2e1f16d5创建于 2023年10月26日历史提交
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Moving src/webrtc into src/. In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}8 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
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 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
Add more refined control over dumping of data and the aecdump content This CL adds the ability in audioproc_f and unpack_aecdump to: -Clearly identify the Init events and when those occur. -Optionally only process a specific Init section of an aecdump. -Optionally selectively turn on dumping of internal data for a specific init section, and a specific time interval. -Optionally let unpack_aecdump produce file names based on inits. Bug: webrtc:5298 Change-Id: Id654b7175407a23ef634fca832994d87d1073239 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196160 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33181} 5 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Moving src/webrtc into src/. In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}8 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
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
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 fuzzer: add SetConfig() to test builder Also stop using ApplyConfig() and in [1] fix the build errors when WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE is defined. [1] modules/audio_processing/test/audio_processing_builder_for_testing.cc Bug: webrtc:5298 Change-Id: I50dc5668b952e7ca7fa83c7a5182c013e928c450 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235365 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35228} 4 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
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
APM: refactor emulated input volume for capture level adjustment Switching to an AGC implementation agnostic solution for the input volume emulation functionality offered by the capture_levels_adjuster sub-module. This CL also fixes a (silent) bug due to which, when the input volume is emulated via the capture adjuster sub-module, AGC2 reads an incorrect value for the applied input volume. Tested: audioproc_f with --analog_mic_gain_emulation 1 used to verify bit-exactness for one Wav file and one AEC dump for which the input volume varies. Bug: webrtc:7494, b/241923537 Change-Id: Ide3085f9a5dfd85888ad812ebd56faa175fb2ba7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273902 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38053} 3 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
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
Add DCHECK and documentation to disallow trying to read more than two audio channels in helper function. Bug: webrtc:8741 Change-Id: Ib55b15bb1802b412be17ef8199d6112937502cd3 Reviewed-on: https://webrtc-review.googlesource.com/39263 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21603}8 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
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
APM: remove kClippedLevelMin from audio_processing.h Bug: webrtc:7494 Change-Id: I91ed3b82592d9801b113ca72a2b2221b5abf20a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278788 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38389} 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
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
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
AEC3: Make RenderSignalAnalyzer multi-channel In this CL: - Render signal analyzer considers a frequency bin a narrow band (peak) if any channel exhibits narrowband (-peak) behavior. - The unit tests have to fill frames with noise because small inaccuracies in the FFT spectrum lead to consistent "narrow bands" despite spectrum being essentially flat. Bug: webrtc:10913 Change-Id: I8fa181412c0ee1beeacfda37ffef18251d5f0cd7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151912 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29176} 6 年前
AEC3: Add support in the echo subtractor for handling multiple channels This CL adds support in the echo subtractor for handling multiple capture and render channels. The changes have passed bitexactness tests for substantial set of mono recordings. Bug: webrtc:10913 Change-Id: Ib448c9edf172ebc31e8c28db7b2f2a389a53adb9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155168 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29389} 6 年前
AEC3: Make RenderSignalAnalyzer multi-channel In this CL: - Render signal analyzer considers a frequency bin a narrow band (peak) if any channel exhibits narrowband (-peak) behavior. - The unit tests have to fill frames with noise because small inaccuracies in the FFT spectrum lead to consistent "narrow bands" despite spectrum being essentially flat. Bug: webrtc:10913 Change-Id: I8fa181412c0ee1beeacfda37ffef18251d5f0cd7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151912 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29176} 6 年前
Enabling a safe fall-back functionality for overruns in the runtime settings Bug: b/177830919 Change-Id: I9369f6fc004ceb2b626d33b36262bc8aeabdb1a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206988 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33371} 5 年前
ClippingPredictor/Evaluator/LevelBuffer and GainMap: Move to agc2 Bug: webrtc:7494 Change-Id: If88795fe34a73faa267a9c0bd5250e36455d4d81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277741 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38296} 3 年前
Remove top-level const from parameters in function declarations. This is a safe cleanup change since top-level const applied to parameters in function declarations (that are not also definitions) are ignored by the compiler. Hence, such changes do not change the type of the declared functions and are simply no-ops. Bug: webrtc:13610 Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35802} 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
Optional: Use nullopt and implicit construction in /modules/audio_processing Changes places where we explicitly construct an Optional to instead use nullopt or the requisite value type only. This CL was uploaded by git cl split. R=henrik.lundin@webrtc.org Bug: None Change-Id: I733a83f702fe11884d229a1713cfac952727bde8 Reviewed-on: https://webrtc-review.googlesource.com/23601 Commit-Queue: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20786}8 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
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209} 6 年前
audioproc_f: input AEC dump as string, output audio to vector This CL adds the following options: pass an input AEC dump as a string (currently, the tool can only accept a path to an AEC dump file) write the processed capture samples to a given vector Bug: webrtc:10808 Change-Id: I02863c97ec3cd8c03ade2ea8521836f2e7417050 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145208 Commit-Queue: Sonia-Florina Horchidan <soniahorchidan@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28826} 6 年前
Reland "Add ability to state whether the APM output will be used" This is a reland of 8be2f201ba8790501f6f3fa39f00017f02fca46d Original change's description: > Add ability to state whether the APM output will be used > > This CL adds the ability for the surrounding code to state that the > APM output will not be used. The intended usecase for this is to allow > APM to run at a lower complexity when the endpoint is muted. > When APM has been informed that the output will not be used, it can > turn off code that is needed only for ensuring that the output audio > will sound good. > > Bug: b/154437967,b/163802450 > Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Per Åhgren <peah@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31949} Bug: b/154437967 Bug: b/163802450 Change-Id: Ia77a9e43f913929d1afa72212f1ea6c192d0e519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181887 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31957} 5 年前
Store RuntimeSetting in Aec Dumps. Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}7 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
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
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs This replaces the current usage of AudioProcessing::level_estimator() in that test. The unit tests that specifically test the level_estimator API are left in place, until the level_estimator API itself is removed. Bug: webrtc:9947 Change-Id: I73301c1478d2c9763bb49598a692142229102876 Reviewed-on: https://webrtc-review.googlesource.com/c/114550 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26049}7 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前
Adopt absl::string_view in modules/audio_processing/ Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800} 3 年前