When the iOS application is not in the foreground, the hardware encoder and
decoder become invalidated. There doesn't seem to be a way to query their state
so we don't know they're invalid until we get an error code after an
encode/decode request. To solve the issue, we just don't encode/decode when the
app is not active, and reinitialize the encoder/decoder when the app is active
again.
Also fixes a leak in the decoder.
BUG=webrtc:4081
Review URL: https://codereview.webrtc.org/1732953003
Cr-Commit-Position: refs/heads/master@{#11916}
initialization and errors.
The stats are counts using enumeration, an instance of
H264EncoderImpl/H264DecoderImpl will report at most 1 Init
and 1 Error for its entire lifetime. This is to avoid
spamming reports if initialization or coding fails and it
retries in a loop. The Init stats will give us an idea of
usage counts for the encoder/decoder. The Error stats will
give us an idea of how many of these usages encounters some
type of problem, such as encode or decode errors.
- WebRTC.Video.H264EncoderImpl.Event:
* kH264EncoderEventInit: Occurs at InitEncode.
* kH264EncoderEventError: Occurs if any type of error
occurs during initialization or encoding.
- WebRTC.Video.H264DecoderImpl.Event:
* kH264DecoderEventInit: Occurs at InitDecode.
* kH264DecoderEventError: Occurs if any type of error
occurs during initialization, AVGetBuffer2 or decoding.
Chromium sibling CL:
https://codereview.chromium.org/1719273002/
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1716173002
Cr-Commit-Position: refs/heads/master@{#11736}
The legacy objc API is not included in the GYP generation if include_tests=0.
This causes problems downstream in some cases, so it's changed in this CL.
The libyuv dependency needs to be possible to disable using the build_libyuv
GYP variable.
NOTRY=True
Review URL: https://codereview.webrtc.org/1705733002
Cr-Commit-Position: refs/heads/master@{#11652}
Until the bug has been further investigated, we're limiting the number
of threads to 1 to avoid problems. See crbug.com/583348.
BUG=chromium:500605, chromium:468365, chromium:583348
Review URL: https://codereview.webrtc.org/1677543002
Cr-Commit-Position: refs/heads/master@{#11536}
There were a couple of GN and GYP references that were incorrect in Chromium builds:
- GN references between WebRTC targets must be using relative paths, not absolute.
- GYP references between WebRTC targets must be using the <(webrtc_root)v variable
in order to be expanded to the correct path in a Chromium build.
NOTRY=True
TBR=hjon@webrtc.org, hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1681493002
Cr-Commit-Position: refs/heads/master@{#11521}
Renamed the WEBRTC_THIRD_PARTY_H264 macro to WEBRTC_USE_H264 to match flag name.
The idea is to be able to turn off H264 from chromium with this function because...
1) The Chromium trybots will soon use this flag, we want to temporarily disable H264 from chromium even if flag is set in case something is broken. That way when we are ready to flip the switch the trybots will run our test code then and not after it is already enabled.
2) If feature is launched and we discover major problems we can easily disable H264 and merge with beta/stable.
3) Or, if feature is behind a *runtime* flag, this is how we would control if it is used or not.
The idea is to call DisableRtcUseH264 in chromium's PeerConnectionDependencyFactory.
BUG=chromium:500605, chromium:468365
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1657273002
Cr-Commit-Position: refs/heads/master@{#11474}
New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium.
In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default.
Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg.
BUG=chromium:500605, chromium:468365, webrtc:5427
Review URL: https://codereview.webrtc.org/1639273002
Cr-Commit-Position: refs/heads/master@{#11456}
It works on all platforms except Android and iOS (FFmpeg limitation).
Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.
Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)
Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)
NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
Review URL: https://codereview.webrtc.org/1306813009
Cr-Commit-Position: refs/heads/master@{#11390}
The FFmpeg video decoder requires up to 8 additional bytes to be allocated for its encoded image buffer input, due to optimized byte readers over-reading on some platforms.
We plan to use FFmpeg for a soon-to-land H.264 enc/dec.
This CL adds support for padding encoded image buffers based on codec type, and makes sure calls to VCMEncodedFrame::VerifyAndAllocate use the padding.
All padding constants are 0 but making H.264 pad with 8 bytes will be a one-line change.
Also, added -framework CoreFoundation to webrtc_h264_video_toolbox which was missing.
BUG=chromium:468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
NOTRY=True
Review URL: https://codereview.webrtc.org/1602523004
Cr-Commit-Position: refs/heads/master@{#11337}
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1418913006
Cr-Commit-Position: refs/heads/master@{#10659}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
VideoFrameBuffer currently has two overloaded data() functions for pixel access, one for const and one for non-const. Unfortunately, it will default to the non-const version, even when 'const scoped_refptr<VideoFrameBuffer>&' is used. This is a problem, because many subclasses use RTC_NOTREACHED() in the non-const version.
This CL makes the non-const version of data() explicit with a different, longer function name MutableData().
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1304143003 .
Cr-Commit-Position: refs/heads/master@{#9787}
