2015-09-25 13:58:30 +02:00
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# Copyright (c) 2015 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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2017-01-24 06:58:22 -08:00
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import("../webrtc.gni")
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2015-09-25 13:58:30 +02:00
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2016-12-07 04:52:58 -08:00
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rtc_source_set("call_interfaces") {
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sources = [
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2018-04-09 14:24:52 +02:00
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"audio_receive_stream.cc",
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2016-12-07 04:52:58 -08:00
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"audio_receive_stream.h",
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"audio_send_stream.h",
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2018-04-09 14:24:52 +02:00
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"audio_state.cc",
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2016-12-07 04:52:58 -08:00
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"audio_state.h",
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"call.h",
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2018-04-09 14:24:52 +02:00
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"call_config.cc",
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"call_config.h",
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"flexfec_receive_stream.cc",
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2016-12-19 01:13:46 -08:00
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"flexfec_receive_stream.h",
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2018-08-07 11:03:12 +02:00
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"packet_receiver.h",
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2017-01-31 03:58:40 -08:00
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"syncable.cc",
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"syncable.h",
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2016-12-07 04:52:58 -08:00
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]
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2018-01-15 10:20:00 -05:00
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if (!build_with_mozilla) {
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sources += [ "audio_send_stream.cc" ]
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}
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2017-02-27 15:57:45 -08:00
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deps = [
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2017-06-01 04:02:35 -07:00
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":rtp_interfaces",
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2017-08-22 05:43:23 -07:00
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":video_stream_api",
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2017-02-27 15:57:45 -08:00
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"..:webrtc_common",
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2018-01-19 17:58:57 +01:00
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"../api:fec_controller_api",
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Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675
TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 07:39:05 -07:00
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"../api:libjingle_peerconnection_api",
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2017-02-27 15:57:45 -08:00
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"../api:transport_api",
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2018-02-16 13:43:49 +01:00
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"../api/audio:audio_mixer_api",
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2017-02-27 15:57:45 -08:00
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"../api/audio_codecs:audio_codecs_api",
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2018-05-18 18:05:10 +02:00
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"../api/transport:network_control",
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2018-04-09 14:24:52 +02:00
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"../modules/audio_device:audio_device",
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2018-11-07 14:29:54 +00:00
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"../modules/audio_processing:api",
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2018-04-09 14:24:52 +02:00
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"../modules/audio_processing:audio_processing",
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2017-11-24 17:29:59 +01:00
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"../modules/audio_processing:audio_processing_statistics",
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2018-04-03 13:40:05 +02:00
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"../rtc_base:audio_format_to_string",
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2018-10-23 12:03:01 +02:00
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"../rtc_base:checks",
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2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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2018-10-15 21:06:26 +02:00
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"../rtc_base/network:sent_packet",
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2018-06-15 12:28:07 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
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2017-02-27 15:57:45 -08:00
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]
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2016-12-07 04:52:58 -08:00
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}
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2017-06-01 04:02:35 -07:00
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# TODO(nisse): These RTP targets should be moved elsewhere
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2017-07-28 10:05:45 -07:00
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# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
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2017-06-01 04:02:35 -07:00
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rtc_source_set("rtp_interfaces") {
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2018-10-04 13:41:03 +02:00
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# Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
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# because there exists client code that uses it.
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# TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
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# client code gets updated.
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visibility = [ "*" ]
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2017-06-01 04:02:35 -07:00
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sources = [
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2017-06-26 11:23:54 -07:00
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"rtcp_packet_sink_interface.h",
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2017-09-01 15:29:28 +02:00
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"rtp_config.cc",
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"rtp_config.h",
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2017-06-01 04:02:35 -07:00
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"rtp_packet_sink_interface.h",
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2017-06-21 01:05:22 -07:00
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"rtp_stream_receiver_controller_interface.h",
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2017-06-01 04:02:35 -07:00
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"rtp_transport_controller_send_interface.h",
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]
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2017-06-26 11:23:54 -07:00
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deps = [
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2017-09-04 05:43:17 -07:00
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"../api:array_view",
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2018-10-04 15:21:55 +02:00
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"../api:fec_controller_api",
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2018-06-21 16:58:01 +02:00
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"../api:libjingle_peerconnection_api",
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2018-05-07 14:01:37 +02:00
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"../api/transport:bitrate_settings",
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2018-07-17 16:03:46 +02:00
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"../logging:rtc_event_log_api",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_base_approved",
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2018-06-15 12:28:07 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
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2017-06-26 11:23:54 -07:00
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]
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2017-06-01 04:02:35 -07:00
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}
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rtc_source_set("rtp_receiver") {
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2018-01-10 15:54:53 +00:00
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visibility = [ "*" ]
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2017-06-01 04:02:35 -07:00
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sources = [
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2017-06-26 11:23:54 -07:00
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"rtcp_demuxer.cc",
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"rtcp_demuxer.h",
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2017-06-01 04:02:35 -07:00
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"rtp_demuxer.cc",
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"rtp_demuxer.h",
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2017-06-26 11:23:54 -07:00
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"rtp_rtcp_demuxer_helper.cc",
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"rtp_rtcp_demuxer_helper.h",
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2017-06-21 01:05:22 -07:00
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"rtp_stream_receiver_controller.cc",
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"rtp_stream_receiver_controller.h",
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2017-06-01 04:02:35 -07:00
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"rtx_receive_stream.cc",
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"rtx_receive_stream.h",
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2017-08-17 15:23:51 -07:00
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"ssrc_binding_observer.h",
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2017-06-01 04:02:35 -07:00
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]
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deps = [
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":rtp_interfaces",
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2017-06-26 11:23:54 -07:00
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"..:webrtc_common",
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2017-09-04 05:43:17 -07:00
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"../api:array_view",
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2017-12-15 14:40:10 +01:00
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"../api:libjingle_peerconnection_api",
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2017-06-01 04:02:35 -07:00
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"../modules/rtp_rtcp",
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2017-12-06 07:51:33 +01:00
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"../modules/rtp_rtcp:rtp_rtcp_format",
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2017-12-13 16:05:42 +01:00
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"../rtc_base:checks",
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2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_base_approved",
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Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
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"//third_party/abseil-cpp/absl/memory",
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2018-06-15 12:28:07 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
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2017-06-01 04:02:35 -07:00
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]
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}
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rtc_source_set("rtp_sender") {
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sources = [
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2018-07-17 10:16:41 +02:00
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"rtp_payload_params.cc",
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"rtp_payload_params.h",
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2017-06-01 04:02:35 -07:00
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"rtp_transport_controller_send.cc",
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"rtp_transport_controller_send.h",
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2018-07-19 10:34:38 +02:00
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"rtp_video_sender.cc",
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"rtp_video_sender.h",
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"rtp_video_sender_interface.h",
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2017-06-01 04:02:35 -07:00
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]
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deps = [
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2018-02-21 13:01:55 +01:00
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":bitrate_configurator",
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2017-06-01 04:02:35 -07:00
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":rtp_interfaces",
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2017-08-09 06:42:32 -07:00
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"..:webrtc_common",
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2018-10-04 15:21:55 +02:00
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"../api:fec_controller_api",
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2018-07-17 16:03:46 +02:00
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"../api:transport_api",
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2018-05-09 10:33:39 +02:00
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"../api/transport:network_control",
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2018-07-11 17:11:31 +02:00
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"../api/video_codecs:video_codecs_api",
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2018-07-17 16:03:46 +02:00
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"../logging:rtc_event_log_api",
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2017-06-01 04:02:35 -07:00
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"../modules/congestion_controller",
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2018-03-13 19:07:46 +01:00
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"../modules/congestion_controller/rtp:congestion_controller",
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2017-10-31 10:19:10 +01:00
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"../modules/pacing",
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2018-07-11 17:11:31 +02:00
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"../modules/rtp_rtcp:rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/rtp_rtcp:rtp_video_header",
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2018-02-22 11:10:18 +01:00
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"../modules/utility",
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2018-07-11 17:11:31 +02:00
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"../modules/video_coding:video_codec_interface",
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"../rtc_base:checks",
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2018-07-17 16:03:46 +02:00
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"../rtc_base:rate_limiter",
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2018-02-21 13:02:51 +01:00
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"../rtc_base:rtc_base",
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2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_base_approved",
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2018-05-04 14:08:15 +02:00
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"../rtc_base:rtc_task_queue",
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2018-09-28 08:51:10 +02:00
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"../system_wrappers:field_trial",
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Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
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"//third_party/abseil-cpp/absl/memory",
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2017-06-01 04:02:35 -07:00
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]
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}
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2018-02-20 19:38:37 +01:00
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rtc_source_set("bitrate_configurator") {
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sources = [
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"rtp_bitrate_configurator.cc",
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"rtp_bitrate_configurator.h",
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]
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deps = [
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":rtp_interfaces",
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2018-06-21 16:58:01 +02:00
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"../api:libjingle_peerconnection_api",
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2018-05-07 14:01:37 +02:00
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"../api/transport:bitrate_settings",
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2018-02-20 19:38:37 +01:00
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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]
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}
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2017-10-10 14:38:11 +02:00
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rtc_source_set("bitrate_allocator") {
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2015-09-25 13:58:30 +02:00
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sources = [
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2015-11-12 21:02:42 -08:00
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"bitrate_allocator.cc",
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2017-10-10 14:38:11 +02:00
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"bitrate_allocator.h",
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]
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deps = [
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2018-11-21 19:13:07 +01:00
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"../api/units:data_rate",
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"../api/units:time_delta",
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2017-10-10 14:38:11 +02:00
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"../modules/bitrate_controller",
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2017-12-13 16:05:42 +01:00
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"../rtc_base:checks",
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2017-10-10 14:38:11 +02:00
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"../rtc_base:rtc_base_approved",
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"../rtc_base:sequenced_task_checker",
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"../system_wrappers",
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2018-09-28 08:51:10 +02:00
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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2017-10-10 14:38:11 +02:00
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]
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}
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rtc_static_library("call") {
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sources = [
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2015-09-25 13:58:30 +02:00
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"call.cc",
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2017-06-15 12:52:32 -07:00
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"callfactory.cc",
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"callfactory.h",
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2018-03-14 15:16:50 +01:00
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"degraded_call.cc",
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"degraded_call.h",
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2016-12-19 01:13:46 -08:00
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"flexfec_receive_stream_impl.cc",
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"flexfec_receive_stream_impl.h",
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2018-03-21 14:38:32 +01:00
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"receive_time_calculator.cc",
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"receive_time_calculator.h",
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2015-09-25 13:58:30 +02:00
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]
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|
|
|
|
|
|
deps = [
|
2017-10-10 14:38:11 +02:00
|
|
|
":bitrate_allocator",
|
2016-12-07 04:52:58 -08:00
|
|
|
":call_interfaces",
|
2018-03-14 15:16:50 +01:00
|
|
|
":fake_network",
|
2017-06-01 04:02:35 -07:00
|
|
|
":rtp_interfaces",
|
|
|
|
|
":rtp_receiver",
|
|
|
|
|
":rtp_sender",
|
2018-08-17 13:00:54 +02:00
|
|
|
":simulated_network",
|
2017-09-01 15:29:28 +02:00
|
|
|
":video_stream_api",
|
2018-02-14 12:20:13 +01:00
|
|
|
"../api:callfactory_api",
|
2018-08-17 13:00:54 +02:00
|
|
|
"../api:simulated_network_api",
|
2016-11-28 07:02:13 -08:00
|
|
|
"../api:transport_api",
|
2018-05-09 10:33:39 +02:00
|
|
|
"../api/transport:network_control",
|
2016-06-03 13:14:28 -07:00
|
|
|
"../audio",
|
2018-02-01 11:04:46 -08:00
|
|
|
"../logging:rtc_event_audio",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../logging:rtc_event_log_api",
|
2018-02-01 11:04:46 -08:00
|
|
|
"../logging:rtc_event_rtp_rtcp",
|
|
|
|
|
"../logging:rtc_event_video",
|
|
|
|
|
"../logging:rtc_stream_config",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/bitrate_controller",
|
2016-02-23 13:30:42 +01:00
|
|
|
"../modules/congestion_controller",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/pacing",
|
2015-09-25 13:58:30 +02:00
|
|
|
"../modules/rtp_rtcp",
|
2017-12-06 07:51:33 +01:00
|
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/utility",
|
2018-01-19 17:58:57 +01:00
|
|
|
"../modules/video_coding:video_coding",
|
2017-12-13 16:05:42 +01:00
|
|
|
"../rtc_base:checks",
|
2018-03-01 15:56:57 +01:00
|
|
|
"../rtc_base:rate_limiter",
|
2017-08-27 23:40:10 -07:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_task_queue",
|
2018-03-12 15:59:12 +01:00
|
|
|
"../rtc_base:safe_minmax",
|
2017-08-18 02:47:08 -07:00
|
|
|
"../rtc_base:sequenced_task_checker",
|
2018-10-12 11:15:09 +02:00
|
|
|
"../rtc_base/experiments:field_trial_parser",
|
2018-03-23 14:53:54 +01:00
|
|
|
"../rtc_base/synchronization:rw_lock_wrapper",
|
2015-09-25 13:58:30 +02:00
|
|
|
"../system_wrappers",
|
2018-09-28 08:51:10 +02:00
|
|
|
"../system_wrappers:field_trial",
|
|
|
|
|
"../system_wrappers:metrics",
|
2016-06-03 13:14:28 -07:00
|
|
|
"../video",
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
2018-06-15 12:28:07 +02:00
|
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
2015-09-25 13:58:30 +02:00
|
|
|
]
|
|
|
|
|
}
|
2016-06-14 12:52:54 +02:00
|
|
|
|
2017-08-22 05:43:23 -07:00
|
|
|
rtc_source_set("video_stream_api") {
|
|
|
|
|
sources = [
|
|
|
|
|
"video_receive_stream.cc",
|
|
|
|
|
"video_receive_stream.h",
|
|
|
|
|
"video_send_stream.cc",
|
|
|
|
|
"video_send_stream.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
2017-09-01 15:29:28 +02:00
|
|
|
":rtp_interfaces",
|
|
|
|
|
"../api:libjingle_peerconnection_api",
|
2017-08-22 05:43:23 -07:00
|
|
|
"../api:transport_api",
|
2018-05-11 11:15:30 +02:00
|
|
|
"../api/video:video_frame",
|
2018-07-24 09:29:58 +02:00
|
|
|
"../api/video:video_stream_encoder",
|
2018-03-27 16:39:01 +02:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
2017-08-22 05:43:23 -07:00
|
|
|
"../common_video:common_video",
|
2017-12-15 14:40:10 +01:00
|
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
2018-01-04 15:58:20 +01:00
|
|
|
"../rtc_base:checks",
|
2017-08-22 05:43:23 -07:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2018-06-15 12:28:07 +02:00
|
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
2017-08-22 05:43:23 -07:00
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2018-08-07 18:58:05 +02:00
|
|
|
rtc_source_set("simulated_network") {
|
|
|
|
|
sources = [
|
|
|
|
|
"simulated_network.cc",
|
|
|
|
|
"simulated_network.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
"../api:simulated_network_api",
|
2018-11-06 19:18:28 +01:00
|
|
|
"../api/units:data_rate",
|
|
|
|
|
"../api/units:data_size",
|
|
|
|
|
"../api/units:time_delta",
|
2018-08-07 18:58:05 +02:00
|
|
|
"../rtc_base:rtc_base_approved",
|
|
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
|
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2018-08-17 14:26:54 +02:00
|
|
|
rtc_source_set("simulated_packet_receiver") {
|
|
|
|
|
sources = [
|
|
|
|
|
"simulated_packet_receiver.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
":call_interfaces",
|
|
|
|
|
"../api:simulated_network_api",
|
|
|
|
|
"../modules:module_api",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2018-03-14 15:16:50 +01:00
|
|
|
rtc_source_set("fake_network") {
|
|
|
|
|
sources = [
|
|
|
|
|
"fake_network_pipe.cc",
|
|
|
|
|
"fake_network_pipe.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
":call_interfaces",
|
2018-08-07 18:58:05 +02:00
|
|
|
":simulated_network",
|
2018-08-17 14:26:54 +02:00
|
|
|
":simulated_packet_receiver",
|
2018-06-21 16:58:01 +02:00
|
|
|
"../api:simulated_network_api",
|
2018-03-14 15:16:50 +01:00
|
|
|
"../api:transport_api",
|
|
|
|
|
"../modules:module_api",
|
2018-10-30 08:23:27 +01:00
|
|
|
"../modules/utility",
|
2018-03-14 15:16:50 +01:00
|
|
|
"../rtc_base:rtc_base_approved",
|
|
|
|
|
"../rtc_base:sequenced_task_checker",
|
|
|
|
|
"../system_wrappers",
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
2018-03-14 15:16:50 +01:00
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2016-06-14 12:52:54 +02:00
|
|
|
if (rtc_include_tests) {
|
2016-09-02 04:10:34 -07:00
|
|
|
rtc_source_set("call_tests") {
|
2016-06-14 12:52:54 +02:00
|
|
|
testonly = true
|
2017-04-25 04:04:50 -07:00
|
|
|
|
2016-06-14 12:52:54 +02:00
|
|
|
sources = [
|
|
|
|
|
"bitrate_allocator_unittest.cc",
|
|
|
|
|
"bitrate_estimator_tests.cc",
|
|
|
|
|
"call_unittest.cc",
|
2016-10-20 04:54:48 -07:00
|
|
|
"flexfec_receive_stream_unittest.cc",
|
2018-03-21 14:38:32 +01:00
|
|
|
"receive_time_calculator_unittest.cc",
|
2017-06-26 11:23:54 -07:00
|
|
|
"rtcp_demuxer_unittest.cc",
|
2018-02-20 19:38:37 +01:00
|
|
|
"rtp_bitrate_configurator_unittest.cc",
|
2017-05-31 09:12:25 -07:00
|
|
|
"rtp_demuxer_unittest.cc",
|
2018-07-17 10:16:41 +02:00
|
|
|
"rtp_payload_params_unittest.cc",
|
2017-06-26 11:23:54 -07:00
|
|
|
"rtp_rtcp_demuxer_helper_unittest.cc",
|
2018-07-19 10:34:38 +02:00
|
|
|
"rtp_video_sender_unittest.cc",
|
2017-05-19 06:15:19 -07:00
|
|
|
"rtx_receive_stream_unittest.cc",
|
2016-06-14 12:52:54 +02:00
|
|
|
]
|
|
|
|
|
deps = [
|
2017-10-10 14:38:11 +02:00
|
|
|
":bitrate_allocator",
|
2018-02-20 19:38:37 +01:00
|
|
|
":bitrate_configurator",
|
2016-06-14 12:52:54 +02:00
|
|
|
":call",
|
2017-12-04 10:50:51 +01:00
|
|
|
":call_interfaces",
|
2017-07-28 10:05:45 -07:00
|
|
|
":mock_rtp_interfaces",
|
2017-06-01 04:02:35 -07:00
|
|
|
":rtp_interfaces",
|
|
|
|
|
":rtp_receiver",
|
|
|
|
|
":rtp_sender",
|
2018-08-20 13:27:45 +02:00
|
|
|
":simulated_network",
|
2017-09-04 05:43:17 -07:00
|
|
|
"../api:array_view",
|
2018-11-15 08:26:19 -08:00
|
|
|
"../api:fake_media_transport",
|
2018-11-16 09:57:58 -08:00
|
|
|
"../api:fake_media_transport",
|
2017-12-15 14:40:10 +01:00
|
|
|
"../api:libjingle_peerconnection_api",
|
2017-05-23 06:07:11 -07:00
|
|
|
"../api:mock_audio_mixer",
|
2017-11-02 13:04:41 +01:00
|
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
2018-01-11 13:52:30 +01:00
|
|
|
"../audio:audio",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../logging:rtc_event_log_api",
|
2018-02-01 11:04:46 -08:00
|
|
|
"../logging:rtc_event_log_impl_base",
|
2016-11-17 06:28:59 -08:00
|
|
|
"../modules/audio_device:mock_audio_device",
|
2016-11-17 06:48:48 -08:00
|
|
|
"../modules/audio_mixer",
|
2017-12-05 15:33:16 +01:00
|
|
|
"../modules/audio_mixer:audio_mixer_impl",
|
2017-12-19 16:44:45 +01:00
|
|
|
"../modules/audio_processing:mocks",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/bitrate_controller",
|
2017-10-10 14:38:11 +02:00
|
|
|
"../modules/congestion_controller",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/pacing",
|
2017-09-04 04:57:11 -07:00
|
|
|
"../modules/pacing:mock_paced_sender",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/rtp_rtcp",
|
2017-05-10 08:41:13 -07:00
|
|
|
"../modules/rtp_rtcp:mock_rtp_rtcp",
|
2017-12-06 07:51:33 +01:00
|
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
2017-08-03 05:57:11 -07:00
|
|
|
"../modules/utility:mock_process_thread",
|
2018-07-11 17:11:31 +02:00
|
|
|
"../modules/video_coding:video_codec_interface",
|
2018-10-04 15:21:55 +02:00
|
|
|
"../modules/video_coding:video_coding",
|
2017-12-13 16:05:42 +01:00
|
|
|
"../rtc_base:checks",
|
2018-07-17 16:03:46 +02:00
|
|
|
"../rtc_base:rate_limiter",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../system_wrappers",
|
2017-04-10 05:15:48 -07:00
|
|
|
"../test:audio_codec_mocks",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../test:direct_transport",
|
2018-08-27 14:12:27 +02:00
|
|
|
"../test:fake_video_codecs",
|
2018-07-11 17:11:31 +02:00
|
|
|
"../test:field_trial",
|
2016-11-17 06:48:48 -08:00
|
|
|
"../test:test_common",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../test:test_support",
|
|
|
|
|
"../test:video_test_common",
|
2018-07-17 16:03:46 +02:00
|
|
|
"../video:video",
|
2018-08-27 14:12:27 +02:00
|
|
|
"//testing/gmock",
|
2016-06-14 12:52:54 +02:00
|
|
|
"//testing/gtest",
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
2018-10-30 22:23:02 +01:00
|
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
2016-06-14 12:52:54 +02:00
|
|
|
]
|
2016-10-16 23:56:12 -07:00
|
|
|
if (!build_with_chromium && is_clang) {
|
|
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
2016-09-02 04:10:34 -07:00
|
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
2016-06-14 12:52:54 +02:00
|
|
|
}
|
|
|
|
|
}
|
2017-01-05 07:09:50 -08:00
|
|
|
|
|
|
|
|
rtc_source_set("call_perf_tests") {
|
|
|
|
|
testonly = true
|
2017-04-25 04:04:50 -07:00
|
|
|
|
2017-01-05 07:09:50 -08:00
|
|
|
sources = [
|
|
|
|
|
"call_perf_tests.cc",
|
|
|
|
|
"rampup_tests.cc",
|
|
|
|
|
"rampup_tests.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
2017-02-27 15:57:45 -08:00
|
|
|
":call_interfaces",
|
2018-08-20 13:30:39 +02:00
|
|
|
":simulated_network",
|
2017-09-01 15:29:28 +02:00
|
|
|
":video_stream_api",
|
2018-08-17 14:26:54 +02:00
|
|
|
"../api:simulated_network_api",
|
2017-05-02 06:46:30 -07:00
|
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
2018-11-08 10:02:56 -08:00
|
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
2018-05-15 15:18:36 +02:00
|
|
|
"../api/video:video_bitrate_allocation",
|
2018-05-18 11:37:23 +02:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../logging:rtc_event_log_api",
|
2018-10-15 13:24:31 +02:00
|
|
|
"../logging:rtc_event_log_impl_output",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/audio_coding",
|
2018-03-07 14:44:00 +01:00
|
|
|
"../modules/audio_device",
|
2018-03-15 12:44:12 +01:00
|
|
|
"../modules/audio_device:audio_device_impl",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../modules/audio_mixer:audio_mixer_impl",
|
|
|
|
|
"../modules/rtp_rtcp",
|
2017-12-13 16:05:42 +01:00
|
|
|
"../rtc_base:checks",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../system_wrappers",
|
2018-09-28 08:51:10 +02:00
|
|
|
"../system_wrappers:metrics",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../test:direct_transport",
|
2018-08-27 14:12:27 +02:00
|
|
|
"../test:fake_video_codecs",
|
2017-07-11 06:20:45 -07:00
|
|
|
"../test:field_trial",
|
2018-03-15 15:05:39 +01:00
|
|
|
"../test:fileutils",
|
2018-01-05 15:34:09 +01:00
|
|
|
"../test:perf_test",
|
2017-07-11 06:20:45 -07:00
|
|
|
"../test:test_common",
|
2017-02-27 15:57:45 -08:00
|
|
|
"../test:test_support",
|
|
|
|
|
"../test:video_test_common",
|
|
|
|
|
"../video",
|
2017-01-05 07:09:50 -08:00
|
|
|
"//testing/gtest",
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
2017-01-05 07:09:50 -08:00
|
|
|
]
|
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
|
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
2017-07-28 10:05:45 -07:00
|
|
|
|
|
|
|
|
# TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
|
|
|
|
|
rtc_source_set("mock_rtp_interfaces") {
|
|
|
|
|
testonly = true
|
|
|
|
|
|
|
|
|
|
sources = [
|
|
|
|
|
"test/mock_rtp_packet_sink_interface.h",
|
2018-02-21 13:07:13 +01:00
|
|
|
"test/mock_rtp_transport_controller_send.h",
|
2017-07-28 10:05:45 -07:00
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
":rtp_interfaces",
|
2018-06-21 16:58:01 +02:00
|
|
|
"../api:libjingle_peerconnection_api",
|
2018-02-21 17:30:36 +01:00
|
|
|
"../modules/congestion_controller",
|
|
|
|
|
"../modules/pacing",
|
|
|
|
|
"../rtc_base:rate_limiter",
|
|
|
|
|
"../rtc_base:rtc_base",
|
2018-10-15 21:06:26 +02:00
|
|
|
"../rtc_base/network:sent_packet",
|
2017-07-28 10:05:45 -07:00
|
|
|
"../test:test_support",
|
|
|
|
|
]
|
|
|
|
|
}
|
2018-04-19 08:27:19 +02:00
|
|
|
rtc_source_set("mock_bitrate_allocator") {
|
|
|
|
|
testonly = true
|
Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
|
|
|
|
2018-04-19 08:27:19 +02:00
|
|
|
sources = [
|
|
|
|
|
"test/mock_bitrate_allocator.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
":bitrate_allocator",
|
2018-10-30 21:12:42 +01:00
|
|
|
"../test:test_support",
|
2018-04-19 08:27:19 +02:00
|
|
|
]
|
|
|
|
|
}
|
Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
|
|
|
rtc_source_set("mock_call_interfaces") {
|
|
|
|
|
testonly = true
|
|
|
|
|
|
|
|
|
|
sources = [
|
|
|
|
|
"test/mock_audio_send_stream.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
":call_interfaces",
|
2018-10-30 21:12:42 +01:00
|
|
|
"../test:test_support",
|
Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
|
|
|
]
|
|
|
|
|
}
|
2018-03-14 15:16:50 +01:00
|
|
|
|
|
|
|
|
rtc_test("fake_network_unittests") {
|
2018-10-15 15:25:34 +02:00
|
|
|
sources = [
|
|
|
|
|
"test/fake_network_pipe_unittest.cc",
|
|
|
|
|
]
|
2018-03-14 15:16:50 +01:00
|
|
|
deps = [
|
|
|
|
|
":call_interfaces",
|
|
|
|
|
":fake_network",
|
2018-08-16 15:51:07 +02:00
|
|
|
":simulated_network",
|
2018-03-14 15:16:50 +01:00
|
|
|
"../modules/rtp_rtcp",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
|
|
|
|
"../system_wrappers",
|
|
|
|
|
"../test:test_common",
|
|
|
|
|
"../test:test_main",
|
2018-10-30 21:12:42 +01:00
|
|
|
"../test:test_support",
|
2018-03-14 15:16:50 +01:00
|
|
|
"//testing/gtest",
|
|
|
|
|
]
|
|
|
|
|
}
|
2016-06-14 12:52:54 +02:00
|
|
|
}
|