From 1227ab89a7c08e4e5af051a63daba889ea0d2da7 Mon Sep 17 00:00:00 2001 From: "kjellander@webrtc.org" Date: Mon, 23 Jun 2014 19:21:07 +0000 Subject: 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 --- webrtc/modules/rtp_rtcp/BUILD.gn | 104 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 104 insertions(+) create mode 100644 webrtc/modules/rtp_rtcp/BUILD.gn (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn new file mode 100644 index 0000000000..e9a2e8d913 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -0,0 +1,104 @@ +# 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") + +source_set("rtp_rtcp") { + sources = [ + # Common + "interface/fec_receiver.h", + "interface/receive_statistics.h", + "interface/remote_ntp_time_estimator.h", + "interface/rtp_header_parser.h", + "interface/rtp_payload_registry.h", + "interface/rtp_receiver.h", + "interface/rtp_rtcp.h", + "interface/rtp_rtcp_defines.h", + "source/bitrate.cc", + "source/bitrate.h", + "source/byte_io.h", + "source/fec_receiver_impl.cc", + "source/fec_receiver_impl.h", + "source/receive_statistics_impl.cc", + "source/receive_statistics_impl.h", + "source/remote_ntp_time_estimator.cc", + "source/rtp_header_parser.cc", + "source/rtp_rtcp_config.h", + "source/rtp_rtcp_impl.cc", + "source/rtp_rtcp_impl.h", + "source/rtcp_packet.cc", + "source/rtcp_packet.h", + "source/rtcp_receiver.cc", + "source/rtcp_receiver.h", + "source/rtcp_receiver_help.cc", + "source/rtcp_receiver_help.h", + "source/rtcp_sender.cc", + "source/rtcp_sender.h", + "source/rtcp_utility.cc", + "source/rtcp_utility.h", + "source/rtp_header_extension.cc", + "source/rtp_header_extension.h", + "source/rtp_receiver_impl.cc", + "source/rtp_receiver_impl.h", + "source/rtp_sender.cc", + "source/rtp_sender.h", + "source/rtp_utility.cc", + "source/rtp_utility.h", + "source/ssrc_database.cc", + "source/ssrc_database.h", + "source/tmmbr_help.cc", + "source/tmmbr_help.h", + # Audio Files + "source/dtmf_queue.cc", + "source/dtmf_queue.h", + "source/rtp_receiver_audio.cc", + "source/rtp_receiver_audio.h", + "source/rtp_sender_audio.cc", + "source/rtp_sender_audio.h", + # Video Files + "source/fec_private_tables_random.h", + "source/fec_private_tables_bursty.h", + "source/forward_error_correction.cc", + "source/forward_error_correction.h", + "source/forward_error_correction_internal.cc", + "source/forward_error_correction_internal.h", + "source/producer_fec.cc", + "source/producer_fec.h", + "source/rtp_packet_history.cc", + "source/rtp_packet_history.h", + "source/rtp_payload_registry.cc", + "source/rtp_receiver_strategy.cc", + "source/rtp_receiver_strategy.h", + "source/rtp_receiver_video.cc", + "source/rtp_receiver_video.h", + "source/rtp_sender_video.cc", + "source/rtp_sender_video.h", + "source/video_codec_information.h", + "source/rtp_format_vp8.cc", + "source/rtp_format_vp8.h", + "source/rtp_format_video_generic.h", + "source/vp8_partition_aggregator.cc", + "source/vp8_partition_aggregator.h", + # Mocks + "mocks/mock_rtp_rtcp.h", + "source/mock/mock_rtp_payload_strategy.h", + ] + + deps = [ + "../../system_wrappers", + "../pacing", + "../remote_bitrate_estimator", + ] + + if (is_win) { + cflags = [ + # TODO(jschuh): Bug 1348: fix this warning. + "/wd4267", # size_t to int truncations + ] + } +} -- cgit v1.2.3 From 2ec560606be6519dc4e32a1e6855b0f362ca498d Mon Sep 17 00:00:00 2001 From: "stefan@webrtc.org" Date: Thu, 31 Jul 2014 14:59:24 +0000 Subject: Add H.264 packetization. This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/rtp_rtcp/BUILD.gn | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index e9a2e8d913..a7499bc986 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -79,6 +79,10 @@ source_set("rtp_rtcp") { "source/rtp_sender_video.cc", "source/rtp_sender_video.h", "source/video_codec_information.h", + 'source/rtp_format.cc', + 'source/rtp_format.h', + 'source/rtp_format_h264.cc', + 'source/rtp_format_h264.h', "source/rtp_format_vp8.cc", "source/rtp_format_vp8.h", "source/rtp_format_video_generic.h", -- cgit v1.2.3 From e1c9caf6eee2d97824d2ecae75dbd5aae2f0a3b4 Mon Sep 17 00:00:00 2001 From: "stefan@webrtc.org" Date: Thu, 31 Jul 2014 15:07:59 +0000 Subject: Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. TEST=buildtools/linux64/gn gen out/Default TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/rtp_rtcp/BUILD.gn | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index a7499bc986..09fa3aca39 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -79,10 +79,10 @@ source_set("rtp_rtcp") { "source/rtp_sender_video.cc", "source/rtp_sender_video.h", "source/video_codec_information.h", - 'source/rtp_format.cc', - 'source/rtp_format.h', - 'source/rtp_format_h264.cc', - 'source/rtp_format_h264.h', + "source/rtp_format.cc", + "source/rtp_format.h", + "source/rtp_format_h264.cc", + "source/rtp_format_h264.h", "source/rtp_format_vp8.cc", "source/rtp_format_vp8.h", "source/rtp_format_video_generic.h", -- cgit v1.2.3 From 42ee5b54b59d766066bac540c3e8ddf7d49b649f Mon Sep 17 00:00:00 2001 From: "kjellander@webrtc.org" Date: Mon, 25 Aug 2014 14:15:35 +0000 Subject: 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 --- webrtc/modules/rtp_rtcp/BUILD.gn | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 09fa3aca39..74af071a5e 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -93,6 +93,12 @@ source_set("rtp_rtcp") { "source/mock/mock_rtp_payload_strategy.h", ] + if (is_clang) { + # Suppress warnings from Chrome's Clang plugins. + # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. + configs -= [ "//build/config/clang:find_bad_constructs" ] + } + deps = [ "../../system_wrappers", "../pacing", -- cgit v1.2.3 From b5e6bfc76a32a588da2400636688d34a71a2f47d Mon Sep 17 00:00:00 2001 From: "pbos@webrtc.org" Date: Fri, 12 Sep 2014 11:05:55 +0000 Subject: Make RTPSender/RTPReceiver generic. Changes include, 1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric. 2) Introduce class RtpDepacketizerVp8. 3) Make RTPSenderVideo::SendH264 generic and used by all packetizers. 4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to RtpPacketizer/RtpDePacketizer sub-classes. R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26399004 Patch from Changbin Shao . git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/rtp_rtcp/BUILD.gn | 1 + 1 file changed, 1 insertion(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 74af071a5e..2ee1a8f0d5 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -85,6 +85,7 @@ source_set("rtp_rtcp") { "source/rtp_format_h264.h", "source/rtp_format_vp8.cc", "source/rtp_format_vp8.h", + "source/rtp_format_video_generic.cc", "source/rtp_format_video_generic.h", "source/vp8_partition_aggregator.cc", "source/vp8_partition_aggregator.h", -- cgit v1.2.3 From f21ea918ad9e4dcbe7f372fd32d130c082641e36 Mon Sep 17 00:00:00 2001 From: "kjellander@webrtc.org" Date: Sun, 28 Sep 2014 17:37:22 +0000 Subject: GN: Add common configs to all targets. This is needed to ensure we have the same build with GN as with GYP, since GYP includes the common.gypi on a global level. Several fixes has been needed in the past because some code have been built without the right defines. BUG=3441 R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/28589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/rtp_rtcp/BUILD.gn | 3 +++ 1 file changed, 3 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 2ee1a8f0d5..65d5f38947 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -94,6 +94,9 @@ source_set("rtp_rtcp") { "source/mock/mock_rtp_payload_strategy.h", ] + configs += [ "../..:common_config" ] + public_configs = [ "../..:common_inherited_config" ] + if (is_clang) { # Suppress warnings from Chrome's Clang plugins. # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. -- cgit v1.2.3 From 8649fed1b83882d2f25d3c58a3464a0a59a22225 Mon Sep 17 00:00:00 2001 From: "kjellander@webrtc.org" Date: Thu, 8 Jan 2015 21:22:01 +0000 Subject: 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 --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 ++ 1 file changed, 2 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 65d5f38947..f343e098c8 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -113,6 +113,8 @@ source_set("rtp_rtcp") { cflags = [ # TODO(jschuh): Bug 1348: fix this warning. "/wd4267", # size_t to int truncations + # TODO(kjellander): Bug 261: fix this warning. + "/wd4373", # virtual function override. ] } } -- cgit v1.2.3 From ac2d27d9ae74eb8d28ec0d5f12f70fa64461ab90 Mon Sep 17 00:00:00 2001 From: "kwiberg@webrtc.org" Date: Thu, 26 Feb 2015 13:59:22 +0000 Subject: 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 --- webrtc/modules/rtp_rtcp/BUILD.gn | 1 + 1 file changed, 1 insertion(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index f343e098c8..528637fe59 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -104,6 +104,7 @@ source_set("rtp_rtcp") { } deps = [ + "../..:webrtc_common", "../../system_wrappers", "../pacing", "../remote_bitrate_estimator", -- cgit v1.2.3 From 5ea8eff55ec21a1d81aaf7d29c0106fe13256150 Mon Sep 17 00:00:00 2001 From: Noah Richards Date: Thu, 23 Apr 2015 16:45:56 -0700 Subject: Add h.264 AVC SPS parsing for resolution. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48129004 Cr-Commit-Position: refs/heads/master@{#9073} --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 ++ 1 file changed, 2 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 528637fe59..bbf69f4f2d 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -67,6 +67,8 @@ source_set("rtp_rtcp") { "source/forward_error_correction.h", "source/forward_error_correction_internal.cc", "source/forward_error_correction_internal.h", + "source/h264_sps_parser.cc", + "source/h264_sps_parser.h", "source/producer_fec.cc", "source/producer_fec.h", "source/rtp_packet_history.cc", -- cgit v1.2.3 From e3827f27c393d31c919142928f50ce04b09636c6 Mon Sep 17 00:00:00 2001 From: Noah Richards Date: Thu, 23 Apr 2015 18:15:19 -0700 Subject: Revert "Add h.264 AVC SPS parsing for resolution." The Mac64 Debug builder is broken for an unknown failure (trybot is green, no failure obvious in the commit break). Reverting this CL to see if it goes green again, and then relanding to see if it is just some weird flaky build issue. This reverts commit 5ea8eff55ec21a1d81aaf7d29c0106fe13256150. BUG= TBR=rollback Review URL: https://webrtc-codereview.appspot.com/47019004 Cr-Commit-Position: refs/heads/master@{#9074} --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 -- 1 file changed, 2 deletions(-) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index bbf69f4f2d..528637fe59 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -67,8 +67,6 @@ source_set("rtp_rtcp") { "source/forward_error_correction.h", "source/forward_error_correction_internal.cc", "source/forward_error_correction_internal.h", - "source/h264_sps_parser.cc", - "source/h264_sps_parser.h", "source/producer_fec.cc", "source/producer_fec.h", "source/rtp_packet_history.cc", -- cgit v1.2.3 From f955b5d3f52db0f7456bd6c6bd4068d3599967da Mon Sep 17 00:00:00 2001 From: Henrik Kjellander Date: Fri, 24 Apr 2015 13:57:06 +0200 Subject: Add h.264 AVC SPS parsing for resolution (re-land) Re-land of noharic@'s CL at https://webrtc-codereview.appspot.com/48129004 which was reverted due to a Mac compile error which most likely was a Goma flake (it passed on all trybots). TBR=stefan@webrtc.org, noharic@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44329005 Cr-Commit-Position: refs/heads/master@{#9079} --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 ++ 1 file changed, 2 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 528637fe59..bbf69f4f2d 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -67,6 +67,8 @@ source_set("rtp_rtcp") { "source/forward_error_correction.h", "source/forward_error_correction_internal.cc", "source/forward_error_correction_internal.h", + "source/h264_sps_parser.cc", + "source/h264_sps_parser.h", "source/producer_fec.cc", "source/producer_fec.h", "source/rtp_packet_history.cc", -- cgit v1.2.3 From 57e5fd2e604ff7e60425c3f7654b40da03fc763c Mon Sep 17 00:00:00 2001 From: Henrik Kjellander Date: Mon, 25 May 2015 12:55:39 +0200 Subject: 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} --- webrtc/modules/rtp_rtcp/BUILD.gn | 83 +++++++++++++++++++--------------------- 1 file changed, 40 insertions(+), 43 deletions(-) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index bbf69f4f2d..0eda287a4c 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -10,7 +10,6 @@ import("../../build/webrtc.gni") source_set("rtp_rtcp") { sources = [ - # Common "interface/fec_receiver.h", "interface/receive_statistics.h", "interface/remote_ntp_time_estimator.h", @@ -19,18 +18,28 @@ source_set("rtp_rtcp") { "interface/rtp_receiver.h", "interface/rtp_rtcp.h", "interface/rtp_rtcp_defines.h", + "mocks/mock_rtp_rtcp.h", "source/bitrate.cc", "source/bitrate.h", "source/byte_io.h", + "source/dtmf_queue.cc", + "source/dtmf_queue.h", + "source/fec_private_tables_bursty.h", + "source/fec_private_tables_random.h", "source/fec_receiver_impl.cc", "source/fec_receiver_impl.h", + "source/forward_error_correction.cc", + "source/forward_error_correction.h", + "source/forward_error_correction_internal.cc", + "source/forward_error_correction_internal.h", + "source/h264_sps_parser.cc", + "source/h264_sps_parser.h", + "source/mock/mock_rtp_payload_strategy.h", + "source/producer_fec.cc", + "source/producer_fec.h", "source/receive_statistics_impl.cc", "source/receive_statistics_impl.h", "source/remote_ntp_time_estimator.cc", - "source/rtp_header_parser.cc", - "source/rtp_rtcp_config.h", - "source/rtp_rtcp_impl.cc", - "source/rtp_rtcp_impl.h", "source/rtcp_packet.cc", "source/rtcp_packet.h", "source/rtcp_receiver.cc", @@ -41,59 +50,46 @@ source_set("rtp_rtcp") { "source/rtcp_sender.h", "source/rtcp_utility.cc", "source/rtcp_utility.h", + "source/rtp_format.cc", + "source/rtp_format.h", + "source/rtp_format_h264.cc", + "source/rtp_format_h264.h", + "source/rtp_format_video_generic.cc", + "source/rtp_format_video_generic.h", + "source/rtp_format_vp8.cc", + "source/rtp_format_vp8.h", "source/rtp_header_extension.cc", "source/rtp_header_extension.h", + "source/rtp_header_parser.cc", + "source/rtp_packet_history.cc", + "source/rtp_packet_history.h", + "source/rtp_payload_registry.cc", + "source/rtp_receiver_audio.cc", + "source/rtp_receiver_audio.h", "source/rtp_receiver_impl.cc", "source/rtp_receiver_impl.h", + "source/rtp_receiver_strategy.cc", + "source/rtp_receiver_strategy.h", + "source/rtp_receiver_video.cc", + "source/rtp_receiver_video.h", + "source/rtp_rtcp_config.h", + "source/rtp_rtcp_impl.cc", + "source/rtp_rtcp_impl.h", "source/rtp_sender.cc", "source/rtp_sender.h", + "source/rtp_sender_audio.cc", + "source/rtp_sender_audio.h", + "source/rtp_sender_video.cc", + "source/rtp_sender_video.h", "source/rtp_utility.cc", "source/rtp_utility.h", "source/ssrc_database.cc", "source/ssrc_database.h", "source/tmmbr_help.cc", "source/tmmbr_help.h", - # Audio Files - "source/dtmf_queue.cc", - "source/dtmf_queue.h", - "source/rtp_receiver_audio.cc", - "source/rtp_receiver_audio.h", - "source/rtp_sender_audio.cc", - "source/rtp_sender_audio.h", - # Video Files - "source/fec_private_tables_random.h", - "source/fec_private_tables_bursty.h", - "source/forward_error_correction.cc", - "source/forward_error_correction.h", - "source/forward_error_correction_internal.cc", - "source/forward_error_correction_internal.h", - "source/h264_sps_parser.cc", - "source/h264_sps_parser.h", - "source/producer_fec.cc", - "source/producer_fec.h", - "source/rtp_packet_history.cc", - "source/rtp_packet_history.h", - "source/rtp_payload_registry.cc", - "source/rtp_receiver_strategy.cc", - "source/rtp_receiver_strategy.h", - "source/rtp_receiver_video.cc", - "source/rtp_receiver_video.h", - "source/rtp_sender_video.cc", - "source/rtp_sender_video.h", "source/video_codec_information.h", - "source/rtp_format.cc", - "source/rtp_format.h", - "source/rtp_format_h264.cc", - "source/rtp_format_h264.h", - "source/rtp_format_vp8.cc", - "source/rtp_format_vp8.h", - "source/rtp_format_video_generic.cc", - "source/rtp_format_video_generic.h", "source/vp8_partition_aggregator.cc", "source/vp8_partition_aggregator.h", - # Mocks - "mocks/mock_rtp_rtcp.h", - "source/mock/mock_rtp_payload_strategy.h", ] configs += [ "../..:common_config" ] @@ -116,6 +112,7 @@ source_set("rtp_rtcp") { cflags = [ # TODO(jschuh): Bug 1348: fix this warning. "/wd4267", # size_t to int truncations + # TODO(kjellander): Bug 261: fix this warning. "/wd4373", # virtual function override. ] -- cgit v1.2.3 From 30409b4dca3d9cfdb0e714a5932b135becb0f822 Mon Sep 17 00:00:00 2001 From: bcornell Date: Fri, 10 Jul 2015 18:10:05 -0700 Subject: Add statistics gathering for packet loss. Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics. BUG= Review URL: https://codereview.webrtc.org/1198853004 Cr-Commit-Position: refs/heads/master@{#9568} --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 ++ 1 file changed, 2 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 0eda287a4c..ca4b812731 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -35,6 +35,8 @@ source_set("rtp_rtcp") { "source/h264_sps_parser.cc", "source/h264_sps_parser.h", "source/mock/mock_rtp_payload_strategy.h", + "source/packet_loss_stats.cc", + "source/packet_loss_stats.h", "source/producer_fec.cc", "source/producer_fec.h", "source/receive_statistics_impl.cc", -- cgit v1.2.3 From f38ea3caa39887c63e7d4862dcf420d4a35c1073 Mon Sep 17 00:00:00 2001 From: asapersson Date: Tue, 28 Jul 2015 04:02:54 -0700 Subject: Add support for VP9 packetization/depacketization. RTP payload format for VP9: https://www.ietf.org/id/draft-uberti-payload-vp9-01.txt BUG=webrtc:4148, webrtc:4168, chromium:500602 TBR=mflodman Review URL: https://codereview.webrtc.org/1232023006 Cr-Commit-Position: refs/heads/master@{#9649} --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 ++ 1 file changed, 2 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index ca4b812731..48c4921ca5 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -60,6 +60,8 @@ source_set("rtp_rtcp") { "source/rtp_format_video_generic.h", "source/rtp_format_vp8.cc", "source/rtp_format_vp8.h", + "source/rtp_format_vp9.cc", + "source/rtp_format_vp9.h", "source/rtp_header_extension.cc", "source/rtp_header_extension.h", "source/rtp_header_parser.cc", -- cgit v1.2.3 From a3b8769860bdb0a45dbff6d1e0092486fa59aaa4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= Date: Wed, 29 Jul 2015 10:46:54 +0200 Subject: Add packetization and coding/decoding of feedback message format. BUG=webrtc:4312 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1175263002 . Cr-Commit-Position: refs/heads/master@{#9651} --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 ++ 1 file changed, 2 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 48c4921ca5..b83faae8d6 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -44,6 +44,8 @@ source_set("rtp_rtcp") { "source/remote_ntp_time_estimator.cc", "source/rtcp_packet.cc", "source/rtcp_packet.h", + "source/rtcp_packet/transport_feedback.cc", + "source/rtcp_packet/transport_feedback.h", "source/rtcp_receiver.cc", "source/rtcp_receiver.h", "source/rtcp_receiver_help.cc", -- cgit v1.2.3 From ebbf8a805b45613b4cb118e4eb0cebe7eeee69ac Mon Sep 17 00:00:00 2001 From: sprang Date: Mon, 21 Sep 2015 15:11:14 -0700 Subject: Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. BUG= Review URL: https://codereview.webrtc.org/1350163005 Cr-Commit-Position: refs/heads/master@{#10005} --- webrtc/modules/rtp_rtcp/BUILD.gn | 1 - 1 file changed, 1 deletion(-) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index b83faae8d6..af2d2c3cd4 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -110,7 +110,6 @@ source_set("rtp_rtcp") { deps = [ "../..:webrtc_common", "../../system_wrappers", - "../pacing", "../remote_bitrate_estimator", ] -- cgit v1.2.3 From 8c266e6baff043a1fa5c9134f46042908a376d5b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Peter=20Bostr=C3=B6m?= Date: Thu, 24 Sep 2015 15:06:50 +0200 Subject: H264 bitstream parser. Parsing the encoded bitstream is required for doing downscaling decisions based on average encoded QP to improve perceived quality. BUG=webrtc:4968 R=noahric@chromium.org, stefan@webrtc.org TBR=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1314473008 . Cr-Commit-Position: refs/heads/master@{#10051} --- webrtc/modules/rtp_rtcp/BUILD.gn | 2 ++ 1 file changed, 2 insertions(+) (limited to 'webrtc/modules/rtp_rtcp/BUILD.gn') diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index af2d2c3cd4..c651424611 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -32,6 +32,8 @@ source_set("rtp_rtcp") { "source/forward_error_correction.h", "source/forward_error_correction_internal.cc", "source/forward_error_correction_internal.h", + "source/h264_bitstream_parser.cc", + "source/h264_bitstream_parser.h", "source/h264_sps_parser.cc", "source/h264_sps_parser.h", "source/mock/mock_rtp_payload_strategy.h", -- cgit v1.2.3