mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00

Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
116 lines
3.2 KiB
Text
116 lines
3.2 KiB
Text
# Copyright (c) 2017 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("../../../webrtc.gni")
|
|
if (is_android) {
|
|
import("//build/config/android/config.gni")
|
|
import("//build/config/android/rules.gni")
|
|
}
|
|
|
|
rtc_library("audio_encoder_opus_config") {
|
|
visibility = [ "*" ]
|
|
sources = [
|
|
"audio_encoder_multi_channel_opus_config.cc",
|
|
"audio_encoder_multi_channel_opus_config.h",
|
|
"audio_encoder_opus_config.cc",
|
|
"audio_encoder_opus_config.h",
|
|
]
|
|
deps = [
|
|
"../../../rtc_base:rtc_base_approved",
|
|
"../../../rtc_base/system:rtc_export",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
defines = []
|
|
if (rtc_opus_variable_complexity) {
|
|
defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
|
|
} else {
|
|
defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("audio_decoder_opus_config") {
|
|
visibility = [ "*" ]
|
|
sources = [
|
|
"audio_decoder_multi_channel_opus_config.h",
|
|
]
|
|
}
|
|
|
|
rtc_library("audio_encoder_opus") {
|
|
visibility = [ "*" ]
|
|
poisonous = [ "audio_codecs" ]
|
|
public = [
|
|
"audio_encoder_opus.h",
|
|
]
|
|
sources = [
|
|
"audio_encoder_opus.cc",
|
|
]
|
|
deps = [
|
|
":audio_encoder_opus_config",
|
|
"..:audio_codecs_api",
|
|
"../../../modules/audio_coding:webrtc_opus",
|
|
"../../../rtc_base:rtc_base_approved",
|
|
"../../../rtc_base/system:rtc_export",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|
|
|
|
rtc_library("audio_decoder_opus") {
|
|
visibility = [ "*" ]
|
|
poisonous = [ "audio_codecs" ]
|
|
sources = [
|
|
"audio_decoder_opus.cc",
|
|
"audio_decoder_opus.h",
|
|
]
|
|
deps = [
|
|
"..:audio_codecs_api",
|
|
"../../../modules/audio_coding:webrtc_opus",
|
|
"../../../rtc_base:rtc_base_approved",
|
|
"../../../rtc_base/system:rtc_export",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|
|
|
|
rtc_library("audio_encoder_multiopus") {
|
|
visibility = [ "*" ]
|
|
poisonous = [ "audio_codecs" ]
|
|
public = [
|
|
"audio_encoder_multi_channel_opus.h",
|
|
]
|
|
sources = [
|
|
"audio_encoder_multi_channel_opus.cc",
|
|
]
|
|
deps = [
|
|
"..:audio_codecs_api",
|
|
"../../../modules/audio_coding:webrtc_multiopus",
|
|
"../../../rtc_base:rtc_base_approved",
|
|
"../../../rtc_base/system:rtc_export",
|
|
"../opus:audio_encoder_opus_config",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|
|
|
|
rtc_library("audio_decoder_multiopus") {
|
|
visibility = [ "*" ]
|
|
poisonous = [ "audio_codecs" ]
|
|
sources = [
|
|
"audio_decoder_multi_channel_opus.cc",
|
|
"audio_decoder_multi_channel_opus.h",
|
|
]
|
|
deps = [
|
|
":audio_decoder_opus_config",
|
|
"..:audio_codecs_api",
|
|
"../../../modules/audio_coding:webrtc_multiopus",
|
|
"../../../rtc_base:rtc_base_approved",
|
|
"../../../rtc_base/system:rtc_export",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|