9aa3f0a200
Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed:35a32700fc> > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed:69dc7dbe24TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
82 lines
2.5 KiB
Plaintext
82 lines
2.5 KiB
Plaintext
# Copyright (c) 2015 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("../webrtc.gni")
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rtc_static_library("audio") {
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sources = [
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"audio_receive_stream.cc",
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.cc",
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"audio_state.h",
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"audio_transport_proxy.cc",
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"audio_transport_proxy.h",
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"conversion.h",
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"scoped_voe_interface.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"..:webrtc_common",
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"../api:audio_mixer_api",
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"../api:call_api",
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"../base:rtc_base_approved",
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"../base:rtc_task_queue",
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"../call:call_interfaces",
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"../common_audio",
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"../modules/audio_device",
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"../modules/audio_processing",
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"../modules/congestion_controller:congestion_controller",
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"../modules/pacing:pacing",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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"../modules/rtp_rtcp:rtp_rtcp",
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"../system_wrappers",
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"../voice_engine",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("audio_tests") {
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testonly = true
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# TODO(kjellander): Remove (bugs.webrtc.org/6828)
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# This needs remote_bitrate_estimator to be moved to webrtc/api first.
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check_includes = false
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sources = [
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"audio_receive_stream_unittest.cc",
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"audio_send_stream_unittest.cc",
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"audio_state_unittest.cc",
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]
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deps = [
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":audio",
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"../api:mock_audio_mixer",
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"../base:rtc_base_approved",
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"../base:rtc_task_queue",
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/congestion_controller:congestion_controller",
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"../modules/pacing:pacing",
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"../test:test_common",
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"../test:test_support",
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"utility:utility_tests",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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