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Updating AsyncAudioProcessing API, part 1. Add an API to pass AudioFrameProcessor as a unique_ptr. Bug: webrtc:15111 Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984 Reviewed-by: Olga Sharonova <olka@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40187} 2 年前
Replace WebRTC-QuickPerfTest field trial with a flag This field trial is configured via command line flag, so may use flag system directly, reducing dependency on global field trial string. Bug: webrtc:7101, webrtc:10335 Change-Id: I1e48e0e3fdc251b73a375c6d7f1a46fa4f8a179b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322624 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40897} 2 年前
修复蓝牙录不到音以及接听系统电话导致crush问题 Signed-off-by: n037579 <shenglong.jing@archermind.com> 1 年前
Fix AudioMixer histogram test If the tests are run in a different order, the test might fail. We fix this by resetting the histogram data at the start of the test. Change-Id: I6fb349609842b55f416cf2ec8cd93d0b4328960e Bug: chromium:1430806 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323801 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Auto-Submit: Fredrik Hernqvist <fhernqvist@google.com> Cr-Commit-Position: refs/heads/main@{#40946} 2 年前
Make AEC3 json parsing code testonly Reasons: - the code is no longer used in Chrome - it is conceptually weird for WebRTC to have JSON parsing in its API - there are concerns around the reliability of the underlying JSON library Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly. Bug: webrtc:1493351 Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41014} 2 年前
Add 1minute as max hold duration to make sure that loss based BWE always tries to increase estimate. Bug: webrtc:12707 Change-Id: I94689431726a37e2bfec52992046305705c6bb8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324741 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Diep Bui <diepbp@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41025} 2 年前
Reduces rate at which TryGetNextFrame returns NULL for WGC This CL is a follow-up of work done in https://webrtc-review.googlesource.com/c/src/+/323882 where the goal was to reduce the amount of FrameDropped error logs in WebRTC.DesktopCapture.Win.WgcCaptureSessionGetFrameResult. The previous work avoids FrameDropped logs for a minimized window being captured with WGC but we still se a large amount of these error (or rather warning) logs. See [1] which comes from Canary. This CL does two different things to improve the situation: 1) It adds kFramePoolEmpty to the existing GetFrameResult::kFrameDropped enum to point out that the warning comes from the frame pool not being able to return a valid new frame. It also makes it more clear that it does not cause an outer/final error as WgcCapturerResult::kFrameDropped. We still keep the inner GetFrameResult::kFrameDropped but it is only produced when the frame pool returns NULL and our external queue is empty. Hence, a real frame-drop error. Note that, it is still easy to provoke kFramePoolEmpty simply by asking for a high resolution at a high rate. The example in [2] comes from a 4K screen @30fps. Hence, we have not fixed anything yet. 2) It also increases the size of the internal frame pool from 1 to 2. This does lead to an almost zero rate of kFramePoolEmpt warnings at the expense of a slightly reduced max capture rate. BUT, with 1 as size, we can "see" a higher max capture rate but it is not a true rate since it comes with a high rate of kFramePoolEmpty errors. Hence, we "emulate" a high capture rate by simply feeding copies of the last frame that we had stored in the external queue. Using 2 leads to a more "true" rate of what we actually can capture in terms of *new* frames and also a substantially lower rate of kFramePoolEmpty. In addition, with 1 as size, if we ask at a too high rate and provide a copy of the last frame, our CPU adaptation will not reduce its rate since we think that things are OK when it is actually not. Also, the samples in [3] and [4] both use 2 as numberOfBuffers as well. Let me also mention that with this small change, I a have not been able to provoke any kFramePoolEmpty error messages. Finally, geDisplayMedia can be called called with constraints where min and max framerate is defined. The mechanism which maintains the min rate is implemented via the RequestRefreshFrame API and it can be sent to the source (DesktopCaptureDevice) back to back with a previous timer interrupt for a capture request. Without this change, these RRFs were also a source of a large amount of kFramePoolEmpty error logs. With 2 buffers instead; this is no longer the case. [1] https://screenshot.googleplex.com/7sfv6HdGXLwyxdj [2] https://paste.googleplex.com/4795680001359872 [3] https://github.com/robmikh/Win32CaptureSample/blob/master/Win32CaptureSample/SimpleCapture.cpp [4] https://learn.microsoft.com/en-us/windows/uwp/audio-video-camera/screen-capture#add-the-screen-capture-capability (cherry picked from commit 4be5927dc788b814dff2a6ca46175254dce59232) Bug: chromium:1314868 Change-Id: I73b823b31a993fd2cd6e007b212826dfe1a80012 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325521 Commit-Queue: Alexander Cooper <alcooper@chromium.org> Reviewed-by: Alexander Cooper <alcooper@chromium.org> Cr-Original-Commit-Position: refs/heads/main@{#41079} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326960 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/branch-heads/6099@{#1} Cr-Branched-From: 507f1cc3270d0577f79882acbd78e63e66008f3d-refs/heads/main@{#41042} 2 年前
[Unwrap] Delete webrtc::Unwrapper Bug: webrtc:13982 Change-Id: I501261b09a05080ec681ae120648938e350a05de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290890 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39088} 3 年前
Throttle 'Very high pacing rate' log message By producing new message only when new max is 10% larger than the previous max. Bug: b/305042040 Change-Id: Id85784939f944de8115b881471b02214c34b3043 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323841 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40950} 2 年前
Use a constant for invalid PipeWire file descriptor We use value -1 on over all the places through our code so it might be better to define a constant and use it instead to make the code more understandable on first look. Bug: webrtc:15203 Change-Id: I4fc3e561bc7a7778c43ec6cfde7acebef2af79e8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306620 Reviewed-by: Alexander Cooper <alcooper@chromium.org> Commit-Queue: Jan Grulich <grulja@gmail.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40156} 2 年前
Subtract an additional 5kbps of the bitrate when backing off. Traditionally, we'd back off to 85% of the measured throughput in response to an overuse. However, this backoff doesn't appear to be sufficient to drain the queues in some low-bitrate scenarios, and the problem has gotten a bit worse with the RobustThroughputEstimator. (The new estimate looks more stable. The old estimator had more variation, the lowest points were lower, causing backoffs to lower rates.) With this change, we back off to 0.85*thoughput-5kbps. The difference is negligible except at low bitrates. Bug: webrtc:13402,b/298636540 Change-Id: I53328953c056b8ad77f6c7561d6017f171b2dfbc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321701 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40827} 2 年前
Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023} 2 年前
[ssci] Added Shipped field to READMEs This CL adds the Shipped field (and may update the License File field) in Chromium READMEs. Changes were automatically created, so if you disagree with any of them (e.g. a package is used only for testing purposes and is not shipped), comment the suggested change and why. See the LSC doc at go/lsc-chrome-metadata. Bug: b:285450740 Change-Id: If4955c6f6e7b58e0c99469fc45ed5b9e8f30a32b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311720 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Auto-Submit: Anne Redulla <aredulla@google.com> Cr-Commit-Position: refs/heads/main@{#40424} 2 年前
Rland "Revert "Reland "Reland "Delete old Android ADM."""" This reverts commit 7534ebd2bf59212cce5e010dd6ed9b3bc944818e. Reason for revert: Downstream projects have been updated, try it again. R=perkj@webrtc.org Bug: webrtc:7452 Change-Id: Ice48a563a6da499b6050b6f6e21bb0a18cd34f57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271841 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40386} 2 年前
添加鸿蒙适配脚本以及部分缺少的模块 Signed-off-by: zhongluping <278527840@qq.com> 2 年前
适配OH平台VP8/VP9/AV1软件编码器的线程数 Signed-off-by: chenhao <haoe.chen@archermind.com> 1 年前
[WebRTC-SendPacketsOnWorkerThread] Delete MaybeWorkerThread This helper class is no longer used. Bug: webrtc:14502 Change-Id: I7940de762ebb9a7c6d04927603f249f5b0061051 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301161 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39880} 3 年前
[Unwrap] Delete webrtc::Unwrapper Bug: webrtc:13982 Change-Id: I501261b09a05080ec681ae120648938e350a05de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290890 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39088} 3 年前