Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
Add seek methods to FileWrapper, and refactor WavReader to use that
class instead.
Bug: webrtc:6463
Change-Id: Ifbb1989a072da6280ea5fc04b4beff991614dd53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147265
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28770}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
After being stuck "forever" (3 seconds) waiting for an event to
trigger, log the stack trace of the current thread to aid debugging of
the deadlock.
Bug: webrtc:10308
Change-Id: I04852f191027294d7e7a7f5e63de4c6c7fdd6326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128342
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27263}
The video decoder thread is the pilot user.
For now this is an Android-only feature, since that's the only
platform we can print stack traces on.
Bug: webrtc:9987
Change-Id: Ie638c619673b5f159d91a32683fd787baf46479a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126222
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27127}
https://webrtc-review.googlesource.com/c/src/+/105301 remove the dependency to rtc_base_generic, it also removed the dependnecy to Foundation.framework. This CL adds it back.
Bug: webrtc:9838
Change-Id: I861e73d13eb36d2c3a09d998a6def9512066f0d5
Reviewed-on: https://webrtc-review.googlesource.com/c/122621
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#26654}
This file has been causing problems for the build. ObjC was required for
a few methods because autoreleasepools are necessary on new threads if
those threads will be running objc code.
This CL introduces a workaround by using ObjC runtime C APIs to create
and drain autoreleasepools, but this comes with the cost of relying on
an internal API that may break on future OS/clang releases.
Bug: webrtc:9838
Change-Id: I18e765020c20c096c9ef8d80dfa82375ecb202ff
Reviewed-on: https://webrtc-review.googlesource.com/c/105301
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25141}
This is a reland of 55daf1aef65218a97eff88999e5190a2f2f6b72e.
In order to avoid problems on case insensitive file systems this CL
moves rtc_export.h to rtc_base/system (avoiding problems with build/).
Diff: https://webrtc-review.googlesource.com/c/src/+/100804/1..2.
Original change's description:
> Add RTC_EXPORT macro to export WebRTC symbols.
>
> This CL introduces the utility macro RTC_EXPORT which will let WebRTC
> developers decide which symbols are supposed to be exported/imported
> and which ones are private.
>
> RTC_EXPORT will only export/import symbols in a component build, more
> info: https://cs.chromium.org/chromium/src/docs/component_build.md.
> During a component build, the macro COMPONENT_BUILD will be globally
> defined in a consistent fashion so it is safe to rely on it to
> understand how to expand RTC_EXPORT.
> In a non component build, RTC_EXPORT will expand to nothing.
>
> Bug: webrtc:9419
> Change-Id: Ic58162783be7f5883136ade27f324d6d34fdf932
> Reviewed-on: https://webrtc-review.googlesource.com/97960
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24757}
Bug: webrtc:9419
Change-Id: Icfedea5fc3416ea1af2185de443fa879fb2dee8b
Reviewed-on: https://webrtc-review.googlesource.com/100804
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24766}
Instead of making multiple calls to the std::stringstream << operator,
collect all the arguments and make a single printf-like variadic call
under the hood.
Besides reducing our reliance on iostreams, this makes each RTC_LOG_*
call site smaller; in aggregate, this reduces the size of
libjingle_peerconnection_so.so by 28-32 kB.
A quick benchmark indicates that this change makes log statements
a few percent slower.
Bug: webrtc:8982, webrtc:9185
Change-Id: I3137a4dd8ac510e8d910acccb0c97ce4fffb61c9
Reviewed-on: https://webrtc-review.googlesource.com/75440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23375}
When the logging severity is statically known, passing it as a
template argument instead of as a function argument saves space at the
call site.
In aggregate, this reduces the size of libjingle_peerconnection_so.so
by 8 kB.
Bug: webrtc:9185
Change-Id: I9ca363845216370e97b230952c86e6d07719962f
Reviewed-on: https://webrtc-review.googlesource.com/74480
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23121}
The audio processing code is using parts of the iSAC codec to do voice
activity detection (VAD), but it's undesirable for it to pull in the
entire iSAC codec as a dependency. So this CL factors out the parts of
iSAC that's needed for VAD to a separate build target.
Bug: webrtc:8396
Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e
Reviewed-on: https://webrtc-review.googlesource.com/69640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23110}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
Bug: webrtc:8445
NOPRESUBMIT=true
Change-Id: I30d01fcb9cbe1427a7703a3cdd7befae751066b5
Reviewed-on: https://webrtc-review.googlesource.com/21982
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22550}
