class which manages all the splitting filters, because in the future we plan to add a 3 band one for 48kHz support.
It also breaks the dependency of the AudioBuffer with the filter states of these filters (which are going to be different for the 3 band one). The AudioBuffer is complicated enough and is going to need changes to support 3 bands in the future, so any simplification is a good idea.
On top of that it eliminates repeated code in the APM (now only iterating over channels, but then also deciding in how many bands to split). This should be managed by the AudioBuffer directly.

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

Review URL: https://webrtc-codereview.appspot.com/32469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 22:18:10 +00:00
pbos@webrtc.org
ece3890d3a Report total bitrate for all streams in GetStats. 
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 11:52:04 +00:00