All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
It allows to use rtc::Buffer in templates that expect std container,
e.g. it can now be used as ::testing::ElementsAreArray parameter
Bug: None
Change-Id: I97d7ffb13393d02850ddb213f7a1d01129b10b05
Reviewed-on: https://chromium-review.googlesource.com/539635
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18809}
I can't believe it's been this long before I found out we didn't have
them...
BUG=none
Review-Url: https://codereview.webrtc.org/2804673002
Cr-Commit-Position: refs/heads/master@{#17544}
The former is always defined (by webrtc/base/checks.h) to either 0 or
1, whereas the latter isn't necessarily defined.
NOTRY=true
BUG=webrtc:6451
Review-Url: https://codereview.webrtc.org/2384693002
Cr-Commit-Position: refs/heads/master@{#14474}
In addition to setting or appending from another Buffer, which was
already possible, this allows for e.g. std::vector and rtc::ArrayView
arguments.
Review-Url: https://codereview.webrtc.org/2293983002
Cr-Commit-Position: refs/heads/master@{#14073}
An rtc::Buffer is used to write output of RBSP parsing, usually one byte
at a time. It turns out that container will then expand its capacity one
byte at a time, for each byte reallocating the whole buffer and copying
the contents over, turning this into an O(n^2) operation.
Fix is for now only to preallocate the container storage. Longer term, I
think we should mull over if we really need custom containers...
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/2073763002 .
Cr-Commit-Position: refs/heads/master@{#13167}
The VUI part an SPS may specify max_num_reorder_frames and
max_dec_frame_buffering. These may cause a decoder to buffer a number
of frame prior allowing decode, leading to delays, even if no frames
using such references (ie B-frames) are sent.
Because of this we update any SPS block emitted by the encoder.
Also, a bunch of refactoring of H264-related code to reduce code
duplication.
BUG=
Review-Url: https://codereview.webrtc.org/1979443004
Cr-Commit-Position: refs/heads/master@{#13010}
And redefine rtc::Buffer as
using Buffer = BufferT<uint8_t>;
(In the long run, I'd like to remove the type alias and rename the
template to just rtc::Buffer, but that requires all current users of
Buffer to start saying Buffer<uint8_t> instead, and since Buffer is
used in the API, we can't do that in one step.)
The immediate reason for the new template is that we'd like to use
BufferT<int16_t> in the AudioDecoder interface.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/1929903002
Cr-Commit-Position: refs/heads/master@{#12564}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1823503002
Cr-Commit-Position: refs/heads/master@{#12062}
Reason for revert:
I'm really sorry for having to revert this but it seems this hit an unexpected compile error downstream:
webrtc/media/sctp/sctpdataengine.cc: In function 'void cricket::VerboseLogPacket(const void*, size_t, int)':
webrtc/media/sctp/sctpdataengine.cc:172:37: error: invalid conversion from 'const void*' to 'void*' [-fpermissive]
data, length, direction)) != NULL) {
^
In file included from webrtc/media/sctp/sctpdataengine.cc:20:0:
third_party/usrsctp/usrsctplib/usrsctp.h:964:1: error: initializing argument 1 of 'char* usrsctp_dumppacket(void*, size_t, int)' [-fpermissive]
usrsctp_dumppacket(void *, size_t, int);
^
I'm sure you can fix this easily and just re-land this CL, while I'm going to look into how to add this warning at the public bots (on Monday).
Original issue's description:
> Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
>
> This CL removes copy and assign support from Buffer and changes various
> parameters from Buffer to CopyOnWriteBuffer so they can be passed along
> and copied without actually copying the underlying data.
>
> With this changed some parameters to be "const" and fixed an issue when
> creating a CopyOnWriteBuffer with empty data.
>
> BUG=webrtc:5155
>
> Committed: https://crrev.com/944c39006f1c52aee20919676002dac7a42b1c05
> Cr-Commit-Position: refs/heads/master@{#12058}
TBR=kwiberg@webrtc.org,tkchin@webrtc.org,tommi@webrtc.org,pthatcher@webrtc.org,jbauch@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1817753003
Cr-Commit-Position: refs/heads/master@{#12060}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1785713005
Cr-Commit-Position: refs/heads/master@{#12058}
They are invoked with the maximum size of the data to be added, and a
callable that generates that data, like this:
buffer.AppendData(10, [] (rtc::ArrayView<uint8_t> av) {
for (uint8_t i = 0; i != 5; ++i)
av[i] = i;
return 5;
});
The callable returns the number of bytes actually written, and the
final Buffer size will be adjusted accordingly. SetData and AppendData
both return the number of bytes added (i.e. the return value of the
callable).
These versions will be useful when converting AudioEncoder::Encode to use Buffer rather than raw pointers.
Also added a few tests for the new functionality.
Review URL: https://codereview.webrtc.org/1717273002
Cr-Commit-Position: refs/heads/master@{#11733}
Also added a test for Clear to ensure this invariant holds.
With this change, it is easy to empty a Buffer and reuse its storage. Further down the line, code filling data into a Buffer could be written to just append to it, with the caller determining if the Buffer should first be cleared or not.
There is currently only one use of Buffer::Clear (in AudioEncoderCopyRed::Reset()) and it should benefit from the change, by not requiring a reallocation after Reset.
Review URL: https://codereview.webrtc.org/1707693002
Cr-Commit-Position: refs/heads/master@{#11680}
We'd like to completely replace rtc::scoped_ptr with std::unique_ptr.
This is a first trial CL to see if using unique_ptr causes any
problems.
(As a side effect of removing the scoped_ptr.h include in buffer.h,
I had to fix broken includes in no less than three files.)
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1687833006
Cr-Commit-Position: refs/heads/master@{#11588}
There's no reason not to use std::move instead now that we can use the
C++11 standard library.
BUG=webrtc:5373
Review URL: https://codereview.webrtc.org/1531013003
Cr-Commit-Position: refs/heads/master@{#11225}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
With it removed, you can now use it with scoped_refptr by wrapping it in
an rtc::RefCountedObject<rtc::Buffer>.
BUG=
Review URL: https://codereview.webrtc.org/1414053003
Cr-Commit-Position: refs/heads/master@{#10386}
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
(This was previously committed (9e1a6d7c) but had to be reverted
(cbf09274) because Chromium on Android wasn't quite ready for it).
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47109004
Cr-Commit-Position: refs/heads/master@{#9132}
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44269004
Cr-Commit-Position: refs/heads/master@{#9121}
We used to accept void* (until 9478437f), and we'll have to continue
to do so for a little while longer, until Chromium doesn't need it
anymore.
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48179004
Cr-Commit-Position: refs/heads/master@{#9035}
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}
And add a constructor for creating an uninitialized Buffer of a
specified size.
(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48579004
Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.
Not inlining virtual functions with simple bodies such as
{ return false; }
strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.
BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47429004
Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
