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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn | 110 |
1 files changed, 110 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn new file mode 100644 index 0000000000..eb90a0b9ac --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn @@ -0,0 +1,110 @@ +# 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/system:rtc_export" ] + absl_deps = [ "//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" ] + deps = [ "..:audio_codecs_api" ] +} + +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", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_opus", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//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", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_opus", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//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", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_multiopus", + "../../../rtc_base/system:rtc_export", + "../opus:audio_encoder_opus_config", + ] + absl_deps = [ "//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", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_multiopus", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/memory", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} |