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zhong-lupingzhong-luping删除resource目录下多余的二进制文件
48f74023创建于 2024年2月27日历史提交
文件最后提交记录最后更新时间
audio_coding
删除resource目录下多余的二进制文件 Signed-off-by: zhongluping <278527840@qq.com> 2 年前
audio_device
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
audio_processing
删除resource目录下多余的二进制文件 Signed-off-by: zhongluping <278527840@qq.com> 2 年前
images
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
media
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495} 10 年前
network_tester
Reland "Add first part of the network_tester functionality". This was originally proposed in https://codereview.webrtc.org/2779233002, but due to upstreaming errors, reverted and relanded a few times. This is a tested reland of it. BUG=webrtc:7426 Review-Url: https://codereview.webrtc.org/2821133004 Cr-Commit-Position: refs/heads/master@{#17756} 8 年前
remote_bitrate_estimator
Add baseline generation/verification to BWE test framework. Updating resource file separately, once LGTM. Generates ~628k of files for current tests, highly compressable, once/if we need that. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5204 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
rtc_event_log
C-style bindings around RTC event log analyzer (2). Parses log, calls analyzer and populates output. Currently only outputs two charts. Chart selection to be added in a followup. Bug: None Change-Id: I960cff15a5935a638a5d979a71230ad598083596 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41000} 2 年前
rtp_rtcp
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
utility
Fix a bug preventing FilePlayer from playing encoded wav files A bug in ACM2 prevented decoding and playout of wav files where the audio data was encoded (i.e., not just linear PCM 16 bit data). This CL fixes the issue, and adds a unit test for the FilePlayer. BUG=3386 R=henrike@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
video_coding
Minor modifications to test::RtpFileReader Adding original_length to the Packet struct. This is populated with the plen value from the RTP dump file. In the case of reading a pcap file, original_length will be equal to length. Also increasing the maximum packet size to 3500 bytes. This is to accomodate some test files that contain PCM16b audio encoding. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
video_engine
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
voice_engine
删除resource目录下多余的二进制文件 Signed-off-by: zhongluping <278527840@qq.com> 2 年前
.gitignore
Ignore .binarypb files. No-Try: True Bug: None Change-Id: If675b0e8e896250cefa7c593d7a684d94b1871d0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325284 Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41042} 2 年前
ConferenceMotion_1280_720_50.yuv.sha1
Add an HD resolution perf test. Also update existing perf tests to use send side bwe. BUG=webrtc:4604, chromium:522001 Review-Url: https://codereview.webrtc.org/2227733004 Cr-Commit-Position: refs/heads/master@{#13726} 9 年前
FourPeople_1280x720_30.yuv.sha1
Adding FourPeople_1280x720_30.yuv. Typical conference content in most popular format (1280x720 30fps). Bug: none Change-Id: I61ec1af44e65e5aec2f2f5e5ecb101b10b423c8b Reviewed-on: https://webrtc-review.googlesource.com/51761 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21986}8 年前
OWNERS
Add OWNERS for resources/ Make it possible for all our committers to upload resource .sha1 files in here. TEST=none BUG=2294 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5085 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
att-downlink.rx.sha1
Add trace-based delivery filter to BWE test framework. R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5889005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
att-uplink.rx.sha1
Add trace-based delivery filter to BWE test framework. R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5889005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
deflicker_before_cif_short.yuv.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
difficult_photo_1850_1110.yuv.sha1
Offline screenshare quality test, plus loopback. BUG=4171 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34109004 Cr-Commit-Position: refs/heads/master@{#8408} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
e2e_audio_in.pcm.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
far16_stereo.pcm.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
far176_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
far192_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
far22_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
far32_stereo.pcm.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
far44_stereo.pcm.sha1
Add resource audio for audio processing tests. This is a prerequisite of: http://review.webrtc.org/9919004/ TBR=bjornv BUG=2894 Review URL: https://webrtc-codereview.appspot.com/12219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5945 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
far48_stereo.pcm.sha1
Add resource audio for audio processing tests. This is a prerequisite of: http://review.webrtc.org/9919004/ TBR=bjornv BUG=2894 Review URL: https://webrtc-codereview.appspot.com/12219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5945 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
far88_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
far8_stereo.pcm.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
far96_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
foremanColorEnhanced_cif_short.yuv.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
foreman_128x96.yuv.sha1
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716} 9 年前
foreman_160x120.yuv.sha1
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716} 9 年前
foreman_176x144.yuv.sha1
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716} 9 年前
foreman_240x136.yuv.sha1
Add MediaCodec VP tests for uncommon resolutions. Bug: None Change-Id: Ibfc35af3635c3b3a50027c4cd828f78e7a438dcd Reviewed-on: https://webrtc-review.googlesource.com/72342 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23020}7 年前
foreman_320x240.yuv.sha1
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716} 9 年前
foreman_480x272.yuv.sha1
Add MediaCodec VP tests for uncommon resolutions. Bug: None Change-Id: Ibfc35af3635c3b3a50027c4cd828f78e7a438dcd Reviewed-on: https://webrtc-review.googlesource.com/72342 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23020}7 年前
foreman_cif.yuv.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
foreman_cif_short.yuv.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
google-wifi-3mbps.rx.sha1
Enable GoogleWifiTrace3Mbps simulations. BUG=3277 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50829004 Cr-Commit-Position: refs/heads/master@{#9131} 10 年前
near16_stereo.pcm.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
near176_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
near192_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
near22_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
near32_stereo.pcm.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
near44_stereo.pcm.sha1
Add resource audio for audio processing tests. This is a prerequisite of: http://review.webrtc.org/9919004/ TBR=bjornv BUG=2894 Review URL: https://webrtc-codereview.appspot.com/12219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5945 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
near48_stereo.pcm.sha1
Add resource audio for audio processing tests. This is a prerequisite of: http://review.webrtc.org/9919004/ TBR=bjornv BUG=2894 Review URL: https://webrtc-codereview.appspot.com/12219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5945 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
near88_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
near8_stereo.pcm.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
near96_stereo.pcm.sha1
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前
paris_qcif.yuv.sha1
Replace old resources download script with depot_tools With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
pc_quality_smoke_test_alice_source.wav
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
pc_quality_smoke_test_alice_source.wav.sha1
Add support for creation of AEC dump during the test with PC framework. Also add conversational speech into PC smoke test (with resource files). Bug: webrtc:10138 Change-Id: I415a5565bc9146821476ffc60f57f47ed51f89c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132323 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27592}7 年前
pc_quality_smoke_test_bob_source.wav
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
pc_quality_smoke_test_bob_source.wav.sha1
Add support for creation of AEC dump during the test with PC framework. Also add conversational speech into PC smoke test (with resource files). Bug: webrtc:10138 Change-Id: I415a5565bc9146821476ffc60f57f47ed51f89c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132323 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27592}7 年前
photo_1850_1110.yuv.sha1
Offline screenshare quality test, plus loopback. BUG=4171 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34109004 Cr-Commit-Position: refs/heads/master@{#8408} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
presentation_1850_1110.yuv.sha1
Offline screenshare quality test, plus loopback. BUG=4171 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34109004 Cr-Commit-Position: refs/heads/master@{#8408} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
ref03.aecdump.sha1
Add tests and modify tools for new float deinterleaved interface. - Add an Initialize() overload to allow specification of format parameters. This is mainly useful for testing, but could be used in the cases where a consumer knows the format before the streams arrive. - Add a reverse_sample_rate_hz_ parameter to prepare for mismatched capture and render rates. There is no functional change as it is currently constrained to match the capture rate. - Fix a bug in the float dump: we need to use add_ rather than set_. - Add a debug dump test for both int and float interfaces. - Enable unpacking of float dumps. - Enable audioproc to read float dumps. - Move more shared functionality to test_utils.h, and generally tidy up a bit by consolidating repeated code. BUG=2894 TESTED=Verified that the output produced by the float debug dump test is correct. Processed the resulting debug dump file with audioproc and ensured that we get identical output. (This is crucial, as we need to be able to exactly reproduce online results offline.) R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
reference_less_video_test_file.y4m.sha1
Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame. Also added a unit test case which compares the extracted frame with the frame stored in text file. NOPRESUBMIT=true NOTRY=true BUG=webrtc:6761 Review-Url: https://codereview.webrtc.org/2532963002 Cr-Commit-Position: refs/heads/master@{#15288} 9 年前
reference_video_640x360_30fps.y4m.sha1
Espresso test case to control loopback call The test case is put inside a new test target. That test target will be started from a test script to asses video quality. BUG=webrtc:6545 Review-Url: https://codereview.webrtc.org/2585813002 Cr-Commit-Position: refs/heads/master@{#16088} 9 年前
short_mixed_mono_48.dat.sha1
Roll chromium_revision a8e5140..c6076f2 (372922:372974) incl. update to Opus v.1.1.2 Includes updates to tests for Opus v.1.1.2, reveiwed in https://codereview.webrtc.org/1629413002/ Change log: https://chromium.googlesource.com/chromium/src/+log/a8e5140..c6076f2 Full diff: https://chromium.googlesource.com/chromium/src/+/a8e5140..c6076f2 Changed dependencies: * src/third_party/catapult: https://chromium.googlesource.com/external/github.com/catapult-project/catapult.git/+log/471db30..d4d48e6 * src/third_party/opus/src: https://chromium.googlesource.com/chromium/deps/opus.git/+log/cae6961..655cc54 DEPS diff: https://chromium.googlesource.com/chromium/src/+/a8e5140..c6076f2/DEPS No update to Clang. BUG=chromium:580524 TBR= Review URL: https://codereview.webrtc.org/1657343002 Cr-Commit-Position: refs/heads/master@{#11464} 10 年前
short_mixed_mono_48.pcm.sha1
adding sha1 files for audio classifier test This needs to done in a separate CL since the Android APK trybots cannot handle patches into the resources directory due to the fact that they work from a Chromium checkout and applies the patch into src/third_party/webrtc. BUG= R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5643 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
short_mixed_mono_48_arm.dat.sha1
Roll chromium_revision a8e5140..c6076f2 (372922:372974) incl. update to Opus v.1.1.2 Includes updates to tests for Opus v.1.1.2, reveiwed in https://codereview.webrtc.org/1629413002/ Change log: https://chromium.googlesource.com/chromium/src/+log/a8e5140..c6076f2 Full diff: https://chromium.googlesource.com/chromium/src/+/a8e5140..c6076f2 Changed dependencies: * src/third_party/catapult: https://chromium.googlesource.com/external/github.com/catapult-project/catapult.git/+log/471db30..d4d48e6 * src/third_party/opus/src: https://chromium.googlesource.com/chromium/deps/opus.git/+log/cae6961..655cc54 DEPS diff: https://chromium.googlesource.com/chromium/src/+/a8e5140..c6076f2/DEPS No update to Clang. BUG=chromium:580524 TBR= Review URL: https://codereview.webrtc.org/1657343002 Cr-Commit-Position: refs/heads/master@{#11464} 10 年前
short_mixed_stereo_48.dat.sha1
Adding new data files for audio classifier unit testing on Android try bots BUG= R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5675 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前
short_mixed_stereo_48.pcm.sha1
adding sha1 files for audio classifier test This needs to done in a separate CL since the Android APK trybots cannot handle patches into the resources directory due to the fact that they work from a Chromium checkout and applies the patch into src/third_party/webrtc. BUG= R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5643 4adac7df-926f-26a2-2b94-8c16560cd09d 12 年前
web_screenshot_1850_1110.yuv.sha1
Offline screenshare quality test, plus loopback. BUG=4171 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34109004 Cr-Commit-Position: refs/heads/master@{#8408} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d 11 年前