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https://github.com/mollyim/webrtc.git
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0afde7614d
Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
102 lines
2.8 KiB
Text
102 lines
2.8 KiB
Text
# Copyright(c) 2020 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_library("voip_core") {
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sources = [
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"voip_core.cc",
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"voip_core.h",
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]
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deps = [
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":audio_channel",
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"..:audio",
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"../../api:scoped_refptr",
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"../../api/audio:audio_processing",
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"../../api/audio_codecs:audio_codecs_api",
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"../../api/task_queue",
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"../../api/voip:voip_api",
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"../../modules/audio_device:audio_device_api",
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"../../modules/audio_mixer:audio_mixer_impl",
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"../../rtc_base:criticalsection",
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"../../rtc_base:logging",
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"../../rtc_base/synchronization:mutex",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_library("audio_channel") {
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sources = [
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"audio_channel.cc",
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"audio_channel.h",
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]
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deps = [
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":audio_egress",
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":audio_ingress",
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"../../api:transport_api",
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"../../api/audio_codecs:audio_codecs_api",
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"../../api/task_queue",
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"../../api/voip:voip_api",
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"../../modules/audio_device:audio_device_api",
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"../../modules/rtp_rtcp",
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"../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../rtc_base:criticalsection",
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"../../rtc_base:logging",
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"../../rtc_base:refcount",
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]
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}
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rtc_library("audio_ingress") {
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sources = [
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"audio_ingress.cc",
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"audio_ingress.h",
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]
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deps = [
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"..:audio",
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"../../api:array_view",
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"../../api:rtp_headers",
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"../../api:scoped_refptr",
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"../../api:transport_api",
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"../../api/audio:audio_mixer_api",
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"../../api/audio_codecs:audio_codecs_api",
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"../../api/voip:voip_api",
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"../../modules/audio_coding",
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"../../modules/rtp_rtcp",
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"../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../rtc_base:criticalsection",
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"../../rtc_base:logging",
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"../../rtc_base:rtc_numerics",
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"../../rtc_base:safe_minmax",
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"../../rtc_base:timeutils",
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"../../rtc_base/synchronization:mutex",
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"../utility:audio_frame_operations",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_library("audio_egress") {
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sources = [
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"audio_egress.cc",
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"audio_egress.h",
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]
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deps = [
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"..:audio",
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"../../api:sequence_checker",
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"../../api/audio_codecs:audio_codecs_api",
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"../../api/task_queue",
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"../../call:audio_sender_interface",
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"../../modules/audio_coding",
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"../../modules/rtp_rtcp",
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"../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../rtc_base:logging",
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"../../rtc_base:timeutils",
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"../../rtc_base/synchronization:mutex",
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"../../rtc_base/system:no_unique_address",
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"../utility:audio_frame_operations",
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]
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}
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