524b8f7304
NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
106 lines
2.8 KiB
Plaintext
106 lines
2.8 KiB
Plaintext
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../build/webrtc.gni")
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source_set("voice_engine") {
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sources = [
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"../common_types.h",
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"../engine_configurations.h",
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"../typedefs.h",
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"include/voe_audio_processing.h",
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"include/voe_base.h",
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"include/voe_codec.h",
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"include/voe_dtmf.h",
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"include/voe_errors.h",
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"include/voe_external_media.h",
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"include/voe_file.h",
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"include/voe_hardware.h",
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"include/voe_neteq_stats.h",
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"include/voe_network.h",
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"include/voe_rtp_rtcp.h",
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"include/voe_video_sync.h",
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"include/voe_volume_control.h",
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"channel.cc",
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"channel.h",
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"channel_manager.cc",
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"channel_manager.h",
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"dtmf_inband.cc",
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"dtmf_inband.h",
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"dtmf_inband_queue.cc",
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"dtmf_inband_queue.h",
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"level_indicator.cc",
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"level_indicator.h",
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"monitor_module.cc",
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"monitor_module.h",
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"network_predictor.cc",
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"network_predictor.h",
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"output_mixer.cc",
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"output_mixer.h",
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"shared_data.cc",
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"shared_data.h",
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"statistics.cc",
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"statistics.h",
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"transmit_mixer.cc",
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"transmit_mixer.h",
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"utility.cc",
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"utility.h",
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"voe_audio_processing_impl.cc",
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"voe_audio_processing_impl.h",
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"voe_base_impl.cc",
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"voe_base_impl.h",
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"voe_codec_impl.cc",
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"voe_codec_impl.h",
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"voe_dtmf_impl.cc",
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"voe_dtmf_impl.h",
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"voe_external_media_impl.cc",
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"voe_external_media_impl.h",
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"voe_file_impl.cc",
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"voe_file_impl.h",
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"voe_hardware_impl.cc",
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"voe_hardware_impl.h",
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"voe_neteq_stats_impl.cc",
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"voe_neteq_stats_impl.h",
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"voe_network_impl.cc",
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"voe_network_impl.h",
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"voe_rtp_rtcp_impl.cc",
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"voe_rtp_rtcp_impl.h",
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"voe_video_sync_impl.cc",
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"voe_video_sync_impl.h",
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"voe_volume_control_impl.cc",
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"voe_volume_control_impl.h",
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"voice_engine_defines.h",
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"voice_engine_impl.cc",
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"voice_engine_impl.h",
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]
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if (is_win) {
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defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
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}
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if (is_clang) {
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# Suppress warnings from Chrome's Clang plugins.
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# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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configs -= [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../common_audio",
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"../modules/audio_coding",
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"../modules/audio_conference_mixer",
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"../modules/audio_device",
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"../modules/audio_processing",
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"../modules/bitrate_controller",
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"../modules/media_file",
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"../modules/rtp_rtcp",
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"../modules/utility",
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"../system_wrappers",
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]
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}
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