Reason for revert:
This CL caused a google3 breakage due to dependencies in Google3.
I will fix that, and reland.
Original issue's description:
> Moved ring-buffer related files from common_audio to audio_processing
>
> BUG=webrtc:5724
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/711ccc8d96490f58cc3d7fd9207c19d4d881d4dc
> Cr-Commit-Position: refs/heads/master@{#12227}
TBR=ivoc@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5724
Review URL: https://codereview.webrtc.org/1856323002
Cr-Commit-Position: refs/heads/master@{#12232}
- Makes vt h264 decoder output CoreVideoFrameBuffer
- Makes iOS renderer convert frame buffer if it is not i420
BUG=
Review URL: https://codereview.webrtc.org/1853503003
Cr-Commit-Position: refs/heads/master@{#12224}
Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
windows for going up quickly in the beginning of a call (before any
downscaling happens due to bad quality).
BUG=webrtc:5678
R=glaznev@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1830593003 .
Cr-Commit-Position: refs/heads/master@{#12219}
The change made in https://codereview.webrtc.org/1757683002 introduced an extra call to RTCPSender::SetRTCPStatus after the video receive stream is created. The SetRTCPStatus call results in no state change, as the RTCP sender is already enabled, however, it reschedules the next RTCP packet to be RTCP_INTERVAL_VIDEO_MS/2 (500) ms in the future.
Before the change, the next packet time was only set by the previous call to RTCPSender::SetSSRC, which placed it 100 ms in the future. The change, therefore, changed the timing of multiple performance tests - as it now takes a different length of time to ramp up to the same bandwidth.
BUG=chromium:597332
Review URL: https://codereview.webrtc.org/1826093004
Cr-Commit-Position: refs/heads/master@{#12203}
This makes the addaptation of the IntelligibilityEnhancer slower, which makes it take more time to kick in or when the background noise changes drastically. But on the other hand, it reduces the risk of clipping and makes the changing in coloring less noticeable.
R=henrik.lundin@webrtc.org, peah@webrtc.org, turaj@webrtc.org
Review URL: https://codereview.webrtc.org/1848123002 .
Cr-Commit-Position: refs/heads/master@{#12202}
VS 2015 has a new or louder warning about 32-bit shifts that are then
assigned to a 64-bit target. This type of code triggers it:
int64_t size = 1 << shift_amount;
Because the '1' being shifted is a 32-bit int the result of the shift
will be a 32-bit result, so assigning it to a 64-bit variable is just
misleading.
In this case the code that triggers it is this:
size_t window_size = static_cast<size_t>(1 << shift_amount);
The destination is a size_t so the warning only shows up on 64-bit
builds and doesn't indicate a real bug. It's curious that the code
had a cast already - presumably to suppress some other warning - but
the cast is not in the ideal place and doesn't avoid this new warning.
Moving the cast allows shift_amount to be log2(size_t) and allows
enabling C4334 in Chromium.
BUG=593448
Review URL: https://codereview.webrtc.org/1849753004
Cr-Commit-Position: refs/heads/master@{#12199}
Improved the existing external denoiser in WebRTC: the filter strength
is adaptive based on the noise level of the whole frame and the moving
object detection result. The adaptive filter effectively removes the
artifacts in previous version, such as trailing and blockiness on moving
objects.
The external denoiser is off by default for now.
BUG=
Review URL: https://codereview.webrtc.org/1822333003
Cr-Commit-Position: refs/heads/master@{#12198}
We can (and should) use std::vector<std::unique_ptr<T>> instead.
Because it's standard, and because it's safer since callers have to
manually wrap elements in std::unique_ptr before inserting them and
manually unwrap them after inserting them.
Review URL: https://codereview.webrtc.org/1839603002
Cr-Commit-Position: refs/heads/master@{#12182}
be similar to the other file names of the tests.
(Also removed a redundant blank line in the highpass filter unittest
file).
BUG=
Review URL: https://codereview.webrtc.org/1841363002
Cr-Commit-Position: refs/heads/master@{#12171}
Removes "SimulcastEncoderAdapter" from single-stream HW VP8 even though
they are wrapped in a SimulcastEncoderAdapter.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1827553002 .
Cr-Commit-Position: refs/heads/master@{#12161}
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.
BUG=webrtc:5155,webrtc:5670
Review URL: https://codereview.webrtc.org/1821083002
Cr-Commit-Position: refs/heads/master@{#12160}
Now that the Rent-A-Codec no longer owns the encoders and decoders it
produces, they may outlive it. It's thus no longer correct for the
Rent-A-Codec to own the bandwidth estimation state; all of the
involved objects need to share ownership.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1821513003
Cr-Commit-Position: refs/heads/master@{#12159}
- The interleaved_ field. Never set to anything but 'true'. AudioFrame data appears to always be treated as interleaved.
- The Append() method.
- operator-=().
BUG=
Review URL: https://codereview.webrtc.org/1830713003
Cr-Commit-Position: refs/heads/master@{#12152}
Revert reason: I unintentionally added a patch when rebasing that is breaking the bots.
This reverts commit 98c69a0ee785adeb9d95fffeb55cdb6cedbe82c6.
BUG=
Review URL: https://codereview.webrtc.org/1837313002 .
Cr-Commit-Position: refs/heads/master@{#12148}
Adds logging of:
- video stats that are recorded when a stream is removed
- bitrate stats that are recorded at the end of a call
- initial bwe rampup stats
BUG=
Review URL: https://codereview.webrtc.org/1788783002
Cr-Commit-Position: refs/heads/master@{#12133}
Reason for revert:
This CL are breaking some of the Android buildbots in Chromium.
The CL will need be be revised to exclude the Android platform.
Original issue's description:
> Added a bitexactness test for the gain controller in the audio processing module.
>
> BUG=webrtc:5339
>
> Committed: https://crrev.com/a49dc36976da44f3d6d75aed2dcab93fe14fc3a0
> Cr-Commit-Position: refs/heads/master@{#12124}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5339
Review URL: https://codereview.webrtc.org/1829323002
Cr-Commit-Position: refs/heads/master@{#12127}
