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https://github.com/mollyim/webrtc.git
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44c21f48ee
Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely. Adds an ENCODER and sets it up to parse SDP config for multistream opus. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27775}
117 lines
3.2 KiB
Text
117 lines
3.2 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_static_library("audio_encoder_opus_config") {
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visibility = [ "*" ]
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sources = [
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"audio_encoder_multi_channel_opus_config.cc",
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"audio_encoder_multi_channel_opus_config.h",
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"audio_encoder_opus_config.cc",
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"audio_encoder_opus_config.h",
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]
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deps = [
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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defines = []
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if (rtc_opus_variable_complexity) {
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defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
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} else {
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defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
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}
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}
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rtc_source_set("audio_decoder_opus_config") {
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visibility = [ "*" ]
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sources = [
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"audio_decoder_multi_channel_opus_config.h",
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]
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}
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rtc_source_set("audio_encoder_opus") {
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visibility = [ "*" ]
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poisonous = [ "audio_codecs" ]
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public = [
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"audio_encoder_opus.h",
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]
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sources = [
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"audio_encoder_opus.cc",
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]
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deps = [
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":audio_encoder_opus_config",
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"..:audio_codecs_api",
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"../../../modules/audio_coding:webrtc_opus",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_static_library("audio_decoder_opus") {
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visibility = [ "*" ]
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poisonous = [ "audio_codecs" ]
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sources = [
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"audio_decoder_opus.cc",
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"audio_decoder_opus.h",
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]
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deps = [
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"..:audio_codecs_api",
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"../../../modules/audio_coding:webrtc_opus",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_source_set("audio_encoder_multiopus") {
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visibility = [ "*" ]
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poisonous = [ "audio_codecs" ]
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public = [
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"audio_encoder_multi_channel_opus.h",
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]
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sources = [
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"audio_encoder_multi_channel_opus.cc",
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]
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deps = [
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"..:audio_codecs_api",
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"../../../modules/audio_coding:webrtc_multiopus",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"../opus:audio_encoder_opus_config",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_static_library("audio_decoder_multiopus") {
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visibility = [ "*" ]
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poisonous = [ "audio_codecs" ]
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sources = [
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"audio_decoder_multi_channel_opus.cc",
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"audio_decoder_multi_channel_opus.h",
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]
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deps = [
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":audio_decoder_opus_config",
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"..:audio_codecs_api",
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"../../../modules/audio_coding:webrtc_multiopus",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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