webrtc/call/rtp_config.h
Henrik Boström f45ca3787f [Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
--- Background ---
The webrtc::VideoSendStream::StreamStats are converted into
VideoSenderInfo objects which turn into "outbound-rtp" stats objects in
getStats() (or "ssrc" objects in legacy getStats()).

StreamStats are created for each type of substream: RTP media streams,
RTX streams and FlexFEC streams - each with individual packet counters.

The RTX stream is responsible for retransmissions of a referenced media
stream and the FlexFEC stream is responsible for FEC of a referenced
media stream. RTX/FEC streams do not show up as separate objects in
getStats(). Only the media streams become "outbound-rtp" objects, but
their packet and byte counters have to include the RTX and FEC counters.

--- Overview of this CL ---
This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes
StreamStats of all kinds as input, and outputs media-only StreamStats
- incorporating the RTX and FEC counters into the relevant media
StreamStats.

The merged StreamStats objects is a smaller set of objects than the
non-merged counterparts, but when aggregating all packet counters
together we end up with exact same packet and count as before.

Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates
the StreamStats into a single VideoSenderInfo (single "outbound-rtp"),
this CL should not have any observable side-effects. Prior to this CL:
aggregate StreamStats. After this CL: merge StreamStats and then
aggregate them.

However, when simulcast stats are implemented (WIP CL:
https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media
stream should turn into an individual "outbound-rtp" object. We will
then no longer aggregate all StreamStats into a single "info". This CL
unblocks simulcast stats by providing StreamStats objects that could be
turned into individual VideoSenderInfos.

--- The Changes ---
1. Methods added to RtpConfig to be able to easily tell the relationship
   between RTP, RTX and FEC ssrcs.
2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that
   replaces the booleans (is_rtx, is_flexfec).
3. "referenced_media_ssrc" is added to StreamStats, making it possible
   to tell which kRtx/kFlexFec stream stats need to be merged with which
   kMedia StreamStats.
4. MergeInfoAboutOutboundRtpSubstreams() added and used.

Bug: webrtc:11439
Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30869}
2020-03-24 13:31:54 +00:00

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5.2 KiB
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/*
* 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 CALL_RTP_CONFIG_H_
#define CALL_RTP_CONFIG_H_
#include <stddef.h>
#include <stdint.h>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
namespace webrtc {
// Currently only VP8/VP9 specific.
struct RtpPayloadState {
int16_t picture_id = -1;
uint8_t tl0_pic_idx = 0;
int64_t shared_frame_id = 0;
};
// Settings for LNTF (LossNotification). Still highly experimental.
struct LntfConfig {
std::string ToString() const;
bool enabled{false};
};
// Settings for NACK, see RFC 4585 for details.
struct NackConfig {
NackConfig() : rtp_history_ms(0) {}
std::string ToString() const;
// Send side: the time RTP packets are stored for retransmissions.
// Receive side: the time the receiver is prepared to wait for
// retransmissions.
// Set to '0' to disable.
int rtp_history_ms;
};
// Settings for ULPFEC forward error correction.
// Set the payload types to '-1' to disable.
struct UlpfecConfig {
UlpfecConfig()
: ulpfec_payload_type(-1),
red_payload_type(-1),
red_rtx_payload_type(-1) {}
std::string ToString() const;
bool operator==(const UlpfecConfig& other) const;
// Payload type used for ULPFEC packets.
int ulpfec_payload_type;
// Payload type used for RED packets.
int red_payload_type;
// RTX payload type for RED payload.
int red_rtx_payload_type;
};
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
struct RtpConfig {
RtpConfig();
RtpConfig(const RtpConfig&);
~RtpConfig();
std::string ToString() const;
std::vector<uint32_t> ssrcs;
// The Rtp Stream Ids (aka RIDs) to send in the RID RTP header extension
// if the extension is included in the list of extensions.
// If rids are specified, they should correspond to the |ssrcs| vector.
// This means that:
// 1. rids.size() == 0 || rids.size() == ssrcs.size().
// 2. If rids is not empty, then |rids[i]| should use |ssrcs[i]|.
std::vector<std::string> rids;
// The value to send in the MID RTP header extension if the extension is
// included in the list of extensions.
std::string mid;
// See RtcpMode for description.
RtcpMode rtcp_mode = RtcpMode::kCompound;
// Max RTP packet size delivered to send transport from VideoEngine.
size_t max_packet_size = kDefaultMaxPacketSize;
// Corresponds to the SDP attribute extmap-allow-mixed.
bool extmap_allow_mixed = false;
// RTP header extensions to use for this send stream.
std::vector<RtpExtension> extensions;
// TODO(nisse): For now, these are fixed, but we'd like to support
// changing codec without recreating the VideoSendStream. Then these
// fields must be removed, and association between payload type and codec
// must move above the per-stream level. Ownership could be with
// RtpTransportControllerSend, with a reference from PayloadRouter, where
// the latter would be responsible for mapping the codec type of encoded
// images to the right payload type.
std::string payload_name;
int payload_type = -1;
// Payload should be packetized using raw packetizer (payload header will
// not be added, additional meta data is expected to be present in generic
// frame descriptor RTP header extension).
bool raw_payload = false;
// See LntfConfig for description.
LntfConfig lntf;
// See NackConfig for description.
NackConfig nack;
// See UlpfecConfig for description.
UlpfecConfig ulpfec;
struct Flexfec {
Flexfec();
Flexfec(const Flexfec&);
~Flexfec();
// Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
int payload_type = -1;
// SSRC of FlexFEC stream.
uint32_t ssrc = 0;
// Vector containing a single element, corresponding to the SSRC of the
// media stream being protected by this FlexFEC stream.
// The vector MUST have size 1.
//
// TODO(brandtr): Update comment above when we support
// multistream protection.
std::vector<uint32_t> protected_media_ssrcs;
} flexfec;
// Settings for RTP retransmission payload format, see RFC 4588 for
// details.
struct Rtx {
Rtx();
Rtx(const Rtx&);
~Rtx();
std::string ToString() const;
// SSRCs to use for the RTX streams.
std::vector<uint32_t> ssrcs;
// Payload type to use for the RTX stream.
int payload_type = -1;
} rtx;
// RTCP CNAME, see RFC 3550.
std::string c_name;
bool IsMediaSsrc(uint32_t ssrc) const;
bool IsRtxSsrc(uint32_t ssrc) const;
bool IsFlexfecSsrc(uint32_t ssrc) const;
absl::optional<uint32_t> GetRtxSsrcAssociatedWithMediaSsrc(
uint32_t media_ssrc) const;
uint32_t GetMediaSsrcAssociatedWithRtxSsrc(uint32_t rtx_ssrc) const;
uint32_t GetMediaSsrcAssociatedWithFlexfecSsrc(uint32_t flexfec_ssrc) const;
};
} // namespace webrtc
#endif // CALL_RTP_CONFIG_H_