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-rw-r--r--third_party/libwebrtc/audio/voip/BUILD.gn103
1 files changed, 103 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/voip/BUILD.gn b/third_party/libwebrtc/audio/voip/BUILD.gn
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+++ b/third_party/libwebrtc/audio/voip/BUILD.gn
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+# Copyright(c) 2020 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")
+
+rtc_library("voip_core") {
+ sources = [
+ "voip_core.cc",
+ "voip_core.h",
+ ]
+ deps = [
+ ":audio_channel",
+ "..:audio",
+ "../../api:scoped_refptr",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../api/task_queue",
+ "../../api/voip:voip_api",
+ "../../modules/audio_device:audio_device_api",
+ "../../modules/audio_mixer:audio_mixer_impl",
+ "../../modules/audio_processing:api",
+ "../../rtc_base:criticalsection",
+ "../../rtc_base:logging",
+ "../../rtc_base/synchronization:mutex",
+ ]
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_channel") {
+ sources = [
+ "audio_channel.cc",
+ "audio_channel.h",
+ ]
+ deps = [
+ ":audio_egress",
+ ":audio_ingress",
+ "../../api:transport_api",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../api/task_queue",
+ "../../api/voip:voip_api",
+ "../../modules/audio_device:audio_device_api",
+ "../../modules/rtp_rtcp",
+ "../../modules/rtp_rtcp:rtp_rtcp_format",
+ "../../rtc_base:criticalsection",
+ "../../rtc_base:logging",
+ "../../rtc_base:refcount",
+ ]
+}
+
+rtc_library("audio_ingress") {
+ sources = [
+ "audio_ingress.cc",
+ "audio_ingress.h",
+ ]
+ deps = [
+ "..:audio",
+ "../../api:array_view",
+ "../../api:rtp_headers",
+ "../../api:scoped_refptr",
+ "../../api:transport_api",
+ "../../api/audio:audio_mixer_api",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../api/voip:voip_api",
+ "../../modules/audio_coding",
+ "../../modules/rtp_rtcp",
+ "../../modules/rtp_rtcp:rtp_rtcp_format",
+ "../../rtc_base:criticalsection",
+ "../../rtc_base:logging",
+ "../../rtc_base:rtc_numerics",
+ "../../rtc_base:safe_minmax",
+ "../../rtc_base:timeutils",
+ "../../rtc_base/synchronization:mutex",
+ "../utility:audio_frame_operations",
+ ]
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_egress") {
+ sources = [
+ "audio_egress.cc",
+ "audio_egress.h",
+ ]
+ deps = [
+ "..:audio",
+ "../../api:sequence_checker",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../api/task_queue",
+ "../../call:audio_sender_interface",
+ "../../modules/audio_coding",
+ "../../modules/rtp_rtcp",
+ "../../modules/rtp_rtcp:rtp_rtcp_format",
+ "../../rtc_base:logging",
+ "../../rtc_base:rtc_task_queue",
+ "../../rtc_base:timeutils",
+ "../../rtc_base/synchronization:mutex",
+ "../../rtc_base/system:no_unique_address",
+ "../utility:audio_frame_operations",
+ ]
+}