webrtc/api/audio_codecs/BUILD.gn
Patrik Höglund d37709b659 Revert "Fix circular dependencies between optional, array_view, and rtc_base."
This reverts commit a9e0924fa7.

Reason for revert: Breaks because of RTC_LAST_SYSTEM_ERROR

Original change's description:
> Fix circular dependencies between optional, array_view, and rtc_base.
> 
> This splits things out of rtc_base and makes dependencies explicit.
> 
> Bug: webrtc:6828
> Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
> Reviewed-on: https://webrtc-review.googlesource.com/31940
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21245}

TBR=phoglund@webrtc.org,kwiberg@webrtc.org

Change-Id: I1a5dcf2223f00ae7c46f9f2a12b990ab3a84397d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6828
Reviewed-on: https://webrtc-review.googlesource.com/32760
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21251}
2017-12-13 14:56:33 +00:00

90 lines
2.4 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_source_set("audio_codecs_api") {
sources = [
"audio_decoder.cc",
"audio_decoder.h",
"audio_decoder_factory.h",
"audio_decoder_factory_template.h",
"audio_encoder.cc",
"audio_encoder.h",
"audio_encoder_factory.h",
"audio_encoder_factory_template.h",
"audio_format.cc",
"audio_format.h",
]
deps = [
"..:array_view",
"..:optional",
"../..:webrtc_common",
"../../rtc_base:rtc_base_approved",
]
}
rtc_static_library("builtin_audio_decoder_factory") {
sources = [
"builtin_audio_decoder_factory.cc",
"builtin_audio_decoder_factory.h",
]
deps = [
":audio_codecs_api",
"../../rtc_base:rtc_base_approved",
"L16:audio_decoder_L16",
"g711:audio_decoder_g711",
"g722:audio_decoder_g722",
"isac:audio_decoder_isac",
]
defines = []
if (rtc_include_ilbc) {
deps += [ "ilbc:audio_decoder_ilbc" ]
defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
} else {
defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
}
if (rtc_include_opus) {
deps += [ "opus:audio_decoder_opus" ]
defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
} else {
defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
}
}
rtc_static_library("builtin_audio_encoder_factory") {
sources = [
"builtin_audio_encoder_factory.cc",
"builtin_audio_encoder_factory.h",
]
deps = [
":audio_codecs_api",
"../../rtc_base:rtc_base_approved",
"L16:audio_encoder_L16",
"g711:audio_encoder_g711",
"g722:audio_encoder_g722",
"isac:audio_encoder_isac",
]
defines = []
if (rtc_include_ilbc) {
deps += [ "ilbc:audio_encoder_ilbc" ]
defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
} else {
defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
}
if (rtc_include_opus) {
deps += [ "opus:audio_encoder_opus" ]
defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
} else {
defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
}
}