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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("video_coding") {
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
sources = [
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"codec_database.cc",
"codec_database.h",
"codec_timer.cc",
"codec_timer.h",
"decoding_state.cc",
"decoding_state.h",
"encoded_frame.cc",
"encoded_frame.h",
Rename fec_tables_xor.h -> fec_rate_table.h This is a slightly more descriptive name, since we only have one type of erasure code (XOR), and we only have one table. BUG=None Review-Url: https://codereview.webrtc.org/2625903004 Cr-Commit-Position: refs/heads/master@{#16032}
2017-01-12 06:16:24 -08:00
"fec_rate_table.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"frame_buffer.cc",
"frame_buffer.h",
Revert "Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ )" Also disabled modules_unittest.TestFrameBuffer2.* in drmemory. This reverts commit b711f10d9683b9de6ee78186f77b225fc7ebfb8f. TBR=honghaiz@webrtc.org BUG= Review URL: https://codereview.webrtc.org/1991133003 . Cr-Commit-Position: refs/heads/master@{#12806}
2016-05-19 12:19:35 +02:00
"frame_buffer2.cc",
"frame_buffer2.h",
Packet buffer for the new jitter buffer. BUG=webrtc:5514 R=stefan@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1772383002 Cr-Commit-Position: refs/heads/master@{#12194}
2016-04-01 02:01:54 -07:00
"frame_object.cc",
"frame_object.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"generic_decoder.cc",
"generic_decoder.h",
"generic_encoder.cc",
"generic_encoder.h",
Decode h264 fmtp sprop-parameter-sets to binary. BUG=webrtc:5948 Review-Url: https://codereview.webrtc.org/2544493005 Cr-Commit-Position: refs/heads/master@{#15474}
2016-12-08 03:57:17 -08:00
"h264_sprop_parameter_sets.cc",
"h264_sprop_parameter_sets.h",
H264SpsPpsTracker class which keep tracks of SPS/PPS. The H264SpsPpsTracker class: - Keeps track of all received SPS/PPS. - Decides whether a packet should be inserted into the PacketBuffer or not. - Don't insert if this packet only contains SPS and/or PPS. - Don't insert if this is the first packet of and IDR and we have not received the required SPS/PPS. - Insert start codes, and in the case of the first packet of an IDR prepend the bitstream with the given SPS/PPS for this IDR. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/2466993003 Cr-Commit-Position: refs/heads/master@{#14906}
2016-11-03 04:03:01 -07:00
"h264_sps_pps_tracker.cc",
"h264_sps_pps_tracker.h",
Experiment for the nack module. Testing the nack module by implementing it into the current jitter buffer under the experiment WebRTC-NewVideoJitterBuffer. BUG=webrtc:5514 Review URL: https://codereview.webrtc.org/1778503002 Cr-Commit-Position: refs/heads/master@{#11969}
2016-03-12 03:30:23 -08:00
"histogram.cc",
"histogram.h",
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
"include/video_codec_initializer.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"include/video_coding.h",
"include/video_coding_defines.h",
"inter_frame_delay.cc",
"inter_frame_delay.h",
"internal_defines.h",
"jitter_buffer.cc",
"jitter_buffer.h",
"jitter_buffer_common.h",
"jitter_estimator.cc",
"jitter_estimator.h",
"media_opt_util.cc",
"media_opt_util.h",
"media_optimization.cc",
"media_optimization.h",
"nack_fec_tables.h",
Experiment for the nack module. Testing the nack module by implementing it into the current jitter buffer under the experiment WebRTC-NewVideoJitterBuffer. BUG=webrtc:5514 Review URL: https://codereview.webrtc.org/1778503002 Cr-Commit-Position: refs/heads/master@{#11969}
2016-03-12 03:30:23 -08:00
"nack_module.cc",
"nack_module.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"packet.cc",
"packet.h",
Packet buffer for the new jitter buffer. BUG=webrtc:5514 R=stefan@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1772383002 Cr-Commit-Position: refs/heads/master@{#12194}
2016-04-01 02:01:54 -07:00
"packet_buffer.cc",
"packet_buffer.h",
Move logic for calculating needed bitrate overhead used by NACK and FEC to VideoSender. This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder Note that the logic of how FEC and the needed bitrates is not changed. BUG=webrtc:5687 R=asapersson@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1972083002 . Cr-Commit-Position: refs/heads/master@{#13018}
2016-06-02 15:45:42 +02:00
"protection_bitrate_calculator.cc",
"protection_bitrate_calculator.h",
Refactor + enable GN check on video_coding_utility To avoid the cyclic dependency BUG=webrtc:6828 NOTRY=True TBR=magjed@webrtc.org Review-Url: https://codereview.webrtc.org/2717113002 Cr-Commit-Position: refs/heads/master@{#17116}
2017-03-08 05:42:26 -08:00
"qp_parser.cc",
"qp_parser.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"receiver.cc",
"receiver.h",
Logic for finding frame references moved from PacketBuffer to new class RtpFrameReferenceFinder. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/1961053002 Cr-Commit-Position: refs/heads/master@{#12725}
2016-05-13 06:01:03 -07:00
"rtp_frame_reference_finder.cc",
"rtp_frame_reference_finder.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"rtt_filter.cc",
"rtt_filter.h",
"session_info.cc",
"session_info.h",
"timestamp_map.cc",
"timestamp_map.h",
"timing.cc",
"timing.h",
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
"video_codec_initializer.cc",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"video_coding_impl.cc",
"video_coding_impl.h",
"video_receiver.cc",
"video_sender.cc",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
GN: Replace Windows suppressions of warning 4267 with config. This makes the GN configurations easier to read. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2020343003 Cr-Commit-Position: refs/heads/master@{#13006}
2016-06-02 02:09:52 -07:00
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
GN: Fix Windows build. This required a tiny include fix in src/third_party/winsdk_samples/src which was committed in https://code.google.com/p/webrtc/source/detail?r=7951 This incorporates contribution from vchigrin@yandex-team.ru in https://webrtc-codereview.appspot.com/29299004/ BUG=261,1348,4105 R=pbos@webrtc.org TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 21:22:01 +00:00
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
deps = [
Move SPS/PPS/IDR requirement from RtpFrameObject to PacketBuffer. BUG=webrtc:8423 Change-Id: I0f0d59461afead700c20c9a2ed9b2bc991590b4a Reviewed-on: https://webrtc-review.googlesource.com/15101 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20559}
2017-11-02 14:28:06 +01:00
":codec_globals_headers",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
":video_coding_utility",
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
":webrtc_h264",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
":webrtc_i420",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
":webrtc_stereo",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
":webrtc_vp8",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
":webrtc_vp9",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
In generic encoder ensure capture timestamp is preserved. Because some hardware encoders may not preserve capture timestamp, use rtp timestamp as a key to capture time storage and restore corrupted capture timestamps whenever possible. Bug: webrtc:8497 Change-Id: Ib3449313c65e864a89cd645a3f5dec99aa9cf8e4 Reviewed-on: https://webrtc-review.googlesource.com/23620 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20732}
2017-11-16 14:33:53 +01:00
"..:module_api_public",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"../..:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
"../../call:video_stream_api",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../../common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:sequenced_task_checker",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../../system_wrappers",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
Piggybacking simulcast id and ALR experiment id into video content type extension. Use it to slice UMA video receive statis. BUG=8032 Review-Url: https://codereview.webrtc.org/2986893002 Cr-Commit-Position: refs/heads/master@{#19598}
2017-08-30 03:32:14 -07:00
"../pacing",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../utility:utility",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
}
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
rtc_source_set("mock_headers") {
testonly = true
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
sources = [
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
"codecs/interface/mock/mock_video_codec_interface.h",
"include/mock/mock_video_codec_interface.h",
]
deps = [
":video_coding_utility",
"../..:webrtc_common",
"../../test:test_support",
]
}
rtc_source_set("codec_globals_headers") {
sources = [
"codecs/h264/include/h264_globals.h",
"codecs/interface/common_constants.h",
Reland "Add stereo codec header and pass it through RTP" This is a reland of 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1 Original change's description: > Add stereo codec header and pass it through RTP > > - Defines CodecSpecificInfoStereo that carries stereo specific header info from > encoded image. > - Defines RTPVideoHeaderStereo that carries the above info to packetizer, > see module_common_types.h. > - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo > header. > - Uses new data containers in StereoAdapter classes. > > This CL is the step 3 for adding alpha channel support over the wire in webrtc. > See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental > CL that gives an idea about how it will come together. > Design Doc: https://goo.gl/sFeSUT > > Bug: webrtc:7671 > Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab > Reviewed-on: https://webrtc-review.googlesource.com/22900 > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20920} TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org Bug: webrtc:7671 Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0 Reviewed-on: https://webrtc-review.googlesource.com/27160 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20946}
2017-11-28 09:45:25 -08:00
"codecs/stereo/include/stereo_globals.h",
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
"codecs/vp8/include/vp8_globals.h",
"codecs/vp9/include/vp9_globals.h",
]
}
rtc_source_set("video_coding_utility") {
sources = [
"codecs/interface/video_codec_interface.h",
"codecs/interface/video_error_codes.h",
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
"utility/default_video_bitrate_allocator.cc",
"utility/default_video_bitrate_allocator.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
"utility/frame_dropper.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/frame_dropper.h",
Add support for writing raw encoder output to .ivf files. Also refactor GenericEncoder to use these file writers, and remove use of preprocessor to enable file writing. BUG= Review URL: https://codereview.webrtc.org/1853813002 Cr-Commit-Position: refs/heads/master@{#12372}
2016-04-15 01:24:14 -07:00
"utility/ivf_file_writer.cc",
"utility/ivf_file_writer.h",
Refactor QualityScaler and MovingAverage The MovingAverage class was very specific to the QualityScaler. This commit generalizes the MovingAverage class to be useful in other situations as well, and adapts the QualityScaler to use the new MovingAverage. BUG=webrtc:6304 Review-Url: https://codereview.webrtc.org/2310853002 Cr-Commit-Position: refs/heads/master@{#14207}
2016-09-14 02:14:58 -07:00
"utility/moving_average.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/moving_average.h",
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
"utility/quality_scaler.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/quality_scaler.h",
Hardware VP8 encoding: Use QP as metric for resize. Add vp8 frame header parser to get QP from vp8 bitstream. BUG= 4273 R=glaznev@webrtc.org, marpan@google.com, pbos@webrtc.org TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49259004 Cr-Commit-Position: refs/heads/master@{#9256}
2015-05-21 11:12:02 -07:00
"utility/vp8_header_parser.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/vp8_header_parser.h",
Add vp9 QP parser. BUG=webrtc:7662 Review-Url: https://codereview.webrtc.org/2891803003 Cr-Commit-Position: refs/heads/master@{#18260}
2017-05-24 10:00:16 -07:00
"utility/vp9_uncompressed_header_parser.cc",
"utility/vp9_uncompressed_header_parser.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../..:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed. (patchset #1 id:1 of https://codereview.webrtc.org/2149553002/ )" This reverts commit efd902cb1d9bbd81247a3e168f2080beae761d78. Originally reviewed in https://codereview.webrtc.org/2149553002 The uptream problem should be fixed by https://codereview.webrtc.org/2145393003/ BUG=webrtc:5687 TBR=tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2152013002 Cr-Commit-Position: refs/heads/master@{#13483}
2016-07-14 23:35:55 -07:00
"../../common_video",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../../modules/rtp_rtcp",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:sequenced_task_checker",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"../../system_wrappers",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_h264") {
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
sources = [
"codecs/h264/h264.cc",
"codecs/h264/include/h264.h",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
}
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. It works on all platforms except Android and iOS (FFmpeg limitation). Implemented behind compile time flags, off by default. The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default. Flags to turn it on: - rtc_use_h264 = true - ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder) Tests using H264: - video_loopback --codec=H264 - screenshare_loopback --codec=H264 - video_engine_tests (EndToEndTest.SendsAndReceivesH264) NOTRY=True BUG=500605, 468365 BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424 Review URL: https://codereview.webrtc.org/1306813009 Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 01:36:03 -08:00
defines = []
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
deps = [
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
":video_coding_utility",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Move list of supported H264 codecs from InternalEncoderFactory to h264.h This CL is a clean-up to prepare for adding more supported codecs for the internal H264 SW codec. Bug: webrtc:8317 Change-Id: If483d05c01c40bbc81cbd1a6aad89961689714ef Reviewed-on: https://webrtc-review.googlesource.com/4940 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20105}
2017-09-29 17:54:09 +02:00
"../../api/video_codecs:video_codecs_api",
Enable include check for webrtc_h264 target BUG=webrtc:7242,webrtc:6828 Review-Url: https://codereview.webrtc.org/2718143002 Cr-Commit-Position: refs/heads/master@{#16888}
2017-02-28 00:28:04 -08:00
"../../media:rtc_media_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:metrics_api",
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
]
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni. BUG=468365 NOTRY=True Review URL: https://codereview.webrtc.org/1601813005 Cr-Commit-Position: refs/heads/master@{#11333}
2016-01-21 03:34:40 -08:00
if (rtc_use_h264) {
Ability to disable the effects of |rtc_use_h264| with DisableRtcUseH264. Renamed the WEBRTC_THIRD_PARTY_H264 macro to WEBRTC_USE_H264 to match flag name. The idea is to be able to turn off H264 from chromium with this function because... 1) The Chromium trybots will soon use this flag, we want to temporarily disable H264 from chromium even if flag is set in case something is broken. That way when we are ready to flip the switch the trybots will run our test code then and not after it is already enabled. 2) If feature is launched and we discover major problems we can easily disable H264 and merge with beta/stable. 3) Or, if feature is behind a *runtime* flag, this is how we would control if it is used or not. The idea is to call DisableRtcUseH264 in chromium's PeerConnectionDependencyFactory. BUG=chromium:500605, chromium:468365 NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1657273002 Cr-Commit-Position: refs/heads/master@{#11474}
2016-02-03 05:09:37 -08:00
defines += [ "WEBRTC_USE_H264" ]
H264: Thread-safe InitializeFFmpeg. Flag to control if InitializeFFmpeg should be called. New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium. In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default. Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg. BUG=chromium:500605, chromium:468365, webrtc:5427 Review URL: https://codereview.webrtc.org/1639273002 Cr-Commit-Position: refs/heads/master@{#11456}
2016-02-02 02:26:05 -08:00
if (rtc_initialize_ffmpeg) {
defines += [ "WEBRTC_INITIALIZE_FFMPEG" ]
}
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. It works on all platforms except Android and iOS (FFmpeg limitation). Implemented behind compile time flags, off by default. The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default. Flags to turn it on: - rtc_use_h264 = true - ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder) Tests using H264: - video_loopback --codec=H264 - screenshare_loopback --codec=H264 - video_engine_tests (EndToEndTest.SendsAndReceivesH264) NOTRY=True BUG=500605, 468365 BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424 Review URL: https://codereview.webrtc.org/1306813009 Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 01:36:03 -08:00
sources += [
"codecs/h264/h264_decoder_impl.cc",
"codecs/h264/h264_decoder_impl.h",
"codecs/h264/h264_encoder_impl.cc",
"codecs/h264/h264_encoder_impl.h",
]
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
deps += [
Fix GYP and GN references that are invalid in Chromium builds. There were a couple of GN and GYP references that were incorrect in Chromium builds: - GN references between WebRTC targets must be using relative paths, not absolute. - GYP references between WebRTC targets must be using the <(webrtc_root)v variable in order to be expanded to the correct path in a Chromium build. NOTRY=True TBR=hjon@webrtc.org, hbos@webrtc.org Review URL: https://codereview.webrtc.org/1681493002 Cr-Commit-Position: refs/heads/master@{#11521}
2016-02-07 23:04:26 -08:00
"../../common_video",
Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ ) Reason for revert: Fixed timeouts in slow tests Original issue's description: > Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ ) > > Reason for revert: > Failures on the Linux Memcheck bot > > Original issue's description: > > This approach passes packetization mode to the encoder as part of > > a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec. > > > > BUG=600254 > > > > Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257 > > Cr-Commit-Position: refs/heads/master@{#15437} > > TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=600254 > > Committed: https://crrev.com/243a0a7a7fd6b5da1e32df31f1bfbb6a68dc09f3 > Cr-Commit-Position: refs/heads/master@{#15441} TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=600254 Review-Url: https://codereview.webrtc.org/2558463002 Cr-Commit-Position: refs/heads/master@{#15445}
2016-12-06 05:36:03 -08:00
"../../media:rtc_media_base",
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
"//third_party/ffmpeg:ffmpeg",
"//third_party/openh264:encoder",
]
}
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_i420") {
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
sources = [
Move i420 files to the right location There's also a presubmit check that disallows .. references in GYP files, which this solves. BUG=webrtc:5095 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1418753002 . Cr-Commit-Position: refs/heads/master@{#10371}
2015-10-22 12:23:11 +02:00
"codecs/i420/i420.cc",
"codecs/i420/include/i420.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
":video_coding_utility",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../..:webrtc_common",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../../common_video:common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"../../system_wrappers",
Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. This CL removes the following GN variables: rtc_build_libyuv, rtc_libyuv_dir (as requested in webrtc:7906). It also removes some unneeded dependencies on //third_party/libyuv. WebRTC targets were using public_deps to depend on //third_party/libyuv and this created a build graph where targets that were depending on //third_party/libyuv were not declaring the dependency to GN because they were somehow getting it from another target that was exposing //third_party/libyuv header files even if it wasn't directly depending on it. Bug: webrtc:8605, webrtc:7906 Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb Reviewed-on: https://webrtc-review.googlesource.com/32201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:24:00 +01:00
"//third_party/libyuv",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
rtc_static_library("webrtc_stereo") {
sources = [
"codecs/stereo/include/stereo_decoder_adapter.h",
"codecs/stereo/include/stereo_encoder_adapter.h",
"codecs/stereo/stereo_decoder_adapter.cc",
"codecs/stereo/stereo_encoder_adapter.cc",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":video_coding_utility",
"..:module_api",
"../..:webrtc_common",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
"../../api/video_codecs:video_codecs_api",
"../../common_video:common_video",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_vp8") {
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
sources = [
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
"codecs/vp8/default_temporal_layers.cc",
"codecs/vp8/default_temporal_layers.h",
Merge VP8 changes. R=stefan@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/35389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:36:40 +00:00
"codecs/vp8/include/vp8.h",
"codecs/vp8/include/vp8_common_types.h",
"codecs/vp8/screenshare_layers.cc",
"codecs/vp8/screenshare_layers.h",
Refactor + enable GN check on video_coding_utility To avoid the cyclic dependency BUG=webrtc:6828 NOTRY=True TBR=magjed@webrtc.org Review-Url: https://codereview.webrtc.org/2717113002 Cr-Commit-Position: refs/heads/master@{#17116}
2017-03-08 05:42:26 -08:00
"codecs/vp8/simulcast_rate_allocator.cc",
"codecs/vp8/simulcast_rate_allocator.h",
Implement ScreenshareTemporalLayersChecker Bug: none Change-Id: Ic95156d0f47d186e2289264aa9a916511a8e4510 Reviewed-on: https://webrtc-review.googlesource.com/4960 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20143}
2017-10-04 15:07:09 +02:00
"codecs/vp8/temporal_layers.cc",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
"codecs/vp8/temporal_layers.h",
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
"codecs/vp8/vp8_impl.cc",
"codecs/vp8/vp8_impl.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
GN: Replace Windows suppressions of warning 4267 with config. This makes the GN configurations easier to read. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2020343003 Cr-Commit-Position: refs/heads/master@{#13006}
2016-06-02 02:09:52 -07:00
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
deps = [
":video_coding_utility",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"../..:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
"../../common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
vp8_impl.cc: Make it possible to base postproc deblocking level for arm on qp (e.g. turn off deblocking for low qp values). BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2970923002 Cr-Commit-Position: refs/heads/master@{#19300}
2017-08-09 23:37:05 -07:00
"../../rtc_base:rtc_numerics",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:sequenced_task_checker",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
"../../system_wrappers",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. This CL removes the following GN variables: rtc_build_libyuv, rtc_libyuv_dir (as requested in webrtc:7906). It also removes some unneeded dependencies on //third_party/libyuv. WebRTC targets were using public_deps to depend on //third_party/libyuv and this created a build graph where targets that were depending on //third_party/libyuv were not declaring the dependency to GN because they were somehow getting it from another target that was exposing //third_party/libyuv header files even if it wasn't directly depending on it. Bug: webrtc:8605, webrtc:7906 Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb Reviewed-on: https://webrtc-review.googlesource.com/32201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:24:00 +01:00
"//third_party/libyuv",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
if (rtc_build_libvpx) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps += [ rtc_libvpx_dir ]
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_vp9") {
Make sure vp9 actually gets excluded in gn as well. It was being excluded in this manner in gyp so it should be in gn as well. R=charujain@webrtc.org,kjellander@webrtc.org BUG=webrtc:6412 Review-Url: https://codereview.webrtc.org/2392053003 Cr-Commit-Position: refs/heads/master@{#14525}
2016-10-05 06:52:22 -07:00
if (rtc_libvpx_build_vp9) {
sources = [
"codecs/vp9/include/vp9.h",
"codecs/vp9/screenshare_layers.cc",
"codecs/vp9/screenshare_layers.h",
"codecs/vp9/vp9_frame_buffer_pool.cc",
"codecs/vp9/vp9_frame_buffer_pool.h",
"codecs/vp9/vp9_impl.cc",
"codecs/vp9/vp9_impl.h",
]
} else {
sources = [
"codecs/vp9/vp9_noop.cc",
]
}
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
}
deps = [
":video_coding_utility",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
"../../common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
"../../system_wrappers",
]
if (rtc_build_libvpx) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps += [ rtc_libvpx_dir ]
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_include_tests) {
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
if (is_ios || is_mac) {
Rename all objc targets to be suffixed for consistency BUG=webrtc:6412 Review-Url: https://codereview.webrtc.org/3001023003 Cr-Commit-Position: refs/heads/master@{#19489}
2017-08-24 05:43:45 -07:00
rtc_static_library("codec_test_objc") {
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
sources = [
"codecs/test/objc_codec_h264_test.h",
"codecs/test/objc_codec_h264_test.mm",
]
deps = [
"../../api:video_frame_api",
"../../api/video_codecs:video_codecs_api",
"../../media:rtc_audio_video",
"../../modules:module_api",
Update videoprocessor tool to use new video factory interface This will also cause us to use the new Android HardwareVideoEncoder, instead of the deprecated MediaCodecVideoEncoderFactory. Unfortunately, the new HW encoder does not seem to work as good as the old (or the new encoder is more strict with return values or something). I don't think it adds much value to continue testing the deprecated encoder, so I filed a bug for fixing the new encoder, and in this CL I disabled the tests on Android. I want to remove as many places as possible where we use the old WebRtcVideoEncoderFactory interface, because it makes it more difficult to migrate to the new interface. Bug: webrtc:7925 Change-Id: If8e34752148a5e5139944d2dfbe7e231fe58aeb9 Reviewed-on: https://webrtc-review.googlesource.com/27540 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21037}
2017-12-04 14:09:40 +01:00
"../../rtc_base:rtc_base_approved",
Rename all objc targets to be suffixed for consistency BUG=webrtc:6412 Review-Url: https://codereview.webrtc.org/3001023003 Cr-Commit-Position: refs/heads/master@{#19489}
2017-08-24 05:43:45 -07:00
"../../sdk:common_objc",
"../../sdk:peerconnection_objc",
"../../sdk:peerconnectionfactory_objc",
"../../sdk:videotoolbox_objc",
"../../sdk:videotracksource_objc",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}
Remove webrtc::VideoEncoderFactory Replace the use of webrtc::VideoEncoderFactory with cricket::WebRtcVideoEncoderFactory and remove the adapter classes between these two factory types. Some code changes were necessary in order to accomplish this: * Move SimulcastEncoderAdapter from webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's where it's used). * Rename simulcast_unittest.h to simulcast_test_utility.h and make it into it's own target, because it's used from both simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc. * Remove ownership of the encoder factory from SimulcastEncoderAdapter, and make the necessary changes in surrounding code. The goal with this CL is to clean up the code, and also to free up the name webrtc::VideoEncoderFactory for future use. BUG=webrtc:7925 Review-Url: https://codereview.webrtc.org/2964953002 Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 03:26:36 -07:00
rtc_source_set("simulcast_test_utility") {
testonly = true
sources = [
"codecs/vp8/simulcast_test_utility.h",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
":mock_headers",
Remove webrtc::VideoEncoderFactory Replace the use of webrtc::VideoEncoderFactory with cricket::WebRtcVideoEncoderFactory and remove the adapter classes between these two factory types. Some code changes were necessary in order to accomplish this: * Move SimulcastEncoderAdapter from webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's where it's used). * Rename simulcast_unittest.h to simulcast_test_utility.h and make it into it's own target, because it's used from both simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc. * Remove ownership of the encoder factory from SimulcastEncoderAdapter, and make the necessary changes in surrounding code. The goal with this CL is to clean up the code, and also to free up the name webrtc::VideoEncoderFactory for future use. BUG=webrtc:7925 Review-Url: https://codereview.webrtc.org/2964953002 Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 03:26:36 -07:00
":video_coding",
":webrtc_vp8",
"../../api:video_frame_api",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Remove webrtc::VideoEncoderFactory Replace the use of webrtc::VideoEncoderFactory with cricket::WebRtcVideoEncoderFactory and remove the adapter classes between these two factory types. Some code changes were necessary in order to accomplish this: * Move SimulcastEncoderAdapter from webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's where it's used). * Rename simulcast_unittest.h to simulcast_test_utility.h and make it into it's own target, because it's used from both simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc. * Remove ownership of the encoder factory from SimulcastEncoderAdapter, and make the necessary changes in surrounding code. The goal with this CL is to clean up the code, and also to free up the name webrtc::VideoEncoderFactory for future use. BUG=webrtc:7925 Review-Url: https://codereview.webrtc.org/2964953002 Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 03:26:36 -07:00
"../../common_video:common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove webrtc::VideoEncoderFactory Replace the use of webrtc::VideoEncoderFactory with cricket::WebRtcVideoEncoderFactory and remove the adapter classes between these two factory types. Some code changes were necessary in order to accomplish this: * Move SimulcastEncoderAdapter from webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's where it's used). * Rename simulcast_unittest.h to simulcast_test_utility.h and make it into it's own target, because it's used from both simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc. * Remove ownership of the encoder factory from SimulcastEncoderAdapter, and make the necessary changes in surrounding code. The goal with this CL is to clean up the code, and also to free up the name webrtc::VideoEncoderFactory for future use. BUG=webrtc:7925 Review-Url: https://codereview.webrtc.org/2964953002 Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 03:26:36 -07:00
"../../test:test_support",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_source_set("video_codecs_test_framework") {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
testonly = true
sources = [
"codecs/test/mock/mock_packet_manipulator.h",
"codecs/test/packet_manipulator.cc",
"codecs/test/packet_manipulator.h",
"codecs/test/stats.cc",
"codecs/test/stats.h",
Move TestConfig to separate file. Move functions Set/PrintCodecSettings, NumberOfTemporalLayers to TestConfig. Add function NumberOfCores. Bug: none Change-Id: Ic33d79681d59d62bf34d9c9ff056a751ed3f8da8 Reviewed-on: https://webrtc-review.googlesource.com/13120 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20358}
2017-10-19 14:05:50 +02:00
"codecs/test/test_config.cc",
"codecs/test/test_config.h",
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
"codecs/test/video_codec_test.cc",
"codecs/test/video_codec_test.h",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"codecs/test/videoprocessor.cc",
"codecs/test/videoprocessor.h",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
deps = [
Adding test for SingleNalUnit mode Test enables single-nalu mode, sets limit for nalu lenght and verifies that encoder follows that limit. I found that QP jumps significantly when the mode is enabled. In result encoder might produce 4kbyte and 0.4kbyte frames back-to-back. But it seems that happens only to couple of frames in the beginning. This caused test to fail with default RC thresholds. To bypass this I increased frame size mismatch threshold from 20 to 30%. This should be Ok considering single-nalu mode is rare. BUG=webrtc:8070 Review-Url: https://codereview.webrtc.org/3014623002 Cr-Commit-Position: refs/heads/master@{#20023}
2017-09-28 09:23:17 -07:00
":codec_globals_headers",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
":video_coding",
":video_coding_utility",
Refactor + enable GN check on video_coding_utility To avoid the cyclic dependency BUG=webrtc:6828 NOTRY=True TBR=magjed@webrtc.org Review-Url: https://codereview.webrtc.org/2717113002 Cr-Commit-Position: refs/heads/master@{#17116}
2017-03-08 05:42:26 -08:00
":webrtc_vp8",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../..:webrtc_common",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../../common_video:common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
VideoProcessor: make it runnable on a task queue. * Guard members with a SequencedTaskChecker. * Intercept encoder/decoder callbacks, and post onto task queue if needed. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2996253002 Cr-Commit-Position: refs/heads/master@{#19428}
2017-08-21 06:44:16 -07:00
"../../rtc_base:rtc_task_queue",
"../../rtc_base:sequenced_task_checker",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"../../test:test_support",
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
"../../test:video_test_common",
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. Remove video_capture as a dependency of test_common and add it as a dependency of modules_unittests, as it was before the refactor in https://codereview.webrtc.org/2629923002 BUG=webrtc:7037 NOTRY=True Review-Url: https://codereview.webrtc.org/2666113003 Cr-Commit-Position: refs/heads/master@{#16439}
2017-02-06 02:21:11 -08:00
"../../test:video_test_support",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
}
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
video_coding_modules_tests_resources = [
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}
2017-09-15 06:47:31 +02:00
"../../resources/foreman_128x96.yuv",
"../../resources/foreman_160x120.yuv",
"../../resources/foreman_176x144.yuv",
"../../resources/foreman_320x240.yuv",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
]
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}
2017-02-20 04:35:52 -08:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
if (is_ios || is_mac) {
bundle_data("video_coding_modules_tests_resources_bundle_data") {
testonly = true
sources = video_coding_modules_tests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
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}
2017-02-20 04:35:52 -08:00
]
}
}
rtc_source_set("video_coding_modules_tests") {
testonly = true
sources = [
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
"codecs/h264/test/h264_impl_unittest.cc",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
"codecs/stereo/test/stereo_adapter_unittest.cc",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"codecs/test/videoprocessor_integrationtest.cc",
"codecs/test/videoprocessor_integrationtest.h",
Split up VideoProcessorIntegrationTest files. Changes made: * VideoProcessorIntegrationTest definition goes in .h file. * Member function definitions go into .cc file. * Tests move to _libvpx.cc and _openh264.cc files. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3008913002 Cr-Commit-Position: refs/heads/master@{#19637}
2017-09-01 00:10:27 -07:00
"codecs/test/videoprocessor_integrationtest_libvpx.cc",
"codecs/test/videoprocessor_integrationtest_openh264.cc",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"codecs/test/videoprocessor_integrationtest_parameterized.cc",
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}
2017-02-20 04:35:52 -08:00
"codecs/vp8/test/vp8_impl_unittest.cc",
Add QP for libvpx VP9 decoder. Also adds a basic unit test for VP9 implementation. BUG=webrtc:6541 Review-Url: https://codereview.webrtc.org/2654813002 Cr-Commit-Position: refs/heads/master@{#16795}
2017-02-23 02:54:57 -08:00
"codecs/vp9/test/vp9_impl_unittest.cc",
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}
2017-02-20 04:35:52 -08:00
]
deps = [
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
":video_codecs_test_framework",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
":video_coding",
":video_coding_utility",
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
":webrtc_h264",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
":webrtc_stereo",
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}
2017-02-20 04:35:52 -08:00
":webrtc_vp8",
Add QP for libvpx VP9 decoder. Also adds a basic unit test for VP9 implementation. BUG=webrtc:6541 Review-Url: https://codereview.webrtc.org/2654813002 Cr-Commit-Position: refs/heads/master@{#16795}
2017-02-23 02:54:57 -08:00
":webrtc_vp9",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"../..:webrtc_common",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
"../../api:mock_video_codec_factory",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../../api:video_frame_api",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"../../common_video",
"../../media:rtc_audio_video",
Add support for H.264 constrained high profile in VideoProcessor. BUG=webrtc:8448 Change-Id: I968d6cd78dd4f3c19a7944ae4cc73c5eddb9a949 Reviewed-on: https://webrtc-review.googlesource.com/16160 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20466}
2017-10-27 14:51:46 +02:00
"../../media:rtc_media_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Add cpu measurements to VideoProcessorIntegrationTest. Remove unused method ExcludeFrameTypesToStr. Bug: webrtc:6634 Change-Id: I2816466ed428b8ce13f3073ca496c2891d5d6368 Reviewed-on: https://webrtc-review.googlesource.com/9400 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20350}
2017-10-19 12:14:09 +02:00
"../../rtc_base:rtc_base_tests_utils",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"../../system_wrappers",
Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings. Make it possible for forced VP8 SW fallback encoder to set min_pixels_per_frame via GetScalingSettings(). Add a min required resolution (in addition to bitrate) before releasing forced SW fallback. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3000693003 Cr-Commit-Position: refs/heads/master@{#19390}
2017-08-17 08:58:54 -07:00
"../../test:field_trial",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../../test:test_support",
"../../test:video_test_common",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"../../test:video_test_support",
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. Remove video_capture as a dependency of test_common and add it as a dependency of modules_unittests, as it was before the refactor in https://codereview.webrtc.org/2629923002 BUG=webrtc:7037 NOTRY=True Review-Url: https://codereview.webrtc.org/2666113003 Cr-Commit-Position: refs/heads/master@{#16439}
2017-02-06 02:21:11 -08:00
"../video_capture",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
]
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}
2017-02-20 04:35:52 -08:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
data = video_coding_modules_tests_resources
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
if (is_android) {
Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." This reverts commit 2666cf7eba4bdd697d59d0451a8f74a05d4d207e. Reason for revert: On Lollipop Nexus 4, the 240p tests fail too. Original change's description: > Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). > > * Add audio_ prefix to CallTest::{en,de}coder_factory_. > * Let VideoQualityTest only instantiate encoders using encoder factories. > * Add HW encoder factories to VideoQualityTest. > * Add full stack tests: > - sqcif7 at 30 kbps: libvpx. > - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx. > > BUG=webrtc:8219 > TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74 > Reviewed-on: https://webrtc-review.googlesource.com/4740 > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20041} TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8219 Reviewed-on: https://webrtc-review.googlesource.com/4900 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20044}
2017-09-29 13:48:12 +00:00
sources += [
"codecs/test/android_test_initializer.cc",
"codecs/test/android_test_initializer.h",
"codecs/test/videoprocessor_integrationtest_mediacodec.cc",
]
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}
2017-02-20 04:35:52 -08:00
deps += [
Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." This reverts commit 2666cf7eba4bdd697d59d0451a8f74a05d4d207e. Reason for revert: On Lollipop Nexus 4, the 240p tests fail too. Original change's description: > Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). > > * Add audio_ prefix to CallTest::{en,de}coder_factory_. > * Let VideoQualityTest only instantiate encoders using encoder factories. > * Add HW encoder factories to VideoQualityTest. > * Add full stack tests: > - sqcif7 at 30 kbps: libvpx. > - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx. > > BUG=webrtc:8219 > TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74 > Reviewed-on: https://webrtc-review.googlesource.com/4740 > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20041} TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8219 Reviewed-on: https://webrtc-review.googlesource.com/4900 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20044}
2017-09-29 13:48:12 +00:00
"../../sdk/android:libjingle_peerconnection_jni",
"//base",
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}
2017-02-20 04:35:52 -08:00
]
}
if (is_ios || is_mac) {
Add VideoProcessor tests verifying that H.264 keyframes contain SPS/PPS/IDR. This CL adds an EncodedFrameChecker interface which can be used by users of the VideoProcessor to inject customized per-frame checks to the encoding/decoding pipeline. This currently has two uses: - Verifying that the QP parser works correctly for VP8 and VP9, by comparing the parsed QP to that produced by libvpx. - Verifying that our H.264 encoders always produce SPS/PPS/IDR in tandem. TESTED=Galaxy S8, Pixel 2 XL, iPhone 7. BUG=webrtc:8423 Change-Id: Ic3e401546e239a9ffaf2ed2907689cebb1127805 Reviewed-on: https://webrtc-review.googlesource.com/14559 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20409}
2017-10-24 10:16:33 +02:00
sources +=
[ "codecs/test/videoprocessor_integrationtest_videotoolbox.cc" ]
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
deps += [
Rename all objc targets to be suffixed for consistency BUG=webrtc:6412 Review-Url: https://codereview.webrtc.org/3001023003 Cr-Commit-Position: refs/heads/master@{#19489}
2017-08-24 05:43:45 -07:00
":codec_test_objc",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
":video_coding_modules_tests_resources_bundle_data",
Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." This reverts commit 2666cf7eba4bdd697d59d0451a8f74a05d4d207e. Reason for revert: On Lollipop Nexus 4, the 240p tests fail too. Original change's description: > Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). > > * Add audio_ prefix to CallTest::{en,de}coder_factory_. > * Let VideoQualityTest only instantiate encoders using encoder factories. > * Add HW encoder factories to VideoQualityTest. > * Add full stack tests: > - sqcif7 at 30 kbps: libvpx. > - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx. > > BUG=webrtc:8219 > TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74 > Reviewed-on: https://webrtc-review.googlesource.com/4740 > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20041} TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8219 Reviewed-on: https://webrtc-review.googlesource.com/4900 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20044}
2017-09-29 13:48:12 +00:00
"../../media:rtc_media_base",
"../../sdk:videotoolbox_objc",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
]
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}
2017-02-20 04:35:52 -08:00
}
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
if (rtc_use_h264) {
defines = [ "WEBRTC_USE_H264" ]
}
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}
2017-02-20 04:35:52 -08:00
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
rtc_source_set("video_coding_unittests") {
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
"codecs/test/packet_manipulator_unittest.cc",
"codecs/test/stats_unittest.cc",
Move TestConfig to separate file. Move functions Set/PrintCodecSettings, NumberOfTemporalLayers to TestConfig. Add function NumberOfCores. Bug: none Change-Id: Ic33d79681d59d62bf34d9c9ff056a751ed3f8da8 Reviewed-on: https://webrtc-review.googlesource.com/13120 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20358}
2017-10-19 14:05:50 +02:00
"codecs/test/test_config_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/test/videoprocessor_unittest.cc",
"codecs/vp8/default_temporal_layers_unittest.cc",
"codecs/vp8/screenshare_layers_unittest.cc",
"codecs/vp8/simulcast_unittest.cc",
"decoding_state_unittest.cc",
"frame_buffer2_unittest.cc",
Implement timing frames. Timing information is gathered in EncodedImage, starting at encoders. Then it's sent using RTP header extension. In the end, it's gathered at the GenericDecoder. Actual reporting and tests will be in the next CLs. BUG=webrtc:7594 Review-Url: https://codereview.webrtc.org/2911193002 Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 07:18:55 -07:00
"generic_encoder_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"h264_sprop_parameter_sets_unittest.cc",
"h264_sps_pps_tracker_unittest.cc",
"histogram_unittest.cc",
"include/mock/mock_vcm_callbacks.h",
"jitter_buffer_unittest.cc",
"jitter_estimator_tests.cc",
"nack_module_unittest.cc",
"protection_bitrate_calculator_unittest.cc",
"receiver_unittest.cc",
"rtp_frame_reference_finder_unittest.cc",
"session_info_unittest.cc",
"test/stream_generator.cc",
"test/stream_generator.h",
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
"test/test_util.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"timing_unittest.cc",
"utility/default_video_bitrate_allocator_unittest.cc",
"utility/frame_dropper_unittest.cc",
"utility/ivf_file_writer_unittest.cc",
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
"utility/mock/mock_frame_dropper.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"utility/moving_average_unittest.cc",
"utility/quality_scaler_unittest.cc",
"utility/simulcast_rate_allocator_unittest.cc",
Reland of Add experimental simulcast screen content mode The original CL was reverted because of a bug discovered by the chromium bots. Description of that CL: > Review-Url: https://codereview.webrtc.org/2636443002 > Cr-Commit-Position: refs/heads/master@{#16135} > Committed: https://chromium.googlesource.com/external/webrtc/+/a28e971e3bc8a06f366f0ef82cc7168b05a83b59 The first patch set of this CL is the same as r16135. Subsequence patch sets are the fixes applied. Some new test cases have been added, which reveal a few more bugs that have also been fixed. BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2641133002 Cr-Commit-Position: refs/heads/master@{#16299}
2017-01-26 06:12:26 -08:00
"video_codec_initializer_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"video_packet_buffer_unittest.cc",
"video_receiver_unittest.cc",
"video_sender_unittest.cc",
]
if (rtc_libvpx_build_vp9) {
sources += [ "codecs/vp9/vp9_screenshare_layers_unittest.cc" ]
}
if (rtc_use_h264) {
sources += [ "codecs/h264/h264_encoder_impl_unittest.cc" ]
}
deps = [
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
":mock_headers",
Remove webrtc::VideoEncoderFactory Replace the use of webrtc::VideoEncoderFactory with cricket::WebRtcVideoEncoderFactory and remove the adapter classes between these two factory types. Some code changes were necessary in order to accomplish this: * Move SimulcastEncoderAdapter from webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's where it's used). * Rename simulcast_unittest.h to simulcast_test_utility.h and make it into it's own target, because it's used from both simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc. * Remove ownership of the encoder factory from SimulcastEncoderAdapter, and make the necessary changes in surrounding code. The goal with this CL is to clean up the code, and also to free up the name webrtc::VideoEncoderFactory for future use. BUG=webrtc:7925 Review-Url: https://codereview.webrtc.org/2964953002 Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 03:26:36 -07:00
":simulcast_test_utility",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":video_codecs_test_framework",
":video_coding",
":video_coding_utility",
":webrtc_h264",
":webrtc_vp8",
":webrtc_vp9",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../..:webrtc_common",
"../../api:video_frame_api",
Separate i420 and i444 implementations to separate targets. This means we can properly declare the dependency between libjingle_peerconnection_api and video_frame_api. i420 pulls in system_wrappers, which can't be a dependency of the public API. Plan: 1) Land this CL + send out PSA 2) Make all direct users of i420_buffer depend on the new video_frame_api_i420 target 3) Move i420_buffer.cc to the new target 4) Make libjingle_peerconnection_api depend on video_frame_api, since it no longer contains i420 code Bug: webrtc:7504 Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9 Reviewed-on: https://webrtc-review.googlesource.com/9382 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 10:19:58 +01:00
"../../api:video_frame_api_i420",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../common_video:common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
Move sequence_number_utils.h to rtc_base/ Bug: webrtc:8440 Change-Id: I36e70da6ce70b95db7d3fce8b0013bff5c795bfc Reviewed-on: https://webrtc-review.googlesource.com/14860 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20429}
2017-10-25 13:07:09 +02:00
"../../rtc_base:rtc_numerics",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_task_queue",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../system_wrappers:metrics_default",
"../../test:field_trial",
"../../test:test_support",
"../../test:video_test_common",
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. Remove video_capture as a dependency of test_common and add it as a dependency of modules_unittests, as it was before the refactor in https://codereview.webrtc.org/2629923002 BUG=webrtc:7037 NOTRY=True Review-Url: https://codereview.webrtc.org/2666113003 Cr-Commit-Position: refs/heads/master@{#16439}
2017-02-06 02:21:11 -08:00
"../../test:video_test_support",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"//testing/gmock",
]
if (rtc_build_libvpx) {
deps += [ rtc_libvpx_dir ]
}
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
# TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
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