Also added some more logging, to help track down start/stop, start
failure, and the name of the file used.
BUG=
Review-Url: https://codereview.webrtc.org/2253763002
Cr-Commit-Position: refs/heads/master@{#13802}
The functions in question were virtual, so we would've wanted to get
rid of the default values even if callers had relied on them.
Review-Url: https://codereview.webrtc.org/2045943004
Cr-Commit-Position: refs/heads/master@{#13800}
Trivial patch which avoids logs that are of no value.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2250403002
Cr-Commit-Position: refs/heads/master@{#13799}
Reason for revert:
Breaks downstream code.
Original issue's description:
> StartTimestamp generated randomly in RtpSender constructor
> instead of not-randomly at SetSendingState(true)
> Renamed to timestamp_offset_ to better match meaning of the variable.
>
> R=asapersson@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> Cr-Commit-Position: refs/heads/master@{#13796}
TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2248413002
Cr-Commit-Position: refs/heads/master@{#13798}
Because passing ownership in raw pointers makes kittens cry.
This also means we can ditch the Destroy functions and the protected
destructors. (Well, almost. We need to keep the old CreateFilePlayer
and DestroyFilePlayer around for a little while longer because of an
external caller.)
Review-Url: https://codereview.webrtc.org/2049683003
Cr-Commit-Position: refs/heads/master@{#13797}
They are implementations of interfaces that are only ever exposed
via "create" functions, so the entire class definitions can be put in
anonymous namespaces in the .cc files that defines the "create"
functions.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2038513002
Cr-Commit-Position: refs/heads/master@{#13794}
Reland of https://codereview.webrtc.org/2072753002/ which broke
chromium due to how their build was setup. This issue should now be
resolved.
Changed WebRtcVoiceEngine to present receive codecs from the formats
provided by its decoder factory. Added supported formats to
BuiltinAudioDecoderFactory. Added helper functions for creating some
simple decoder factories for mocking.
Created a PayloadTypeMapper for assigning payload types to formats. I
think we'll eventually want to use this further up, or possibly in
both the audio and video sides. It would be best if the engines didn't
have to talk payload types at all, but it might be more difficult to
get around when payload types depend on each-other, like the RTX
codecs for video.
BUG=webrtc:5805
TBR=ivoc@webrtc.org
Review-Url: https://codereview.webrtc.org/2250683002
Cr-Commit-Position: refs/heads/master@{#13793}
Conceptually, dummy audio file devices are a "platform", like
win/mac/linux, and so the conditional slots under
include_internal_audio_device. When enabled, use_dummy_audio_file_devices
disables whatever platform-specific audio layer would have been used and
turns on dummy file device support.
BUG=
Review-Url: https://codereview.webrtc.org/2250483002
Cr-Commit-Position: refs/heads/master@{#13790}
implementation can accurately capture updated regions. Especially in
ScreenCapturerWinDirectx, which has a specific updated region spreading logic
and cannot be tested through regular code path. So we need a controllable
ScreenDrawer to draw some basic shapes on the screen. And a platform independent
test case can use the ScreenDrawer to test a ScreenCapturer.
So this change addes a ScreenDrawer virtual class, and its Windows
implementation ScreenDrawerWin. A disabled gtest ScreenDrawerTest.DrawRectangles
is also added to manually test whether ScreenDrawer can work on a certain
platform.
BUG=314516
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2210443002
Cr-Commit-Position: refs/heads/master@{#13788}
This CL does literally nothing else but run "git cl format --full"
on the touched files.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2035663002
Cr-Commit-Position: refs/heads/master@{#13782}
It was disabled for some reason, even though in GYP it's enabled.
BUG=626067
NOTRY=True
Review-Url: https://codereview.webrtc.org/2247293002
Cr-Commit-Position: refs/heads/master@{#13780}
Reason for revert:
Reverting, because it turns out that third-party code was using webrtc::FilePlayer. I'm not at all sure that this is something WebRTC ought to be exporting, but since we did export it, we have to live with it for now.
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This has been landed twice before, as
> https://codereview.webrtc.org/2037623002 and
> https://codereview.webrtc.org/2240163002. Third time's a charm!)
>
> NOPRESUBMIT=True
> TBR=kjellander@webrtc.org
>
> Committed: https://crrev.com/427ce3d86f6328dc994f84a15c28bb7bfbaa46ef
> Cr-Commit-Position: refs/heads/master@{#13777}
TBR=
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2245413002
Cr-Commit-Position: refs/heads/master@{#13779}
-Removed the old probe cluster logic and use the new ProbeBitrateEstimator
instead.
-Removed all logic related to ssrcs from DelayBasedBwe as they have no function
on the sender side.
BUG=webrtc:5859
R=stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2234363002 .
Cr-Commit-Position: refs/heads/master@{#13771}
This is in preparation for adding a gn target for audio_device_tests.
BUG=webrtc:6170,webrtc:163
NOTRY=True
Review-Url: https://codereview.webrtc.org/2222563002
Cr-Commit-Position: refs/heads/master@{#13768}
A recent DrMemory failure has been detected after change 2099123002. After some
investigation, an uninitialized read has been detected in
NtUserGetThreadDesktop
webrtc::Desktop::GetThreadDesktop
webrtc::ScopedThreadDesktop::ScopedThreadDesktop
webrtc::ScreenCapturerWinGdi::ScreenCapturerWinGdi
webrtc::ScreenCapturer::Create
webrtc::ScreenCapturerTest_UseDirectxCapturer_Test::TestBody
So there are two issues,
1. The Directx capturer won't be triggered as the system does not support it. So
these tests should be disabled in this scenario.
2. An uninitialized read in NtUserGetThreadDesktop -> ScopedThreadDesktop
stacks, which should be suppressed. By default, these suppressions should be
placed in chromium/external with other suppressions.
So this change is a quick fix to the failure, do not use ScreenCapturerWinGdi in
ScreenCaputrerWinDirectx tests.
BUG=
Review-Url: https://codereview.webrtc.org/2247943002
Cr-Commit-Position: refs/heads/master@{#13766}
H.264 frames may have AUD before SPS. This change detects AUD and try to extract SPS and PPS behind AUD.
BUG=webrtc:6173
Review-Url: https://codereview.webrtc.org/2209143002
Cr-Commit-Position: refs/heads/master@{#13765}
When playing out, for example, you'd see 3 lines for every call to
PlayoutDelay, which happens quite often (every sample?).
The ones around the Playout/Recording Warning/Error are only once a
second, but they don't seem to add anything. Same with
Process/TimeUntilNextProcess, which just log that the method is called.
BUG=
Review-Url: https://codereview.webrtc.org/2202243004
Cr-Commit-Position: refs/heads/master@{#13763}
when building with default warnings.
This is in preparation for making a gn target for audio_device_tests.
BUG=webrtc:6170, webrtc:163
NOTRY=True
Review-Url: https://codereview.webrtc.org/2219653004
Cr-Commit-Position: refs/heads/master@{#13759}
Reason for revert:
Breaks downstream code, so revert again. Yay.
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}
TBR=perkj@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}
Also fixed one small chromium-style error in the new mixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2234293002
Cr-Commit-Position: refs/heads/master@{#13752}
The last in-tree call site recently disappeared, so they were unused.
BUG=webrtc:5922
Review-Url: https://codereview.webrtc.org/2066473002
Cr-Commit-Position: refs/heads/master@{#13751}
The folder structure is now as was agreed on in the 'Slim and Modular
WebRTC' effort. Also added some dependencies that were previously in
another part of the tree.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2238803002
Cr-Commit-Position: refs/heads/master@{#13742}
Since all FrameObjects have a reference to its PacketBuffer and since
the PacketBuffer can be thrown away at any moment the PacketBuffer
has to be ref counted in order to avoid FrameObjects dereferencing a potentially
destroyed object.
BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2199133004 .
Cr-Commit-Position: refs/heads/master@{#13725}
Since the interval between the timestamps does not include the send/receive
time of the last/first packet we correct the interval by adding the average
of the interval between probing packets.
BUG=webrtc:5859
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2224173003 .
Cr-Commit-Position: refs/heads/master@{#13719}
- Make more use of std::unique_ptr.
- Auto type deduction for iterator type names.
- More extensive comments.
- Variable renaming.
- Make ProducerFec::BuildRedPacket() static.
- Avoid dynamic allocation of ProducerFec::fec_.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2110763002
Cr-Commit-Position: refs/heads/master@{#13700}
