2a8779763a
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}
310 lines
8.3 KiB
Plaintext
310 lines
8.3 KiB
Plaintext
# 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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import("../webrtc.gni")
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rtc_source_set("call_interfaces") {
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sources = [
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.h",
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"call.h",
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"callfactoryinterface.h",
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"flexfec_receive_stream.h",
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"syncable.cc",
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"syncable.h",
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]
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deps = [
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":rtp_interfaces",
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":video_stream_api",
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"..:webrtc_common",
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"../:typedefs",
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"../api:audio_mixer_api",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:transport_api",
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"../api/audio_codecs:audio_codecs_api",
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"../modules/audio_processing:audio_processing_statistics",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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]
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}
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# TODO(nisse): These RTP targets should be moved elsewhere
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# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
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rtc_source_set("rtp_interfaces") {
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sources = [
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"rtcp_packet_sink_interface.h",
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"rtp_config.cc",
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"rtp_config.h",
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"rtp_packet_sink_interface.h",
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"rtp_stream_receiver_controller_interface.h",
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"rtp_transport_controller_send_interface.h",
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]
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deps = [
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"../api:array_view",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("rtp_receiver") {
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sources = [
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"rtcp_demuxer.cc",
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"rtcp_demuxer.h",
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"rtp_demuxer.cc",
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"rtp_demuxer.h",
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"rtp_rtcp_demuxer_helper.cc",
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"rtp_rtcp_demuxer_helper.h",
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"rtp_stream_receiver_controller.cc",
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"rtp_stream_receiver_controller.h",
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"rtx_receive_stream.cc",
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"rtx_receive_stream.h",
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"ssrc_binding_observer.h",
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]
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deps = [
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":rtp_interfaces",
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"..:webrtc_common",
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"../api:array_view",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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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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rtc_source_set("rtp_sender") {
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sources = [
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"rtp_transport_controller_send.cc",
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"rtp_transport_controller_send.h",
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]
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deps = [
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":rtp_interfaces",
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"..:webrtc_common",
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"../modules/congestion_controller",
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"../modules/pacing",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("bitrate_allocator") {
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sources = [
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"bitrate_allocator.cc",
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"bitrate_allocator.h",
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]
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deps = [
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"../modules/bitrate_controller",
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"../rtc_base:checks",
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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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"../system_wrappers:metrics_api",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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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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"call.cc",
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"callfactory.cc",
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"callfactory.h",
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"flexfec_receive_stream_impl.cc",
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"flexfec_receive_stream_impl.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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":bitrate_allocator",
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":call_interfaces",
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":rtp_interfaces",
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":rtp_receiver",
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":rtp_sender",
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":video_stream_api",
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"..:webrtc_common",
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"../api:optional",
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"../api:transport_api",
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"../audio",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl",
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"../modules/bitrate_controller",
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"../modules/congestion_controller",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:sequenced_task_checker",
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"../system_wrappers",
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"../system_wrappers:metrics_api",
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"../video",
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]
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}
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rtc_source_set("video_stream_api") {
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# TODO(bugs.webrtc.org/6828): Remove dependency cycle:
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# //call:video_stream_api ->
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# //media:rtc_media_base ->
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# //call:call_interfaces ->
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# //call:video_stream_api
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check_includes = false
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sources = [
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"video_config.cc",
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"video_config.h",
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"video_receive_stream.cc",
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"video_receive_stream.h",
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"video_send_stream.cc",
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"video_send_stream.h",
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]
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deps = [
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":rtp_interfaces",
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"../:webrtc_common",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:transport_api",
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"../common_video:common_video",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:rtc_base_approved",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("call_tests") {
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testonly = true
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sources = [
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"bitrate_allocator_unittest.cc",
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"bitrate_estimator_tests.cc",
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"call_unittest.cc",
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"flexfec_receive_stream_unittest.cc",
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"rtcp_demuxer_unittest.cc",
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"rtp_demuxer_unittest.cc",
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"rtp_rtcp_demuxer_helper_unittest.cc",
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"rtx_receive_stream_unittest.cc",
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]
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deps = [
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":bitrate_allocator",
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":call",
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":call_interfaces",
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":mock_rtp_interfaces",
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":rtp_interfaces",
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":rtp_receiver",
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":rtp_sender",
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"..:webrtc_common",
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"../api:array_view",
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"../api:libjingle_peerconnection_api",
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"../api:mock_audio_mixer",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../logging:rtc_event_log_api",
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/bitrate_controller",
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"../modules/congestion_controller",
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"../modules/congestion_controller:mock_congestion_controller",
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"../modules/pacing",
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"../modules/pacing:mock_paced_sender",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:mock_rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility:mock_process_thread",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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"../test:audio_codec_mocks",
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"../test:direct_transport",
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"../test:test_common",
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"../test:test_support",
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"../test:video_test_common",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("call_perf_tests") {
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testonly = true
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sources = [
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"call_perf_tests.cc",
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"rampup_tests.cc",
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"rampup_tests.h",
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]
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deps = [
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":call_interfaces",
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":video_stream_api",
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"..:webrtc_common",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../logging:rtc_event_log_api",
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"../modules/audio_coding",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/rtp_rtcp",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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"../system_wrappers:metrics_default",
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"../test:direct_transport",
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"../test:fake_audio_device",
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"../test:field_trial",
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"../test:test_common",
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"../test:test_support",
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"../test:video_test_common",
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"../video",
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"../voice_engine",
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"//testing/gtest",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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# TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
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rtc_source_set("mock_rtp_interfaces") {
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testonly = true
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sources = [
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"fake_rtp_transport_controller_send.h",
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"test/mock_rtp_packet_sink_interface.h",
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]
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deps = [
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":rtp_interfaces",
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"..:webrtc_common",
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"../modules/congestion_controller:congestion_controller",
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"../modules/pacing:pacing",
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"../test:test_support",
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"//testing/gmock",
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]
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}
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rtc_source_set("mock_call_interfaces") {
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testonly = true
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sources = [
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"test/mock_audio_send_stream.h",
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]
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deps = [
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":call_interfaces",
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"//test:test_support",
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]
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}
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}
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