webrtc_m130/modules/rtp_rtcp/source/playout_delay_oracle.cc
Erik Språng 632a03c0cd Revert "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
This reverts commit 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2.

Reason for revert: Breaks downstream project

Original change's description:
> Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery
> 
> The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
> header extension was successfully propagated to the receiving side. Once
> it was determined that the receiver had received a frame with the new
> delay tag, it's no longer necessary to propagate.
> 
> The issue with this implementation is that it is based on max
> extended sequence number reported via RTCP, which makes it often slow
> to react, could theoretically fail to produce desired outcome (max
> received > X does not guarantee X was fully received and decoded), and
> added a lot of code complexity.
> 
> The guarantee of delivery can in fact be accomplished more reliably and
> with less code by making sure to tag each frame until an undiscardable
> frame is sent.
> 
> This allows containing the logic fully within RTPSenderVideo.
> 
> Bug: webrtc:11340
> Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30473}

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Change-Id: Ide922e680ae36bb69b95e58002482cf5ed57e254
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30475}
2020-02-06 16:05:02 +00:00

91 lines
3.0 KiB
C++

/*
* Copyright (c) 2016 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.
*/
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include <algorithm>
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
PlayoutDelayOracle::PlayoutDelayOracle() = default;
PlayoutDelayOracle::~PlayoutDelayOracle() = default;
absl::optional<PlayoutDelay> PlayoutDelayOracle::PlayoutDelayToSend(
PlayoutDelay requested_delay) const {
rtc::CritScope lock(&crit_sect_);
if (requested_delay.min_ms > PlayoutDelayLimits::kMaxMs ||
requested_delay.max_ms > PlayoutDelayLimits::kMaxMs) {
RTC_DLOG(LS_ERROR)
<< "Requested playout delay values out of range, ignored";
return absl::nullopt;
}
if (requested_delay.max_ms != -1 &&
requested_delay.min_ms > requested_delay.max_ms) {
RTC_DLOG(LS_ERROR) << "Requested playout delay values out of order";
return absl::nullopt;
}
if ((requested_delay.min_ms == -1 ||
requested_delay.min_ms == latest_delay_.min_ms) &&
(requested_delay.max_ms == -1 ||
requested_delay.max_ms == latest_delay_.max_ms)) {
// Unchanged.
return unacked_sequence_number_ ? absl::make_optional(latest_delay_)
: absl::nullopt;
}
if (requested_delay.min_ms == -1) {
RTC_DCHECK_GE(requested_delay.max_ms, 0);
requested_delay.min_ms =
std::min(latest_delay_.min_ms, requested_delay.max_ms);
}
if (requested_delay.max_ms == -1) {
requested_delay.max_ms =
std::max(latest_delay_.max_ms, requested_delay.min_ms);
}
return requested_delay;
}
void PlayoutDelayOracle::OnSentPacket(uint16_t sequence_number,
absl::optional<PlayoutDelay> delay) {
rtc::CritScope lock(&crit_sect_);
int64_t unwrapped_sequence_number = unwrapper_.Unwrap(sequence_number);
if (!delay) {
return;
}
RTC_DCHECK_LE(0, delay->min_ms);
RTC_DCHECK_LE(delay->max_ms, PlayoutDelayLimits::kMaxMs);
RTC_DCHECK_LE(delay->min_ms, delay->max_ms);
if (delay->min_ms != latest_delay_.min_ms ||
delay->max_ms != latest_delay_.max_ms) {
latest_delay_ = *delay;
unacked_sequence_number_ = unwrapped_sequence_number;
}
}
// If an ACK is received on the packet containing the playout delay extension,
// we stop sending the extension on future packets.
void PlayoutDelayOracle::OnReceivedAck(
int64_t extended_highest_sequence_number) {
rtc::CritScope lock(&crit_sect_);
if (unacked_sequence_number_ &&
extended_highest_sequence_number > *unacked_sequence_number_) {
unacked_sequence_number_ = absl::nullopt;
}
}
} // namespace webrtc