2e1f16d5创建于 2023年10月26日历史提交
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Avoiding to increase an iterator when the result can be larger than their container end. Bug: webrtc:15438 Change-Id: I0d75436bc845590c76466bde7007e921f842a9d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317320 Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40605} 2 年前
APM: add AudioProcessingImpl::capture_::applied_input_volume(_changed) The recommended_stream_analog_level() getter is used to retrieve both the applied and the recommended input volume. This behavior is error-prone since the caller must know what is returned based on the point in the code (namely, before/after the AGC has changed the last applied input volume into a recommended level). This CL is a first step to make clarity on which input volume is handled in different parts of APM. Next in the pipeline: make recommended_stream_analog_level() a trivial getter that always returns the recommended level. Main changes: - When recommended_stream_analog_level() is called but set_stream_analog_level() is not called, APM logs an error and returns a fall-back volume (which should not be applied since, when set_stream_analog_level() is not called, no external input volume is expected to be present - When APM is used without calling the *_stream_analog_level() methods (e.g., when the caller does not provide any input volume), the recorded AEC dumps won't store Stream::applied_input_level Other changes: - Removed AudioProcessingImpl::capture_::prev_analog_mic_level - Removed redundant code in GainController2 around detecting input volume changes (already done by APM) - Adapted the audioproc_f and unpack_aecdump tools - Data dumps clean-up: the applied and the recommended input volumes are now recorded in an AGC implementation agnostic way Bug: webrtc:7494, b/241923537 Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38054} 3 年前
Format /modules git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^modules/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jared Siskin <jtsiskin@meta.com> Cr-Commit-Position: refs/heads/main@{#39901} 3 年前
Format /modules git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^modules/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jared Siskin <jtsiskin@meta.com> Cr-Commit-Position: refs/heads/main@{#39901} 3 年前
Use sinf instead of std::sinf to improve libstdc++ compatibility libstdc++ does not define std::sinf in <cmath>. See also: https://stackoverflow.com/a/56420862. BUG=b:235200394 Change-Id: Idfb80ac6f54fbf57a20425391b0c4165b7945b2f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306681 Commit-Queue: Li-Yu Yu <aaronyu@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40183} 2 年前
Add refined handling of the internal scaling of the audio in APM This CL adds functionality that allows adjusting the audio levels internally in APM. The main purpose of the functionality is to allow APM to optionally be moved to an integration that does not provide an analog gain to control, and the implementation of this has been tailored specifically to meet the requirements for that. More specifically, this CL does -Add a new variant of the pre-amplifier gain that is intended to replace the pre-amplifier gain (but at the moment can coexist with that). The main differences with the pre-amplifier gain is that an attenuating gain is allowed, the gain is applied jointly with any emulated analog gain, and that its packaging fits better with the post gain. -Add an emulation of an analog microphone gain. The emulation is designed to match the analog mic gain functionality in Chrome OS (which is digital) but should be usable also on other platforms. -Add a post-gain which is applied after all processing has been applied. The purpose of this gain is for it to work well with the integration in ChromeOS, and be used to compensate for the offset that there is applied on some USB audio devices. Bug: b/177830918 Change-Id: I0f312996e4088c9bd242a713a703eaaeb17f188a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209707 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33466} 5 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Reland "Migrate WebRTC documentation to new renderer" This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1. Reason for revert: Downstream infrastructure should be ready now Original change's description: > Revert "Migrate WebRTC documentation to new renderer" > > This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. > > Reason for revert: > > Original change's description: > > Migrate WebRTC documentation to new renderer > > > > Bug: b/258408932 > > Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39205} > > Bug: b/258408932 > Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39209} Bug: b/258408932 Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660 Owners-Override: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39231} 3 年前
AGC2 adaptive digital controller config clean-up - Remove dry-run option - Hard-code adjacent_speech_frames_threshold and vad_reset_period_ms - Expose initial_gain_db via field trial Tested: adaptive digital controller bit-exactness verified Bug: webrtc:7494 Change-Id: I6166611f91320b6c37de3f8e553c06c2ed95b772 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287222 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38862} 3 年前
Fix improper buffer size in call to rtc::strcpyn rtc::strcpyn second param should be the size of the destination buffer, not the size of the source string. The result is that the final character (usually a trailing directory path separator) is lost during the copy. This has been masked because FormFileName helpfully adds a trailing path separator if one is missing. BUG=webrtc:15441 Change-Id: I992e69cad86a7e8bc2057ec629063f34c75fe75f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317502 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40736} 2 年前
Format /modules git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^modules/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jared Siskin <jtsiskin@meta.com> Cr-Commit-Position: refs/heads/main@{#39901} 3 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
Fix missing libc++ includes in webrtc Several files refer to symbols declared in headers not explicitly included. This compiles now because libc++ tranitively includes these headers via other libc++ headers; however, these transitive includes are not guaranteed to exist and in Chrome, will no longer exist once libc++ is compiled with modules. Bug: chromium:543704 Change-Id: I638bb02df3d050a48345248e80aebd2dd60956c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295762 Auto-Submit: Alan Zhao <ayzhao@google.com> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39448} 3 年前
Fix math involving enums in C++20 (-Wdeprecated-anon-enum-enum-conversion) - Replace enum with constexpr if necessary. - Merge multiple definitions for H.264 NalDefs and FuDefs and apply constexpr. Bug: chromium:1284275 Change-Id: I4a4d95ed6aba258e7c19c3ae6251c8b78caf84ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276561 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#38215} 3 年前
Fix downstream review comments for C++20 This CL addresses the review comments for https://webrtc-review.googlesource.com/c/src/+/261221 in the downstream cherry-pick: https://crrev.com/c/4660950. * Always use size_t{} for casting. * Remove unneeded size_t casts. * Avoid using __x as it is reserved for the compiler. Bug: b:217226507 Change-Id: I13c57cb69d7db066ac9a6dbd15b7f6de54abb613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311360 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Li-Yu Yu <aaronyu@google.com> Cr-Commit-Position: refs/heads/main@{#40395} 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
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}8 年前
Update some audio modules with new OWNERS Bug: b/260832909 Change-Id: I3d2ebad978988eabf228475c3fc46708e12cf5d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285780 Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Christoffer Jansson <jansson@webrtc.org> Commit-Queue: Christoffer Jansson <jansson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38788} 3 年前
AudioBuffer: Remove deprecated constructor Removes webrtc::AudioBuffer deprecated constructor. Bug: webrtc:5298 Change-Id: If8debe990c0358e58fbdc59bfdfb7e4af1f38f00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257305 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36518} 4 年前
APM: Signal error on unsupported sample rates This CL adds more explicit tests for unsupported sample rates in the WebRTC audio processing module (APM). Rates are restricted to the range [8000, 384000] Hz. Rates outside this range are handled as best as possible, depending on the format. Tested: bitexact on a large number of aecdumps Bug: chromium:1332484, chromium:1334991 Change-Id: I9639d03dc837e1fdff64d1f9d1fff0edc0fb299f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276920 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38663} 3 年前
Rename more death test to *DeathTest Bug: webrtc:11577 Change-Id: If45e322fed3f2935e64c9e4d7e8c096eccc53ac4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176140 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31362} 5 年前
Remove more traces of keyboard mic support from APM The 6-parameter Initialize method is removed. The has_keyboard parameter in the StreamConfig constructor is removed together with the underlying member and helper functions. Bug: chromium:1271981, b/217349489 Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221 Reviewed-by: Sam Zackrisson <saza@google.com> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35908} 4 年前
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219} 3 年前
APM: fix TS initialization bugs with WebRTC-Audio-GainController2 When the WebRTC-Audio-GainController2 field trial is used, the initial APM configuration is adjusted depending on its original values and the field trial parameters. This CL fixes two cases when the code crashes: 1. when, in the initial APM config, AGC1 is enabled, AGC2 is disabled and TS is enabled 2. when the initial APM sample rate is different from the capture one and the VAD APM sub-module is not re-initialized This CL also improves the unit tests coverage and it has been tested offline to check that the VAD sub-module is created only when expected and that AGC2 uses its internal VAD when expected. The tests ran on a few Wav files with different sample rates and one AEC dump and on 16 different APM and field trial configurations. Bug: chromium:1407341, b/265112132 Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39118} 3 年前
APM: fix TS initialization bugs with WebRTC-Audio-GainController2 When the WebRTC-Audio-GainController2 field trial is used, the initial APM configuration is adjusted depending on its original values and the field trial parameters. This CL fixes two cases when the code crashes: 1. when, in the initial APM config, AGC1 is enabled, AGC2 is disabled and TS is enabled 2. when the initial APM sample rate is different from the capture one and the VAD APM sub-module is not re-initialized This CL also improves the unit tests coverage and it has been tested offline to check that the VAD sub-module is created only when expected and that AGC2 uses its internal VAD when expected. The tests ran on a few Wav files with different sample rates and one AEC dump and on 16 different APM and field trial configurations. Bug: chromium:1407341, b/265112132 Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39118} 3 年前
Update rtc::Event::Wait call sites to use TimeDelta. Bug: webrtc:14366 Change-Id: I949c1d26f030696b18153afef977633c9a5bd4cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272003 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37835} 3 年前
Remove mentions of already deleted field trials - WebRTC-Audio-Agc2ForceExtraSaturationMargin - WebRTC-Audio-Agc2ForceInitialSaturationMargin - WebRTC-Audio-BitrateAdaptation - WebRTC-Audio-TransientSuppressorVadMode - WebRTC-FrameBuffer3 - WebRTC-IntelVP8 - WebRTC-UseActiveIceController Bug: None Change-Id: I3545727c09f761867f2f4c2bb5c400012ce146d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295723 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39444} 3 年前
Migrate CallSimulator to the new perf metrics logging API Bug: b/246095034 Change-Id: I613f702d2f469b6bc8d1634f8dda40d444ff7cf2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276632 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38213} 3 年前
AGC2 adaptive digital controller config clean-up - Remove dry-run option - Hard-code adjacent_speech_frames_threshold and vad_reset_period_ms - Expose initial_gain_db via field trial Tested: adaptive digital controller bit-exactness verified Bug: webrtc:7494 Change-Id: I6166611f91320b6c37de3f8e553c06c2ed95b772 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287222 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38862} 3 年前
AEC dump Stream::level renamed Making it clear that the field is used to store the applied input volume and not the recommended input volume. Bug: webrtc:7494, b/241923537 Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342 Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38051} 3 年前
Remove more traces of keyboard mic support from APM The 6-parameter Initialize method is removed. The has_keyboard parameter in the StreamConfig constructor is removed together with the underlying member and helper functions. Bug: chromium:1271981, b/217349489 Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221 Reviewed-by: Sam Zackrisson <saza@google.com> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35908} 4 年前
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/ Bug: webrtc:13555, webrtc:13082 Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35771} 4 年前
Use backticks not vertical bars to denote variables in comments for /modules/audio_processing Bug: webrtc:12338 Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001 No-Presubmit: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34690} 4 年前
Remove unused critical section includes. Bug: webrtc:11567 Change-Id: Ic5e43c51ce06c0619adc265d12ad4bef73a9df76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179521 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31745} 5 年前
Format /modules git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^modules/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jared Siskin <jtsiskin@meta.com> Cr-Commit-Position: refs/heads/main@{#39901} 3 年前
AGC1: remove unused field trial WebRTC-UseLegacyDigitalGainApplier Bug: webrtc:14685 Change-Id: I7c9e07c56f20bd9c4b8848787d0b6e4f9785af60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283764 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38687} 3 年前
Remove more traces of keyboard mic support from APM The 6-parameter Initialize method is removed. The has_keyboard parameter in the StreamConfig constructor is removed together with the underlying member and helper functions. Bug: chromium:1271981, b/217349489 Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221 Reviewed-by: Sam Zackrisson <saza@google.com> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35908} 4 年前
APM: fix TS initialization bugs with WebRTC-Audio-GainController2 When the WebRTC-Audio-GainController2 field trial is used, the initial APM configuration is adjusted depending on its original values and the field trial parameters. This CL fixes two cases when the code crashes: 1. when, in the initial APM config, AGC1 is enabled, AGC2 is disabled and TS is enabled 2. when the initial APM sample rate is different from the capture one and the VAD APM sub-module is not re-initialized This CL also improves the unit tests coverage and it has been tested offline to check that the VAD sub-module is created only when expected and that AGC2 uses its internal VAD when expected. The tests ran on a few Wav files with different sample rates and one AEC dump and on 16 different APM and field trial configurations. Bug: chromium:1407341, b/265112132 Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39118} 3 年前
AGC2: Return the recommended volume from RecommendInputVolume() In InputVolumeController, rename AnalyzePreProcess() and Process() to reflect their use and replace the use of the getter recommended_input_volume() with an optional return value from the latter one. The added return value carries the recommended input volume if the call sequence follows the API contract. Make the member applied_input_volume_ optional. Restrict the use of the getter recommended_input_volume() for test use. Add a method capture_output_used() for test use. In GainController2, store the output of InputVolumeController::Process() in a new member variable that's updated in Analyze() and Process(). Use a trivial getter to read the value in APM. Bug: webrtc:7494 Change-Id: Ifcfb466c4f558be560eb6d2f45410d04adb7e2ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287862 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38889} 3 年前
APM: fix TS initialization bugs with WebRTC-Audio-GainController2 When the WebRTC-Audio-GainController2 field trial is used, the initial APM configuration is adjusted depending on its original values and the field trial parameters. This CL fixes two cases when the code crashes: 1. when, in the initial APM config, AGC1 is enabled, AGC2 is disabled and TS is enabled 2. when the initial APM sample rate is different from the capture one and the VAD APM sub-module is not re-initialized This CL also improves the unit tests coverage and it has been tested offline to check that the VAD sub-module is created only when expected and that AGC2 uses its internal VAD when expected. The tests ran on a few Wav files with different sample rates and one AEC dump and on 16 different APM and field trial configurations. Bug: chromium:1407341, b/265112132 Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39118} 3 年前
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED. The new macros will replace the old one when old one's usage will be removed. The idea of the renaming to provide a clear signal that this is debug build only macros and will be stripped in the production build. Bug: webrtc:9065 Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35348} 4 年前
Reduce for reallocations the pre-amplifier and high-pass filter This CL ensures that the pre-amplifier and the high-pass filter submodules are not reallocated more than needed. Bug: webrtc:5298 Change-Id: I7ed23807d4d2d9fef0eda2e7dca9de9b0b1a4649 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163988 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30143} 6 年前
Remove more traces of keyboard mic support from APM The 6-parameter Initialize method is removed. The has_keyboard parameter in the StreamConfig constructor is removed together with the underlying member and helper functions. Bug: chromium:1271981, b/217349489 Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221 Reviewed-by: Sam Zackrisson <saza@google.com> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35908} 4 年前
APM Transient Suppressor (TS): initialization params in ctor More robust API option that allows to fully initialize TS when created. Bug: webrtc:13663 Change-Id: I42c38612ef772eb6d0bbde49d04ea39332a0e3c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255821 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36490} 4 年前
APM Transient Suppressor (TS): initialization params in ctor More robust API option that allows to fully initialize TS when created. Bug: webrtc:13663 Change-Id: I42c38612ef772eb6d0bbde49d04ea39332a0e3c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255821 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36490} 4 年前
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}8 年前
Delete rtc_base/atomic_ops.h Bug: webrtc:9305 Change-Id: I3e8b0db03b84b5361d63db31ee23e6db3deabfe4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266497 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37348} 3 年前
Delete rtc_base/atomic_ops.h Bug: webrtc:9305 Change-Id: I3e8b0db03b84b5361d63db31ee23e6db3deabfe4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266497 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37348} 3 年前
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219} 3 年前
Ensure that an RTP audio level of 127 represents digital silence. To quote rfc6464: The audio level for digital silence -- for a muted audio source, for example -- MUST be represented as 127 (-127 dBov), regardless of the dynamic range of the encoded audio format. The behavior in webrtc is correct that digital silence is represented with 127, but it is also possible to get a value of 127 for not quite digitally silent audio buffer (as in, not strictly 0s). Bug: webrtc:14029 Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36793} 3 年前
Ensure that an RTP audio level of 127 represents digital silence. To quote rfc6464: The audio level for digital silence -- for a muted audio source, for example -- MUST be represented as 127 (-127 dBov), regardless of the dynamic range of the encoded audio format. The behavior in webrtc is correct that digital silence is represented with 127, but it is also possible to get a value of 127 for not quite digitally silent audio buffer (as in, not strictly 0s). Bug: webrtc:14029 Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36793} 3 年前
Ensure that an RTP audio level of 127 represents digital silence. To quote rfc6464: The audio level for digital silence -- for a muted audio source, for example -- MUST be represented as 127 (-127 dBov), regardless of the dynamic range of the encoded audio format. The behavior in webrtc is correct that digital silence is represented with 127, but it is also possible to get a value of 127 for not quite digitally silent audio buffer (as in, not strictly 0s). Bug: webrtc:14029 Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36793} 3 年前
Optimizations and refactoring of the APM 3-band split filter This CL refactors and optimizes the 3-band split-filter in APM, which is a very computationally complex component. Beyond optimizing the code, the filter coefficients are also quantized to avoid denormals. The changes reduces the complexity of the split filter by about 30-50%. The CL has been tested for bitexactness on a number of aecdump recordings. (the CL also removes the now unused code for the sparse_fir_filter) Bug: webrtc:6181 Change-Id: If45f8d1f189c6812ccb03721156c77eb68181211 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168189 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30592} 6 年前
Optimizations and refactoring of the APM 3-band split filter This CL refactors and optimizes the 3-band split-filter in APM, which is a very computationally complex component. Beyond optimizing the code, the filter coefficients are also quantized to avoid denormals. The changes reduces the complexity of the split filter by about 30-50%. The CL has been tested for bitexactness on a number of aecdump recordings. (the CL also removes the now unused code for the sparse_fir_filter) Bug: webrtc:6181 Change-Id: If45f8d1f189c6812ccb03721156c77eb68181211 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168189 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30592} 6 年前
Reland "Simplification and refactoring of the AudioBuffer code" This is a reland of 81c0cf287c8514cb1cd6f3baca484d668c6eb128 Original change's description: > Simplification and refactoring of the AudioBuffer code > > This CL performs a major refactoring and simplification > of the AudioBuffer code that. > -Removes 7 of the 9 internal buffers of the AudioBuffer. > -Avoids the implicit copying required to keep the > internal buffers in sync. > -Removes all code relating to handling of fixed-point > sample data in the AudioBuffer. > -Changes the naming of the class methods to reflect > that only floating point is handled. > -Corrects some bugs in the code. > -Extends the handling of internal downmixing to be > more generic. > > Bug: webrtc:10882 > Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828 > Commit-Queue: Per Åhgren <peah@webrtc.org> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28928} Bug: webrtc:10882 Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28938} 6 年前
Optimize the three band filter bank. Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag. Bug: None Change-Id: I8e2d841fa543b28c59eb08c654a2b0515ab39d69 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241780 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Christian Schuldt <cschuldt@google.com> Cr-Commit-Position: refs/heads/main@{#35548} 4 年前
Use backticks not vertical bars to denote variables in comments for /modules/audio_processing Bug: webrtc:12338 Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001 No-Presubmit: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34690} 4 年前