In the case of H264 we can't know which packet that is the fist packet of a
frame. In order to avoid creating incomplete frames we keep track of which
packets that we haven't received, and if there is a gap in the packet sequence
number leading up to this frame then a frame wont be created.
BUG=chromium:716558
Review-Url: https://codereview.webrtc.org/2926083002
Cr-Commit-Position: refs/heads/master@{#18559}
Track UIApplication applicationState changes from a C++ class. Uses
NSNotificationCenter to access changes on the main thread and exposes
a local variable that can be checked from any thread.
This fixes a runtime warning on iOS 11 beta.
My Objective-C++ is a little rusty so please check if this follows
the conventions for C++ code in the project. It also changes the
interface exposed by RTCUIApplication.h, not sure if that has impact
on any public APIs that needs to be documented somewhere?
Bug: webrtc:7773
Change-Id: I9c8ba090ef9f28d812114026a906cef742192c39
Reviewed-on: https://chromium-review.googlesource.com/527442
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Kári Tristan Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18558}
This reverts commit b008b45f1e609556a04c1aabb4e8ed6a894265af.
Reason for revert: Breaks external clients.
Original change's description:
> Update webrtc/sdk/objc to new VideoFrameBuffer interface
>
> More thorough refactoring work is planned for RTCVideoFrame (see webrtc:7785), and this CL just unblocks removing the old interface from webrtc::VideoFrameBuffer.
>
> Bug: webrtc:7632,webrtc:7785
> Change-Id: I351536c5ca454c2acd8944bbc2ebb1d1439dc50c
> Reviewed-on: https://chromium-review.googlesource.com/530231
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18553}
TBR=magjed@webrtc.org,andersc@webrtc.org
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7632,webrtc:7785
Change-Id: Ib5c6fcb939175c67c3ac7b3df7cea0f7c2bb0af0
Reviewed-on: https://chromium-review.googlesource.com/533013
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18557}
More thorough refactoring work is planned for RTCVideoFrame (see webrtc:7785), and this CL just unblocks removing the old interface from webrtc::VideoFrameBuffer.
Bug: webrtc:7632,webrtc:7785
Change-Id: I351536c5ca454c2acd8944bbc2ebb1d1439dc50c
Reviewed-on: https://chromium-review.googlesource.com/530231
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18553}
All of the buffer size returned by Windows Core Audio APIs are in unit
of audio frames (which is sample times number of channels), while
WebRTC's AudioDeviceBuffer RequestPlayoutData method takes in samples
per channel (equivalent to frames per channel) but returns number of
audio samples in all the channels. This CL makes sure that we compare
playout block size in frames with frames and size in samples with
samples, which should fix the excessive logging issues and audio quality
problems due to the mismatch when comparing.
BUG=webrtc:7797
Review-Url: https://codereview.webrtc.org/2933953003
Cr-Commit-Position: refs/heads/master@{#18546}
Since this is a test for a fake network, it's only natural that it uses
a fake clock as well. This makes the tests much faster, less flaky, and
lets them be moved out of "webrtc_nonparallel_tests", since they no
longer have a dependency on any "real" thing (sockets, or time) and
can be run in parallel as easily as any other tests.
As part of this CL, added the fake clock as an argument to
VirtualSocketServer's and TestClient's constructors, since these classes
have methods that wait synchronously for something to occur, and if the
test is using a fake clock, they need to advance it in order to make
progress.
Lastly, added a DCHECK in Thread::ProcessMessages. If called with a
nonzero time while a fake clock is used, it will get stuck in an
infinite loop; a DCHECK is easier to notice than an infinite loop.
BUG=webrtc:7727, webrtc:2409
Review-Url: https://codereview.webrtc.org/2927413002
Cr-Commit-Position: refs/heads/master@{#18544}
const int16_t* data() const;
int16_t* mutable_data();
- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.
These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.
This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.
BUG=webrtc:7343
TBR=henrika
Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
Specifically, just like SafeMin() and SafeMax() it handles all
combinations of integer and all
combinations of floating-point arguments by picking a
result type that is guaranteed to be able to hold the result.
This CL also replaces a bunch of std::min + std:max call pairs with
calls to SafeClamp()---the ones that could easily be found by grep
because "min" and "max" were on the same line. :-)
BUG=webrtc:7459
Review-Url: https://codereview.webrtc.org/2808513003
Cr-Commit-Position: refs/heads/master@{#18542}
This allows to protect ssrc_to_last_received_extended_high_seq_num_ member and
make calls to OnReceivedRtcpReceiverReport thread-safe without introducing new critical section.
Bug: webrtc:7735
Change-Id: Iee23bb780d07b0f906f1f8eeddde2b74cc0a2b89
Reviewed-on: https://chromium-review.googlesource.com/518130
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18540}
This is to make it possible to override the rtc_task_queue target only.
BUG=none
Review-Url: https://codereview.webrtc.org/2931273002
Cr-Commit-Position: refs/heads/master@{#18534}
For devices with multiple cameras, all supported resolutions from both
the front-facing and back cameras are listed.
Bug: webrtc:7783
Change-Id: I228eda28ea48181c86d344413dda9f3a71b0864f
Reviewed-on: https://chromium-review.googlesource.com/529045
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18533}
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.
BUG=None
Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
After landing https://chromium-review.googlesource.com/528173
only one test needs to be disabled: VirtualSocketServerTest.delay_v4
BUG=webrtc:7727
NOTRY=True
TESTED=gn gen out/x64-Debug --args='target_os="ios" ios_enable_code_signing=false is_component_build=false target_cpu="x64"'
ninja -C out/x64-Debug webrtc_nonparallel_tests
out/x64-Debug/iossim -d "iPhone 6s" -s 10.3 out/x64-Debug/webrtc_nonparallel_tests.app
Review-Url: https://codereview.webrtc.org/2909073002
Cr-Commit-Position: refs/heads/master@{#18519}
The default implementations of AudioProcessing::{AttachAecDump,
DetachAecDump} are removed and audio_processing.cc is decoupled from
aec_dump.h. After this CL, the two methods are pure virtual. The
default implementations were added because doing otherwise would break
internal projects.
Bug: webrtc:7404
Change-Id: If94f60aeefe4ad1eefed3744f857692cc645bdf4
Reviewed-on: https://chromium-review.googlesource.com/528132
Commit-Queue: Alex Loiko <aleloi@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18517}
When using the SetEncoder interface, there's no actual CodecInst to return from Channel::GetSendCodec. Before this CL, this was done by calling the ACM, which has functionality for generating a CodecInst with the necessary values even when handed an external encoder. Unfortunately, this call takes a lock and does some extra processing which isn't strictly necessary in this case. Since GetSendCodec is called inside the audio input callback code, this can cause problems.
This CL instead generates a CodecInst in the SetEncoder call and has GetSendCodec simply return that one if it's available. If it isn't the value from codec_manager_ is returned instead, as was the case before injectable audio codec related changes were added to Channel.
BUG=b/38018041
Review-Url: https://codereview.webrtc.org/2924363004
Cr-Commit-Position: refs/heads/master@{#18515}
This decoder is only used for iOS/Mac and it will simplify to make it
ObjC++ instead of C++, similar to how the encoder is .mm already.
Bug: None
Change-Id: I13f62f018432e9c23e7277eea29258a73e1590e1
Reviewed-on: https://chromium-review.googlesource.com/529084
Reviewed-by: Kári Tristan Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18513}
This solves an issue where if the first packet happens to be an RTX packet, it is recovered with an incorrect SSRC.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2928123002
Cr-Commit-Position: refs/heads/master@{#18512}
Make sure to call ensureGLContext before calling OpenGL functions
BUG=webrtc:7751
Review-Url: https://codereview.webrtc.org/2916583005
Cr-Commit-Position: refs/heads/master@{#18511}
This class is video-specific, and we want to free the name
"RtpStreamReceiver" so it can be reused for a media-independent RTP
receive class.
Also renames related files.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2926253002
Cr-Commit-Position: refs/heads/master@{#18510}
