public function RtpHeaderParser::Parse with old signature restored as deprecated.
BUG=webrtc:5277
TBR=åsapersson
NOTRY=True
Review URL: https://codereview.webrtc.org/1550283002
Cr-Commit-Position: refs/heads/master@{#11135}
The issue occurred when deserializing and then serializing a rejected
content description, which doesn't have the ICE ufrag/pwd in the first
place.
BUG=webrtc:5105
Review URL: https://codereview.webrtc.org/1534363002
Cr-Commit-Position: refs/heads/master@{#11134}
We'll only use temporary address for IPv6. However, due to a bug in iOS sdk, the necessary headers are not included. This change copies the minimum necessary definitions such that we could retrieve the ip attributes.
BUG=webrtc:4343
Review URL: https://codereview.webrtc.org/1531763006
Cr-Commit-Position: refs/heads/master@{#11114}
Opus has become the mostly used codec in WebRTC. There, however, is no bit exactness test for Opus decoding in NetEq.
The new RTP file is generated by the following steps:
1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960 opus 1
2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source code for RTPjitter to know how to define such a file.)
BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the reference output.
Review URL: https://codereview.webrtc.org/1515113002
Cr-Commit-Position: refs/heads/master@{#11113}
PrepareReportBlock and AddReportBlock private functions merged:
PrepareReportBlock moved report block from statistic to temporary structure
AddReportBlock copied that temporary structure into temporary map right after.
Thanks to rtcp packet classes that temporary structure is now unneccesary.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1538833002
Cr-Commit-Position: refs/heads/master@{#11112}
VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps.
Unused in webrtc, also unused in everything indexed by google and chromium code search. With the exception of the magicflute plugin, which I'm told doesn't matter.
Review URL: https://codereview.webrtc.org/1532133002
Cr-Commit-Position: refs/heads/master@{#11108}
1. It signals network changed events whenever there are more than one IP address in a network.
2. It does not signal network changed events if a network disconnects and connects again.
Also changed DumpNetworks for better debugging.
BUG=webrtc:5096
Review URL: https://codereview.webrtc.org/1421433003
Cr-Commit-Position: refs/heads/master@{#11107}
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.
No functional changes.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1537273003
Cr-Commit-Position: refs/heads/master@{#11101}
This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
This will allow an app to create senders with the same stream id,
without SDP munging.
Review URL: https://codereview.webrtc.org/1538673002
Cr-Commit-Position: refs/heads/master@{#11092}
Currently, FFT is performance when AEC buffers farend signal. This has some drawbacks
1. memory inefficiency: two ring buffers are needed;
2. computation inefficiency: if ringbuffer gets wrapped around, some FFT computation will be wasted;
3. accessibility: the main AEC function looses accessibility to the time-domain signal.
Therefore, this CL tries to buffer time domain data, which is buffered any way if a debugging macro is defined, and calculate the FFTs where they are actually used.
BUG=
Review URL: https://codereview.webrtc.org/1512573003
Cr-Commit-Position: refs/heads/master@{#11091}
Add the --quiet flag to the download_from_google_storage runhooks
step to prevent it from spamming the console when all the files
are already downloaded.
NOTRY=True
Review URL: https://codereview.webrtc.org/1527713003
Cr-Commit-Position: refs/heads/master@{#11090}
