mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 15:20:42 +01:00
a3728d310d
This is a reland of 1dddaa1a84
The regression that caused the original CL to be reverted was the fact that
invoking SetLocalDescription() inside of the CreateOffer() callback was no
longer executing synchronously and immediately.
In this CL, the original CL is patched so that the CreateOffer() operation
is marked as completed just before invoking the CreateOffer() callback
(versus doing it just afterwards). This ensures that the OperationsChain is
popped before the callback runs. The same applies for CreateAnswer().
See diff between Patch Set 1 (Original CL) and the latest Patch Set.
Original change's description:
> [PeerConnection] Use an OperationsChain in PeerConnection for async ops.
>
> For background, motivation, requirements and implementation notes, see
> https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing
>
> Using the OperationsChain will unblock future CLs from chaining multiple
> operations together such as implementing parameterless
> setLocalDescription().
>
> In this CL, the OperationsChain is used in existing signaling operations
> with little intended side-effects. An operation that is chained onto an
> empty OperationsChain will for instance execute immediately, and
> SetLocalDescription() and SetRemoteDescription() are implemented as
> "synchronous operations".
>
> The lifetime of the PeerConnection is not indended to change as a result
> of this CL: All chained operations use a WeakPtr to the PC to ensure
> use-after-free does not happen.
>
> There is one notable change though: CreateOffer() and CreateAnswer() will
> asynchronously delay other signaling methods from executing until they
> have completed.
>
> Drive-by fix: This CL also ensures that early failing
> CreateOffer/CreateAnswer operation's observers are invoked if the
> PeerConnection is destroyed while a PostCreateSessionDescriptionFailure
> is pending.
>
> Bug: webrtc:11019
> Change-Id: I521333e41d20d9bbfb1e721609f2c9db2a5f93a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157305
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29605}
TBR=steveanton@webrtc.org
Bug: webrtc:11019
Change-Id: I57b4496e63378c91c24679ee496e21f5cb6a0e59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158524
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29630}
665 lines
20 KiB
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665 lines
20 KiB
Text
# Copyright (c) 2016 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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group("pc") {
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deps = [
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":rtc_pc",
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]
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}
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config("rtc_pc_config") {
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defines = []
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if (rtc_enable_sctp) {
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defines += [ "HAVE_SCTP" ]
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}
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}
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rtc_library("rtc_pc_base") {
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visibility = [ "*" ]
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defines = []
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sources = [
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"channel.cc",
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"channel.h",
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"channel_interface.h",
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"channel_manager.cc",
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"channel_manager.h",
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"composite_data_channel_transport.cc",
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"composite_data_channel_transport.h",
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"composite_rtp_transport.cc",
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"composite_rtp_transport.h",
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"datagram_rtp_transport.cc",
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"datagram_rtp_transport.h",
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"dtls_srtp_transport.cc",
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"dtls_srtp_transport.h",
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"dtls_transport.cc",
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"dtls_transport.h",
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"external_hmac.cc",
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"external_hmac.h",
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"ice_transport.cc",
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"ice_transport.h",
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"jsep_transport.cc",
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"jsep_transport.h",
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"jsep_transport_controller.cc",
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"jsep_transport_controller.h",
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"media_session.cc",
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"media_session.h",
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"rtcp_mux_filter.cc",
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"rtcp_mux_filter.h",
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"rtp_media_utils.cc",
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"rtp_media_utils.h",
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"rtp_transport.cc",
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"rtp_transport.h",
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"rtp_transport_internal.h",
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"sctp_data_channel_transport.cc",
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"sctp_data_channel_transport.h",
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"sctp_transport.cc",
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"sctp_transport.h",
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"sctp_utils.cc",
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"sctp_utils.h",
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"session_description.cc",
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"session_description.h",
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"simulcast_description.cc",
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"simulcast_description.h",
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"srtp_filter.cc",
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"srtp_filter.h",
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"srtp_session.cc",
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"srtp_session.h",
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"srtp_transport.cc",
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"srtp_transport.h",
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"transport_stats.cc",
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"transport_stats.h",
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"used_ids.h",
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]
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deps = [
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":media_protocol_names",
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"../api:array_view",
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"../api:audio_options_api",
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"../api:call_api",
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"../api:libjingle_peerconnection_api",
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"../api:rtc_error",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api/crypto:options",
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"../api/rtc_event_log",
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"../api/transport:datagram_transport_interface",
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"../api/transport/media:media_transport_interface",
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"../api/video:builtin_video_bitrate_allocator_factory",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../call:call_interfaces",
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"../call:rtp_interfaces",
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"../call:rtp_receiver",
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"../common_video",
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"../common_video:common_video",
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"../logging:ice_log",
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"../media:rtc_data",
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"../media:rtc_h264_profile_id",
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"../media:rtc_media_base",
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"../media:rtc_media_config",
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"../modules/rtp_rtcp:rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../p2p:rtc_p2p",
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:deprecation",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:stringutils",
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"../rtc_base/system:file_wrapper",
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"../rtc_base/system:rtc_export",
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"../rtc_base/third_party/base64",
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"../rtc_base/third_party/sigslot",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/base:core_headers",
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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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if (rtc_build_libsrtp) {
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deps += [ "//third_party/libsrtp" ]
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}
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public_configs = [ ":rtc_pc_config" ]
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}
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rtc_source_set("rtc_pc") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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deps = [
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":rtc_pc_base",
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"../media:rtc_audio_video",
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]
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}
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rtc_library("media_protocol_names") {
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sources = [
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"media_protocol_names.cc",
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"media_protocol_names.h",
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]
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}
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rtc_library("peerconnection") {
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visibility = [ "*" ]
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cflags = []
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sources = [
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"audio_rtp_receiver.cc",
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"audio_rtp_receiver.h",
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"audio_track.cc",
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"audio_track.h",
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"data_channel.cc",
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"data_channel.h",
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"dtmf_sender.cc",
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"dtmf_sender.h",
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"ice_server_parsing.cc",
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"ice_server_parsing.h",
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"jitter_buffer_delay.cc",
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"jitter_buffer_delay.h",
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"jitter_buffer_delay_interface.h",
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"jitter_buffer_delay_proxy.h",
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"jsep_ice_candidate.cc",
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"jsep_session_description.cc",
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"local_audio_source.cc",
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"local_audio_source.h",
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"media_stream.cc",
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"media_stream.h",
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"media_stream_observer.cc",
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"media_stream_observer.h",
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"media_stream_track.h",
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"peer_connection.cc",
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"peer_connection.h",
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"peer_connection_factory.cc",
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"peer_connection_factory.h",
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"peer_connection_internal.h",
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"remote_audio_source.cc",
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"remote_audio_source.h",
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"rtc_stats_collector.cc",
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"rtc_stats_collector.h",
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"rtc_stats_traversal.cc",
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"rtc_stats_traversal.h",
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"rtp_parameters_conversion.cc",
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"rtp_parameters_conversion.h",
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"rtp_receiver.cc",
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"rtp_receiver.h",
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"rtp_sender.cc",
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"rtp_sender.h",
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"rtp_transceiver.cc",
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"rtp_transceiver.h",
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"sdp_serializer.cc",
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"sdp_serializer.h",
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"sdp_utils.cc",
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"sdp_utils.h",
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"stats_collector.cc",
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"stats_collector.h",
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"stream_collection.h",
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"track_media_info_map.cc",
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"track_media_info_map.h",
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"video_rtp_receiver.cc",
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"video_rtp_receiver.h",
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"video_track.cc",
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"video_track.h",
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"video_track_source.cc",
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"video_track_source.h",
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"webrtc_sdp.cc",
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"webrtc_sdp.h",
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"webrtc_session_description_factory.cc",
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"webrtc_session_description_factory.h",
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]
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deps = [
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":rtc_pc_base",
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"../api:array_view",
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"../api:audio_options_api",
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"../api:call_api",
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"../api:fec_controller_api",
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"../api:libjingle_peerconnection_api",
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"../api:media_stream_interface",
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"../api:network_state_predictor_api",
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"../api:rtc_error",
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"../api:rtc_event_log_output_file",
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"../api:rtc_stats_api",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api/crypto:frame_decryptor_interface",
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"../api/rtc_event_log",
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"../api/task_queue",
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"../api/transport:datagram_transport_interface",
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"../api/transport/media:media_transport_interface",
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"../api/units:data_rate",
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"../api/video:builtin_video_bitrate_allocator_factory",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../common_video",
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"../logging:ice_log",
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"../media:rtc_data",
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"../media:rtc_media_base",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../p2p:rtc_p2p",
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_operations_chain",
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"../rtc_base:safe_minmax",
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"../rtc_base:weak_ptr",
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"../rtc_base/experiments:field_trial_parser",
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"../rtc_base/system:fallthrough",
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"../rtc_base/system:file_wrapper",
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"../rtc_base/system:rtc_export",
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"../rtc_base/third_party/base64",
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"../rtc_base/third_party/sigslot",
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"../stats",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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"//third_party/abseil-cpp/absl/algorithm:container",
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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("libjingle_peerconnection") {
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visibility = [ "*" ]
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deps = [
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":peerconnection",
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"../api:libjingle_peerconnection_api",
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]
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}
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if (rtc_include_tests) {
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rtc_test("rtc_pc_unittests") {
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testonly = true
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sources = [
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"channel_manager_unittest.cc",
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"channel_unittest.cc",
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"composite_rtp_transport_test.cc",
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"dtls_srtp_transport_unittest.cc",
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"dtls_transport_unittest.cc",
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"ice_transport_unittest.cc",
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"jsep_transport_controller_unittest.cc",
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"jsep_transport_unittest.cc",
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"media_session_unittest.cc",
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"rtcp_mux_filter_unittest.cc",
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"rtp_transport_unittest.cc",
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"sctp_transport_unittest.cc",
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"session_description_unittest.cc",
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"srtp_filter_unittest.cc",
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"srtp_session_unittest.cc",
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"srtp_transport_unittest.cc",
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"test/rtp_transport_test_util.h",
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"test/srtp_test_util.h",
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"used_ids_unittest.cc",
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]
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include_dirs = [ "//third_party/libsrtp/srtp" ]
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if (is_win) {
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libs = [ "strmiids.lib" ]
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}
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deps = [
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":libjingle_peerconnection",
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":pc_test_utils",
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":rtc_pc",
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":rtc_pc_base",
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"../api:array_view",
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"../api:audio_options_api",
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"../api:fake_media_transport",
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"../api:ice_transport_factory",
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"../api:libjingle_peerconnection_api",
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"../api:loopback_media_transport",
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"../api:rtc_error",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api/transport/media:media_transport_interface",
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"../api/video:builtin_video_bitrate_allocator_factory",
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"../call:rtp_interfaces",
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"../call:rtp_receiver",
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"../media:rtc_data",
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"../media:rtc_media_base",
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"../media:rtc_media_tests_utils",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../p2p:fake_ice_transport",
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"../p2p:fake_port_allocator",
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"../p2p:p2p_test_utils",
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"../p2p:rtc_p2p",
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:gunit_helpers",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../rtc_base/third_party/sigslot",
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"../system_wrappers:metrics",
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"../test:test_main",
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"../test:test_support",
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/memory",
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]
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if (rtc_build_libsrtp) {
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deps += [ "//third_party/libsrtp" ]
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}
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if (is_android) {
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deps += [ "//testing/android/native_test:native_test_support" ]
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}
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}
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rtc_library("peerconnection_perf_tests") {
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testonly = true
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sources = [
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"peer_connection_rampup_tests.cc",
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]
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deps = [
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":pc_test_utils",
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":peerconnection_wrapper",
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"../api:audio_options_api",
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"../api:create_peerconnection_factory",
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"../api:libjingle_peerconnection_api",
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"../api:media_stream_interface",
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"../api:rtc_stats_api",
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"../api:scoped_refptr",
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"../api/audio:audio_mixer_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../api/video_codecs:builtin_video_decoder_factory",
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"../api/video_codecs:builtin_video_encoder_factory",
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"../api/video_codecs:video_codecs_api",
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"../media:rtc_media_tests_utils",
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"../modules/audio_device:audio_device_api",
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"../modules/audio_processing:api",
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"../p2p:p2p_test_utils",
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"../p2p:rtc_p2p",
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"../pc:peerconnection",
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:gunit_helpers",
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"../rtc_base:rtc_base_tests_utils",
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"../system_wrappers",
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"../test:perf_test",
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"../test:test_support",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("peerconnection_wrapper") {
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testonly = true
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sources = [
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"peer_connection_wrapper.cc",
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"peer_connection_wrapper.h",
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]
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deps = [
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":pc_test_utils",
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"../api:function_view",
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"../api:libjingle_peerconnection_api",
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"../api:media_stream_interface",
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"../api:rtc_error",
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"../api:rtc_stats_api",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../pc:peerconnection",
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"../rtc_base:checks",
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"../rtc_base:gunit_helpers",
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"../rtc_base:rtc_base_approved",
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"../test:test_support",
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]
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}
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rtc_library("pc_test_utils") {
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testonly = true
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sources = [
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"test/fake_audio_capture_module.cc",
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"test/fake_audio_capture_module.h",
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"test/fake_data_channel_provider.h",
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"test/fake_peer_connection_base.h",
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"test/fake_peer_connection_for_stats.h",
|
|
"test/fake_periodic_video_source.h",
|
|
"test/fake_periodic_video_track_source.h",
|
|
"test/fake_rtc_certificate_generator.h",
|
|
"test/fake_sctp_transport.h",
|
|
"test/fake_video_track_renderer.h",
|
|
"test/fake_video_track_source.h",
|
|
"test/frame_generator_capturer_video_track_source.h",
|
|
"test/mock_channel_interface.h",
|
|
"test/mock_data_channel.h",
|
|
"test/mock_delayable.h",
|
|
"test/mock_peer_connection_observers.h",
|
|
"test/mock_rtp_receiver_internal.h",
|
|
"test/mock_rtp_sender_internal.h",
|
|
"test/peer_connection_test_wrapper.cc",
|
|
"test/peer_connection_test_wrapper.h",
|
|
"test/rtc_stats_obtainer.h",
|
|
"test/test_sdp_strings.h",
|
|
]
|
|
|
|
deps = [
|
|
":libjingle_peerconnection",
|
|
":peerconnection",
|
|
":rtc_pc_base",
|
|
"../api:audio_options_api",
|
|
"../api:create_peerconnection_factory",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:media_stream_interface",
|
|
"../api:rtc_error",
|
|
"../api:rtc_stats_api",
|
|
"../api:scoped_refptr",
|
|
"../api/audio:audio_mixer_api",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../api/task_queue",
|
|
"../api/task_queue:default_task_queue_factory",
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
|
"../api/video:video_frame",
|
|
"../api/video:video_rtp_headers",
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../media:rtc_data",
|
|
"../media:rtc_media",
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../modules/audio_device",
|
|
"../modules/audio_processing",
|
|
"../modules/audio_processing:api",
|
|
"../p2p:fake_port_allocator",
|
|
"../p2p:p2p_test_utils",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:gunit_helpers",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:task_queue_for_test",
|
|
"../rtc_base/task_utils:repeating_task",
|
|
"../rtc_base/third_party/sigslot",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|
|
|
|
rtc_test("peerconnection_unittests") {
|
|
testonly = true
|
|
sources = [
|
|
"data_channel_unittest.cc",
|
|
"dtmf_sender_unittest.cc",
|
|
"ice_server_parsing_unittest.cc",
|
|
"jitter_buffer_delay_unittest.cc",
|
|
"jsep_session_description_unittest.cc",
|
|
"local_audio_source_unittest.cc",
|
|
"media_stream_unittest.cc",
|
|
"peer_connection_bundle_unittest.cc",
|
|
"peer_connection_crypto_unittest.cc",
|
|
"peer_connection_data_channel_unittest.cc",
|
|
"peer_connection_end_to_end_unittest.cc",
|
|
"peer_connection_factory_unittest.cc",
|
|
"peer_connection_histogram_unittest.cc",
|
|
"peer_connection_ice_unittest.cc",
|
|
"peer_connection_integrationtest.cc",
|
|
"peer_connection_interface_unittest.cc",
|
|
"peer_connection_jsep_unittest.cc",
|
|
"peer_connection_media_unittest.cc",
|
|
"peer_connection_rtp_unittest.cc",
|
|
"peer_connection_signaling_unittest.cc",
|
|
"peer_connection_simulcast_unittest.cc",
|
|
"peer_connection_wrapper.cc",
|
|
"peer_connection_wrapper.h",
|
|
"proxy_unittest.cc",
|
|
"rtc_stats_collector_unittest.cc",
|
|
"rtc_stats_integrationtest.cc",
|
|
"rtc_stats_traversal_unittest.cc",
|
|
"rtp_media_utils_unittest.cc",
|
|
"rtp_parameters_conversion_unittest.cc",
|
|
"rtp_sender_receiver_unittest.cc",
|
|
"rtp_transceiver_unittest.cc",
|
|
"sctp_utils_unittest.cc",
|
|
"sdp_serializer_unittest.cc",
|
|
"stats_collector_unittest.cc",
|
|
"test/fake_audio_capture_module_unittest.cc",
|
|
"test/test_sdp_strings.h",
|
|
"track_media_info_map_unittest.cc",
|
|
"video_track_unittest.cc",
|
|
"webrtc_sdp_unittest.cc",
|
|
]
|
|
|
|
if (rtc_enable_sctp) {
|
|
defines = [ "HAVE_SCTP" ]
|
|
}
|
|
|
|
deps = [
|
|
":peerconnection",
|
|
":rtc_pc_base",
|
|
"../api:array_view",
|
|
"../api:audio_options_api",
|
|
"../api:create_peerconnection_factory",
|
|
"../api:fake_frame_decryptor",
|
|
"../api:fake_frame_encryptor",
|
|
"../api:function_view",
|
|
"../api:libjingle_logging_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:loopback_media_transport",
|
|
"../api:media_stream_interface",
|
|
"../api:mock_rtp",
|
|
"../api:rtc_error",
|
|
"../api:scoped_refptr",
|
|
"../api/audio:audio_mixer_api",
|
|
"../api/crypto:frame_decryptor_interface",
|
|
"../api/crypto:frame_encryptor_interface",
|
|
"../api/crypto:options",
|
|
"../api/rtc_event_log",
|
|
"../api/rtc_event_log:rtc_event_log_factory",
|
|
"../api/task_queue:default_task_queue_factory",
|
|
"../api/transport/media:media_transport_interface",
|
|
"../api/transport/rtp:rtp_source",
|
|
"../api/units:time_delta",
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
|
"../logging:fake_rtc_event_log",
|
|
"../media:rtc_media_config",
|
|
"../media:rtc_media_engine_defaults",
|
|
"../modules/audio_device:audio_device_api",
|
|
"../modules/audio_processing:audio_processing_statistics",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../p2p:fake_port_allocator",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:gunit_helpers",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base/third_party/base64",
|
|
"../rtc_base/third_party/sigslot",
|
|
"../system_wrappers:metrics",
|
|
"../test:field_trial",
|
|
"../test:fileutils",
|
|
"../test:rtp_test_utils",
|
|
"//third_party/abseil-cpp/absl/algorithm:container",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
]
|
|
if (is_android) {
|
|
deps += [ ":android_black_magic" ]
|
|
}
|
|
|
|
deps += [
|
|
":libjingle_peerconnection",
|
|
":pc_test_utils",
|
|
"../api:callfactory_api",
|
|
"../api:fake_media_transport",
|
|
"../api:rtc_event_log_output_file",
|
|
"../api:rtc_stats_api",
|
|
"../api:rtp_parameters",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
"../api/audio_codecs/L16:audio_decoder_L16",
|
|
"../api/audio_codecs/L16:audio_encoder_L16",
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../media:rtc_audio_video",
|
|
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../modules/audio_processing",
|
|
"../modules/audio_processing:api",
|
|
"../modules/utility",
|
|
"../p2p:p2p_test_utils",
|
|
"../p2p:rtc_p2p",
|
|
"../pc:rtc_pc",
|
|
"../rtc_base",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:safe_conversions",
|
|
"../test:audio_codec_mocks",
|
|
"../test:test_main",
|
|
"../test:test_support",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
|
|
if (is_android) {
|
|
deps += [
|
|
"//testing/android/native_test:native_test_support",
|
|
|
|
# We need to depend on this one directly, or classloads will fail for
|
|
# the voice engine BuildInfo, for instance.
|
|
"../sdk/android:libjingle_peerconnection_java",
|
|
]
|
|
|
|
shard_timeout = 900
|
|
}
|
|
}
|
|
|
|
if (is_android) {
|
|
rtc_library("android_black_magic") {
|
|
# The android code uses hacky includes to chromium-base and the ssl code;
|
|
# having this in a separate target enables us to keep the peerconnection
|
|
# unit tests clean.
|
|
check_includes = false
|
|
testonly = true
|
|
sources = [
|
|
"test/android_test_initializer.cc",
|
|
"test/android_test_initializer.h",
|
|
]
|
|
deps = [
|
|
"../sdk/android:libjingle_peerconnection_jni",
|
|
"//testing/android/native_test:native_test_support",
|
|
]
|
|
}
|
|
}
|
|
}
|