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New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

Browse Source 
BUG=webrtc:7135
TBR=sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2913143003
Cr-Commit-Position: refs/heads/master@{#18371}
This commit is contained in:
nisse 2017-06-01 04:02:35 -07:00 committed by Commit Bot
parent 405f90c871
commit d76b7b294a
12 changed files with 79 additions and 23 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/audio/BUILD.gn
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@ -40,6 +40,7 @@ rtc_static_library("audio") {
"../base:rtc_base_approved",
"../base:rtc_task_queue",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
"../modules/audio_coding:cng",
"../modules/audio_device",

2
webrtc/audio/audio_receive_stream.h
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@ -19,7 +19,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/rtp_demuxer.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
#include "webrtc/call/syncable.h"
namespace webrtc {

49
webrtc/call/BUILD.gn
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@ -16,12 +16,11 @@ rtc_source_set("call_interfaces") {
"audio_state.h",
"call.h",
"flexfec_receive_stream.h",
"rtp_demuxer.h",
"rtp_transport_controller_send_interface.h",
"syncable.cc",
"syncable.h",
]
deps = [
":rtp_interfaces",
"..:video_stream_api",
"..:webrtc_common",
"../api:audio_mixer_api",
@ -33,17 +32,47 @@ rtc_source_set("call_interfaces") {
]
}
# TODO(nisse): These RTP targets should be moved elsewhere
# when interfaces have stabilized.
rtc_source_set("rtp_interfaces") {
sources = [
"rtp_packet_sink_interface.h",
"rtp_transport_controller_send_interface.h",
]
}
rtc_source_set("rtp_receiver") {
sources = [
"rtp_demuxer.cc",
"rtp_demuxer.h",
"rtx_receive_stream.cc",
"rtx_receive_stream.h",
]
deps = [
":rtp_interfaces",
"../base:rtc_base_approved",
"../modules/rtp_rtcp",
]
}
rtc_source_set("rtp_sender") {
sources = [
"rtp_transport_controller_send.cc",
"rtp_transport_controller_send.h",
]
deps = [
":rtp_interfaces",
"../base:rtc_base_approved",
"../modules/congestion_controller",
]
}
rtc_static_library("call") {
sources = [
"bitrate_allocator.cc",
"call.cc",
"flexfec_receive_stream_impl.cc",
"flexfec_receive_stream_impl.h",
"rtp_demuxer.cc",
"rtp_transport_controller_send.cc",
"rtp_transport_controller_send.h",
"rtx_receive_stream.cc",
"rtx_receive_stream.h",
]
if (!build_with_chromium && is_clang) {
@ -58,6 +87,9 @@ rtc_static_library("call") {
deps = [
":call_interfaces",
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
"..:webrtc_common",
"../api:transport_api",
"../audio",
@ -94,6 +126,9 @@ if (rtc_include_tests) {
]
deps = [
":call",
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
"../api:mock_audio_mixer",
"../base:rtc_base_approved",
"../logging:rtc_event_log_api",

2
webrtc/call/flexfec_receive_stream_impl.h
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@ -15,7 +15,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/call/flexfec_receive_stream.h"
#include "webrtc/call/rtp_demuxer.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
namespace webrtc {

1
webrtc/call/rtp_demuxer.cc
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@ -10,6 +10,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/call/rtp_demuxer.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
namespace webrtc {

8
webrtc/call/rtp_demuxer.h
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@ -15,13 +15,7 @@
namespace webrtc {
class RtpPacketReceived;
// This class represents a receiver of an already parsed RTP packets.
class RtpPacketSinkInterface {
public:
virtual ~RtpPacketSinkInterface() {}
virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0;
};
class RtpPacketSinkInterface;
// This class represents the RTP demuxing, for a single RTP session (i.e., one
// ssrc space, see RFC 7656). It isn't thread aware, leaving responsibility of

1
webrtc/call/rtp_demuxer_unittest.cc
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@ -15,6 +15,7 @@
#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/ptr_util.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"

26
webrtc/call/rtp_packet_sink_interface.h Normal file
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@ -0,0 +1,26 @@
/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_H_
#define WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_H_
namespace webrtc {
class RtpPacketReceived;
// This class represents a receiver of an already parsed RTP packets.
class RtpPacketSinkInterface {
public:
virtual ~RtpPacketSinkInterface() {}
virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0;
};
} // namespace webrtc
#endif // WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_H_

2
webrtc/call/rtx_receive_stream.h
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@ -13,7 +13,7 @@
#include <map>
#include "webrtc/call/rtp_demuxer.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
namespace webrtc {

1
webrtc/video/BUILD.gn
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@ -62,6 +62,7 @@ rtc_static_library("video") {
"../base:rtc_numerics",
"../base:rtc_task_queue",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
"../media:rtc_media_base",

2
webrtc/video/video_receive_stream.h
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@ -15,7 +15,7 @@
#include <vector>
#include "webrtc/base/thread_checker.h"
#include "webrtc/call/rtp_demuxer.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
#include "webrtc/call/syncable.h"
#include "webrtc/common_video/include/incoming_video_stream.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"

7
webrtc/voice_engine/BUILD.gn
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@ -144,13 +144,10 @@ rtc_static_library("voice_engine") {
"../audio/utility:audio_frame_operations",
"../base:rtc_base_approved",
"../base:rtc_task_queue",
"../modules:module_api",
# TODO(nisse): Delete when declaration of RtpTransportController
# and related interfaces move to api/.
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:rent_a_codec",
"../modules/audio_conference_mixer",