mirror of
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}
111 lines
3 KiB
Text
111 lines
3 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_source_set("audio_codecs_api") {
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visibility = [ "*" ]
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sources = [
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"audio_codec_pair_id.cc",
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"audio_codec_pair_id.h",
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"audio_decoder.cc",
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"audio_decoder.h",
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"audio_decoder_factory.h",
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"audio_decoder_factory_template.h",
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"audio_encoder.cc",
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"audio_encoder.h",
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"audio_encoder_factory.h",
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"audio_encoder_factory_template.h",
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"audio_format.cc",
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"audio_format.h",
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]
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deps = [
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"..:array_view",
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"..:bitrate_allocation",
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"..:scoped_refptr",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:sanitizer",
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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("builtin_audio_decoder_factory") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ]
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sources = [
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"builtin_audio_decoder_factory.cc",
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"builtin_audio_decoder_factory.h",
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]
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deps = [
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":audio_codecs_api",
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"..:scoped_refptr",
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"../../rtc_base:rtc_base_approved",
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"L16:audio_decoder_L16",
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"g711:audio_decoder_g711",
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"g722:audio_decoder_g722",
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"isac:audio_decoder_isac",
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]
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defines = []
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if (rtc_include_ilbc) {
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deps += [ "ilbc:audio_decoder_ilbc" ]
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
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}
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if (rtc_include_opus) {
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deps += [
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"opus:audio_decoder_multiopus",
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"opus:audio_decoder_opus",
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]
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
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}
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}
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rtc_static_library("builtin_audio_encoder_factory") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ]
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sources = [
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"builtin_audio_encoder_factory.cc",
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"builtin_audio_encoder_factory.h",
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]
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deps = [
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":audio_codecs_api",
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"..:scoped_refptr",
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"../../rtc_base:rtc_base_approved",
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"L16:audio_encoder_L16",
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"g711:audio_encoder_g711",
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"g722:audio_encoder_g722",
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"isac:audio_encoder_isac",
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]
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defines = []
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if (rtc_include_ilbc) {
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deps += [ "ilbc:audio_encoder_ilbc" ]
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
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}
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if (rtc_include_opus) {
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deps += [
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"opus:audio_encoder_multiopus",
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"opus:audio_encoder_opus",
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]
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
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} else {
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defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
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}
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}
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