webrtc/api/audio_codecs/opus/BUILD.gn
Mirko Bonadei 86d053c2db Use source_sets in component builds and static_library in release builds.
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}
2019-10-17 21:17:18 +00:00

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# 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",
]
}