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3e113438b1
One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
94 lines
2.5 KiB
Text
94 lines
2.5 KiB
Text
# Copyright (c) 2017 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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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_source_set("audio_codecs_api") {
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sources = [
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"audio_decoder.cc",
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"audio_decoder.h",
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"audio_decoder_factory.h",
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"audio_decoder_factory_template.h",
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"audio_encoder.cc",
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"audio_encoder.h",
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"audio_encoder_factory.h",
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"audio_encoder_factory_template.h",
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"audio_format.cc",
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"audio_format.h",
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]
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deps = [
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"..:array_view",
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"..:optional",
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"../..:webrtc_common",
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"../../:typedefs",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:sanitizer",
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]
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}
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rtc_static_library("builtin_audio_decoder_factory") {
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sources = [
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"builtin_audio_decoder_factory.cc",
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"builtin_audio_decoder_factory.h",
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]
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deps = [
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":audio_codecs_api",
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"../../rtc_base:rtc_base_approved",
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"L16:audio_decoder_L16",
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"g711:audio_decoder_g711",
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"g722:audio_decoder_g722",
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"isac:audio_decoder_isac",
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]
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defines = []
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if (rtc_include_ilbc) {
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deps += [ "ilbc:audio_decoder_ilbc" ]
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
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}
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if (rtc_include_opus) {
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deps += [ "opus:audio_decoder_opus" ]
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
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}
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}
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rtc_static_library("builtin_audio_encoder_factory") {
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sources = [
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"builtin_audio_encoder_factory.cc",
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"builtin_audio_encoder_factory.h",
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]
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deps = [
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":audio_codecs_api",
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"../../rtc_base:rtc_base_approved",
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"L16:audio_encoder_L16",
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"g711:audio_encoder_g711",
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"g722:audio_encoder_g722",
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"isac:audio_encoder_isac",
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]
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defines = []
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if (rtc_include_ilbc) {
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deps += [ "ilbc:audio_encoder_ilbc" ]
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
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}
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if (rtc_include_opus) {
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deps += [ "opus:audio_encoder_opus" ]
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
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}
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}
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