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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/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/BUILD.gn')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/BUILD.gn | 144 |
1 files changed, 144 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/BUILD.gn new file mode 100644 index 0000000000..82ed31a5da --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/BUILD.gn @@ -0,0 +1,144 @@ +# 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_codecs_api") { + visibility = [ "*" ] + sources = [ + "audio_codec_pair_id.cc", + "audio_codec_pair_id.h", + "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", + "..:bitrate_allocation", + "..:make_ref_counted", + "..:scoped_refptr", + "../../api:field_trials_view", + "../../rtc_base:buffer", + "../../rtc_base:checks", + "../../rtc_base:event_tracer", + "../../rtc_base:refcount", + "../../rtc_base:sanitizer", + "../../rtc_base/system:rtc_export", + "../units:time_delta", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/base:core_headers", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("builtin_audio_decoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "builtin_audio_decoder_factory.cc", + "builtin_audio_decoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "L16:audio_decoder_L16", + "g711:audio_decoder_g711", + "g722:audio_decoder_g722", + ] + 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_multiopus", + "opus:audio_decoder_opus", + ] + defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ] + } else { + defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] + } +} + +rtc_library("builtin_audio_encoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "builtin_audio_encoder_factory.cc", + "builtin_audio_encoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "L16:audio_encoder_L16", + "g711:audio_encoder_g711", + "g722:audio_encoder_g722", + ] + 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_multiopus", + "opus:audio_encoder_opus", + ] + defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ] + } else { + defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] + } +} + +rtc_library("opus_audio_decoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "opus_audio_decoder_factory.cc", + "opus_audio_decoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "opus:audio_decoder_multiopus", + "opus:audio_decoder_opus", + ] +} + +rtc_library("opus_audio_encoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "opus_audio_encoder_factory.cc", + "opus_audio_encoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "opus:audio_encoder_multiopus", + "opus:audio_encoder_opus", + ] +} |