The problem was that only ACKed packets were subtracted from in_flight_, but lost packets were never removed, which caused TCP to stop sending eventually.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43239004
Cr-Commit-Position: refs/heads/master@{#9041}
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.
Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.
BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36889004
Cr-Commit-Position: refs/heads/master@{#9040}
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
int8_t*, and char*. Previously, they accepted void*, meaning that
any kind of pointer was accepted. I think requiring an explicit
cast in cases where the input array isn't already of a byte-sized
type is a better compromise between convenience and safety.
2. data() can now return a uint8_t* instead of a char*, which seems
more appropriate for a byte array, and is harder to mix up with
zero-terminated C strings. data<int8_t>() is also available so
that callers that want that type instead won't have to cast, as
is data<char>() (which remains the default until all existing
callers have been fixed).
3. Constructors, SetData(), and AppendData() now accept arrays
natively, not just decayed to pointers. The advantage of this is
that callers don't have to pass the size separately.
4. There are new constructors that allow setting size and capacity
without initializing the array. Previously, this had to be done
separately after construction.
5. Instead of TransferTo(), Buffer now supports swap(), and move
construction and assignment, and has a Pass() method that works
just like std::move(). (The Pass method is modeled after
scoped_ptr::Pass().)
R=jmarusic@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42989004
Cr-Commit-Position: refs/heads/master@{#9033}
Also change to use virtual_packet_length_bytes in order to print the
actual packet size of the complete packet even when the RTP file only
contains RTP headers.
BUG=2692
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51559004
Cr-Commit-Position: refs/heads/master@{#9025}
Add separate functions for returning stats from send/receive stream and updated how functions are used.
Add test implementation for histogram methods in system_wrappers/interface/metrics.h.
BUG=4519
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49639004
Cr-Commit-Position: refs/heads/master@{#9009}
The AGC is currently bumping up the mic volume to 33% at startup if it is below that level. This is to avoid getting stuck in a poor state from which the AGC can not move, simply a too low input audio level. For some users, 33% is instead too loud.
This CL gives the user the possibility to set that level at create time.
- Extends the Config ExperimentalAgc with a startup_mic_volume for the user to set if desired. Note that the bump up does not apply to the legacy AGC and the "regular" AGC is controlled by ExperimentalAgc.
- Without any actions, the same default value as previously is used.
- In addition I removed a return value from InitializeExperimentalAgc() and InitializeTransient()
This has been tested by building Chromium on Mac and verify through apprtc that
1) startup_mic_volume = 128 bumps up to 50%.
2) startup_mic_volume = 500 (out of range) bumps up to 100%.
3) startup_mic_volume = 0 bumps up to 4%, the AGC min level.
BUG=4529
TESTED=locally
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43109004
Cr-Commit-Position: refs/heads/master@{#9004}
With this change, the currently used encoder is held in a scoped_ptr.
iSAC is a special case, since the encoder instance is also a decoder
instance, so it may have to be available also if another send codec is
used. This is accomplished by having a separate scoped_ptr for iSAC.
Remove mirror ID from ACM codec database functions, and remove unused
functions from the database.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48729004
Cr-Commit-Position: refs/heads/master@{#8982}
ViEChannels without default encoders doesn't register a receive codec by
default. This makes VideoReceiver::Decode return early, causing a
high-priority thread to effectively be busy looping. This would be
expected to wreck more havoc in a more cross-platform manner than it has
visibly done. On Windows XP however it manages to bring the whole
machine to a grinding halt forcing a reboot if CPU usage hits 100%.
BUG=chromium:470013
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48049004
Cr-Commit-Position: refs/heads/master@{#8976}
Also prevents that we try to restore audio mode when it has not been changed.
TBR=glaznev
BUG=NONE
TEST=AppRTCDemo and verify that volume control switches from "Ringtone to Phone" mode when call starts and switches back to Ringtone mode when call ends.
Review URL: https://webrtc-codereview.appspot.com/46879004
Cr-Commit-Position: refs/heads/master@{#8975}
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.
This should fix pbos' TODO in i420_video_frame.h.
Tested on Linux with the following command lines:
$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug
BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.orgTBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46819004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8973}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
- removed commented code lines used during development
- excluded fft.c since there are neon optimizations used and a removal may cause a performance regression
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48799004
Cr-Commit-Position: refs/heads/master@{#8967}
There is no point in returning an error when Free() fails. In fact it can only happen if we have a null pointer as object. There is further no place where the return value is used.
Affected components are
- aec
- aecm
- agc
- ns
BUG=441
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50579004
Cr-Commit-Position: refs/heads/master@{#8966}
Now that android_webview_build is no longer supported, remove build
conditionals referencing it and also remove the extra level of
indirection used to reference the cpufeatures target.
BUG=chromium:440793
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44119005
Patch from Richard Coles <torne@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8963}
CopyCodecSpecific nulls out the rtpheader pointer hence causing the crash downstream.
More details about the codec type enums:
There are 2 enums defined. webrtc::VideoCodecType webrtc::RtpCodecTypes and they don't match. Inside CopyCodecSpecific in generic_encoder.cc, it was converted from the first to the 2nd type. At that point, it'll be kRtpVideoNone (as the effect of memset to 0). kRtpVideoNone is a bad value as it could cause assert. Later, it'll be reset to kRtpVideoGeneric in RTPSender::SendOutgoingData so it's not a concern.
BUG=4511
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Committed: https://crrev.com/29b1a1c0c7c6f4b1ae4d63844b1dfaa7a72530a0
Cr-Commit-Position: refs/heads/master@{#8951}
Review URL: https://webrtc-codereview.appspot.com/47999004
Cr-Commit-Position: refs/heads/master@{#8955}
CopyCodecSpecific nulls out the rtpheader pointer hence causing the crash downstream.
More details about the codec type enums:
There are 2 enums defined. webrtc::VideoCodecType webrtc::RtpCodecTypes and they don't match. Inside CopyCodecSpecific in generic_encoder.cc, it was converted from the first to the 2nd type. At that point, it'll be kRtpVideoNone (as the effect of memset to 0). kRtpVideoNone is a bad value as it could cause assert. Later, it'll be reset to kRtpVideoGeneric in RTPSender::SendOutgoingData so it's not a concern.
BUG=4511
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47999004
Cr-Commit-Position: refs/heads/master@{#8951}
This reverts commit cf3c83e76c273309558c86fda915410f65b7a899.
Reverting EventWrapper split did not fix the issue, re-landing.
BUG=chromium:470013
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49629004
Cr-Commit-Position: refs/heads/master@{#8946}
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
hence trivial.
The macro name may in fact mislead the user to assume a cast/truncation to int32_t is done.
- Removing usage of it.
- Some style changes.
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46749005
Cr-Commit-Position: refs/heads/master@{#8918}
