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webrtc_m130/webrtc/modules/video_coding/BUILD.gn

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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.
import("../../build/webrtc.gni")
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",
"fec_tables_xor.h",
"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",
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 = [
":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",
":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",
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 smoothing filter to common audio and exp_filter to base/analytics. An earlier attempt of this work can be found here https://codereview.webrtc.org/2520003005/#ps100001, but was reverted. PS4 in that CL was not valid since separation of BUILD.gn can cause internal bot to fail. This is a new attempt, which is the same as https://codereview.webrtc.org/2520003005/#ps100001 but PS4 reverted. BUG=webrtc:6443 TBR=tommi@webrtc.org, solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/2532523002 Cr-Commit-Position: refs/heads/master@{#15233}
2016-11-24 11:01:09 -08:00
"../../base:rtc_analytics",
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
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../base:rtc_task_queue",
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",
"../../system_wrappers",
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
"../rtp_rtcp:rtp_rtcp",
"../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
]
}
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_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
# TODO(ehmaldonado): Remove (bugs.webrtc.org/6828)
# Errors on cyclic dependency with :video_coding if enabled.
check_includes = false
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 = [
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",
Add stats for average QP per frame for VP8 (for received video streams): "WebRTC.Video.Decoded.VP8.Qp" BUG=chromium:512752 Review URL: https://codereview.webrtc.org/1340623002 Cr-Commit-Position: refs/heads/master@{#10349}
2015-10-20 23:55:26 -07:00
"utility/qp_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/qp_parser.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",
Extract simulcast rate allocation outside of video encoder. This is a first step to refactor this code. I'm deprecating https://codereview.webrtc.org/1913073002 and implementing this in smaller more isolated steps. BUG=webrtc:5206 R=asapersson@webrtc.org, kjellander@webrtc.org, noahric@chromium.org Review URL: https://codereview.webrtc.org/2288223002 . Cr-Commit-Position: refs/heads/master@{#14186}
2016-09-12 16:04:43 +02:00
"utility/simulcast_rate_allocator.cc",
"utility/simulcast_rate_allocator.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 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 = [
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 smoothing filter to common audio and exp_filter to base/analytics. An earlier attempt of this work can be found here https://codereview.webrtc.org/2520003005/#ps100001, but was reverted. PS4 in that CL was not valid since separation of BUILD.gn can cause internal bot to fail. This is a new attempt, which is the same as https://codereview.webrtc.org/2520003005/#ps100001 but PS4 reverted. BUG=webrtc:6443 TBR=tommi@webrtc.org, solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/2532523002 Cr-Commit-Position: refs/heads/master@{#15233}
2016-11-24 11:01:09 -08:00
"../../base:rtc_analytics",
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
"../../base:rtc_base_approved",
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",
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",
]
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") {
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
# TODO(ehmaldonado): Remove (bugs.webrtc.org/6828)
# Errors on cyclic dependency if enabled:
# //webrtc/call:call ->
# //webrtc/video:video ->
# //webrtc/modules/video_coding:video_coding ->
# //webrtc/modules/video_coding:webrtc_h264 ->
# //webrtc/media:rtc_media ->
# //webrtc/call:call
check_includes = false
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",
"../../base:rtc_base_approved",
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
"../../system_wrappers",
]
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 = [
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",
"../../base:rtc_base_approved",
"../../common_video:common_video",
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",
]
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_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",
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/realtime_temporal_layers.cc",
"codecs/vp8/reference_picture_selection.cc",
"codecs/vp8/reference_picture_selection.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/screenshare_layers.cc",
"codecs/vp8/screenshare_layers.h",
"codecs/vp8/simulcast_encoder_adapter.cc",
"codecs/vp8/simulcast_encoder_adapter.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
"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",
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",
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
"../../base:rtc_base_approved",
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",
"../../system_wrappers",
]
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",
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
"../../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
"../../common_video",
"../../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) {
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/predictive_packet_manipulator.cc",
"codecs/test/predictive_packet_manipulator.h",
"codecs/test/stats.cc",
"codecs/test/stats.h",
"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 = [
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",
"../..:webrtc_common",
"../../base:rtc_base_approved",
"../../common_video:common_video",
"../../system_wrappers: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",
]
}
}
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