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Revert "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"

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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}
This commit is contained in:
Erik Språng
2020-02-06 16:04:48 +00:00
committed by Commit Bot
parent 67dba30178
commit 632a03c0cd
18 changed files with 278 additions and 162 deletions
Download Patch File Download Diff File Expand all files Collapse all files

15
call/rtp_video_sender.cc
View File

@ -24,6 +24,7 @@
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/include/video_codec_interface.h"
@ -36,9 +37,13 @@ namespace webrtc {
namespace webrtc_internal_rtp_video_sender {
RtpStreamSender::RtpStreamSender(std::unique_ptr<RtpRtcp> rtp_rtcp,
RtpStreamSender::RtpStreamSender(
std::unique_ptr<PlayoutDelayOracle> playout_delay_oracle,
std::unique_ptr<RtpRtcp> rtp_rtcp,
std::unique_ptr<RTPSenderVideo> sender_video)
: rtp_rtcp(std::move(rtp_rtcp)), sender_video(std::move(sender_video)) {}
: playout_delay_oracle(std::move(playout_delay_oracle)),
rtp_rtcp(std::move(rtp_rtcp)),
sender_video(std::move(sender_video)) {}
RtpStreamSender::~RtpStreamSender() = default;
@ -172,7 +177,9 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
configuration.local_media_ssrc) !=
flexfec_protected_ssrcs.end();
configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
auto playout_delay_oracle = std::make_unique<PlayoutDelayOracle>();
configuration.ack_observer = playout_delay_oracle.get();
if (rtp_config.rtx.ssrcs.size() > i) {
configuration.rtx_send_ssrc = rtp_config.rtx.ssrcs[i];
}
@ -189,6 +196,7 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
video_config.clock = configuration.clock;
video_config.rtp_sender = rtp_rtcp->RtpSender();
video_config.flexfec_sender = configuration.flexfec_sender;
video_config.playout_delay_oracle = playout_delay_oracle.get();
video_config.frame_encryptor = frame_encryptor;
video_config.require_frame_encryption =
crypto_options.sframe.require_frame_encryption;
@ -206,7 +214,8 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
video_config.ulpfec_payload_type = rtp_config.ulpfec.ulpfec_payload_type;
}
auto sender_video = std::make_unique<RTPSenderVideo>(video_config);
rtp_streams.emplace_back(std::move(rtp_rtcp), std::move(sender_video));
rtp_streams.emplace_back(std::move(playout_delay_oracle),
std::move(rtp_rtcp), std::move(sender_video));
}
return rtp_streams;
}

4
call/rtp_video_sender.h
View File

@ -50,7 +50,8 @@ namespace webrtc_internal_rtp_video_sender {
// RTP state for a single simulcast stream. Internal to the implementation of
// RtpVideoSender.
struct RtpStreamSender {
RtpStreamSender(std::unique_ptr<RtpRtcp> rtp_rtcp,
RtpStreamSender(std::unique_ptr<PlayoutDelayOracle> playout_delay_oracle,
std::unique_ptr<RtpRtcp> rtp_rtcp,
std::unique_ptr<RTPSenderVideo> sender_video);
~RtpStreamSender();
@ -58,6 +59,7 @@ struct RtpStreamSender {
RtpStreamSender& operator=(RtpStreamSender&&) = default;
// Note: Needs pointer stability.
std::unique_ptr<PlayoutDelayOracle> playout_delay_oracle;
std::unique_ptr<RtpRtcp> rtp_rtcp;
std::unique_ptr<RTPSenderVideo> sender_video;
};

8
common_types.h
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@ -89,16 +89,8 @@ typedef SpatialLayer SimulcastStream;
// Note: Given that this gets embedded in a union, it is up-to the owner to
// initialize these values.
struct PlayoutDelay {
PlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {}
int min_ms;
int max_ms;
static PlayoutDelay Noop() { return PlayoutDelay(-1, -1); }
bool IsNoop() const { return min_ms == -1 && max_ms == -1; }
bool operator==(const PlayoutDelay& rhs) const {
return min_ms == rhs.min_ms && max_ms == rhs.max_ms;
}
};
} // namespace webrtc

2
modules/rtp_rtcp/BUILD.gn
View File

@ -156,6 +156,7 @@ rtc_library("rtp_rtcp") {
"source/forward_error_correction_internal.h",
"source/packet_loss_stats.cc",
"source/packet_loss_stats.h",
"source/playout_delay_oracle.cc",
"source/playout_delay_oracle.h",
"source/receive_statistics_impl.cc",
"source/receive_statistics_impl.h",
@ -428,6 +429,7 @@ if (rtc_include_tests) {
"source/flexfec_sender_unittest.cc",
"source/nack_rtx_unittest.cc",
"source/packet_loss_stats_unittest.cc",
"source/playout_delay_oracle_unittest.cc",
"source/receive_statistics_unittest.cc",
"source/remote_ntp_time_estimator_unittest.cc",
"source/rtcp_nack_stats_unittest.cc",

1
modules/rtp_rtcp/include/rtp_rtcp.h
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@ -101,6 +101,7 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
SendPacketObserver* send_packet_observer = nullptr;
RateLimiter* retransmission_rate_limiter = nullptr;
OverheadObserver* overhead_observer = nullptr;
RtcpAckObserver* ack_observer = nullptr;
StreamDataCountersCallback* rtp_stats_callback = nullptr;
int rtcp_report_interval_ms = 0;

13
modules/rtp_rtcp/include/rtp_rtcp_defines.h
View File

@ -392,6 +392,19 @@ struct RtpReceiveStats {
RtpPacketCounter packet_counter;
};
class RtcpAckObserver {
public:
// This method is called on received report blocks matching the sender ssrc.
// TODO(nisse): Use of "extended" sequence number is a bit brittle, since the
// observer for this callback typically has its own sequence number unwrapper,
// and there's no guarantee that they are in sync. Change to pass raw sequence
// number, possibly augmented with timestamp (if available) to aid
// disambiguation.
virtual void OnReceivedAck(int64_t extended_highest_sequence_number) = 0;
virtual ~RtcpAckObserver() = default;
};
// Callback, used to notify an observer whenever new rates have been estimated.
class BitrateStatisticsObserver {
public:

3
modules/rtp_rtcp/source/nack_rtx_unittest.cc
View File

@ -21,6 +21,7 @@
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/rate_limiter.h"
@ -139,6 +140,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock;
video_config.rtp_sender = rtp_rtcp_module_->RtpSender();
video_config.playout_delay_oracle = &playout_delay_oracle_;
video_config.field_trials = &field_trials;
rtp_sender_video_ = std::make_unique<RTPSenderVideo>(video_config);
rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
@ -225,6 +227,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
std::unique_ptr<ReceiveStatistics> receive_statistics_;
std::unique_ptr<RtpRtcp> rtp_rtcp_module_;
PlayoutDelayOracle playout_delay_oracle_;
std::unique_ptr<RTPSenderVideo> rtp_sender_video_;
RtxLoopBackTransport transport_;
const std::map<int, int> rtx_associated_payload_types_ = {

90
modules/rtp_rtcp/source/playout_delay_oracle.cc Normal file
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@ -0,0 +1,90 @@
/*
* 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

58
modules/rtp_rtcp/source/playout_delay_oracle.h
View File

@ -11,12 +11,64 @@
#ifndef MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
#define MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
#include <stdint.h>
#include "absl/types/optional.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types_public.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// TODO(sprang): Remove once downstream usage is gone.
class PlayoutDelayOracle {
// This class tracks the application requests to limit minimum and maximum
// playout delay and makes a decision on whether the current RTP frame
// should include the playout out delay extension header.
//
// Playout delay can be defined in terms of capture and render time as follows:
//
// Render time = Capture time in receiver time + playout delay
//
// The application specifies a minimum and maximum limit for the playout delay
// which are both communicated to the receiver and the receiver can adapt
// the playout delay within this range based on observed network jitter.
class PlayoutDelayOracle : public RtcpAckObserver {
public:
PlayoutDelayOracle() = default;
PlayoutDelayOracle();
~PlayoutDelayOracle() override;
// The playout delay to be added to a packet. The input delays are provided by
// the application, with -1 meaning unchanged/unspecified. The output delay
// are the values to be attached to packets on the wire. Presence and value
// depends on the current input, previous inputs, and received acks from the
// remote end.
absl::optional<PlayoutDelay> PlayoutDelayToSend(
PlayoutDelay requested_delay) const;
void OnSentPacket(uint16_t sequence_number,
absl::optional<PlayoutDelay> playout_delay);
void OnReceivedAck(int64_t extended_highest_sequence_number) override;
private:
// The playout delay information is updated from the encoder thread(s).
// The sequence number feedback is updated from the worker thread.
// Guards access to data across multiple threads.
rtc::CriticalSection crit_sect_;
// The oldest sequence number on which the current playout delay values have
// been sent. When set, it means we need to attach extension to sent packets.
absl::optional<int64_t> unacked_sequence_number_ RTC_GUARDED_BY(crit_sect_);
// Sequence number unwrapper for sent packets.
// TODO(nisse): Could potentially get out of sync with the unwrapper used by
// the caller of OnReceivedAck.
SequenceNumberUnwrapper unwrapper_ RTC_GUARDED_BY(crit_sect_);
// Playout delay values on the next frame if |send_playout_delay_| is set.
PlayoutDelay latest_delay_ RTC_GUARDED_BY(crit_sect_) = {-1, -1};
RTC_DISALLOW_COPY_AND_ASSIGN(PlayoutDelayOracle);
};
} // namespace webrtc

52
modules/rtp_rtcp/source/playout_delay_oracle_unittest.cc Normal file
View File

@ -0,0 +1,52 @@
/*
* 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 "rtc_base/logging.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
constexpr int kSequenceNumber = 100;
constexpr int kMinPlayoutDelay = 0;
constexpr int kMaxPlayoutDelay = 150;
} // namespace
TEST(PlayoutDelayOracleTest, DisabledByDefault) {
PlayoutDelayOracle playout_delay_oracle;
EXPECT_FALSE(playout_delay_oracle.PlayoutDelayToSend({-1, -1}));
}
TEST(PlayoutDelayOracleTest, SendPlayoutDelayUntilSeqNumberExceeds) {
PlayoutDelayOracle playout_delay_oracle;
PlayoutDelay playout_delay = {kMinPlayoutDelay, kMaxPlayoutDelay};
playout_delay_oracle.OnSentPacket(kSequenceNumber, playout_delay);
absl::optional<PlayoutDelay> delay_to_send =
playout_delay_oracle.PlayoutDelayToSend({-1, -1});
ASSERT_TRUE(delay_to_send.has_value());
EXPECT_EQ(kMinPlayoutDelay, delay_to_send->min_ms);
EXPECT_EQ(kMaxPlayoutDelay, delay_to_send->max_ms);
// Oracle indicates playout delay should be sent if highest sequence number
// acked is lower than the sequence number of the first packet containing
// playout delay.
playout_delay_oracle.OnReceivedAck(kSequenceNumber - 1);
EXPECT_TRUE(playout_delay_oracle.PlayoutDelayToSend({-1, -1}));
// Oracle indicates playout delay should not be sent if sequence number
// acked on a matching ssrc indicates the receiver has received the playout
// delay values.
playout_delay_oracle.OnReceivedAck(kSequenceNumber + 1);
EXPECT_FALSE(playout_delay_oracle.PlayoutDelayToSend({-1, -1}));
}
} // namespace webrtc

5
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -68,6 +68,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
nack_last_time_sent_full_ms_(0),
nack_last_seq_number_sent_(0),
remote_bitrate_(configuration.remote_bitrate_estimator),
ack_observer_(configuration.ack_observer),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
if (!configuration.receiver_only) {
@ -735,7 +736,7 @@ void ModuleRtpRtcpImpl::OnReceivedNack(
void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
const ReportBlockList& report_blocks) {
if (rtp_sender_) {
if (ack_observer_) {
uint32_t ssrc = SSRC();
absl::optional<uint32_t> rtx_ssrc;
if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
@ -746,6 +747,8 @@ void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
if (ssrc == report_block.source_ssrc) {
rtp_sender_->packet_generator.OnReceivedAckOnSsrc(
report_block.extended_highest_sequence_number);
ack_observer_->OnReceivedAck(
report_block.extended_highest_sequence_number);
} else if (rtx_ssrc && *rtx_ssrc == report_block.source_ssrc) {
rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc(
report_block.extended_highest_sequence_number);

2
modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -340,6 +340,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
RemoteBitrateEstimator* const remote_bitrate_;
RtcpAckObserver* const ack_observer_;
RtcpRttStats* const rtt_stats_;
// The processed RTT from RtcpRttStats.

3
modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -17,6 +17,7 @@
#include "api/transport/field_trial_based_config.h"
#include "api/video_codecs/video_codec.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
@ -181,6 +182,7 @@ class RtpRtcpImplTest : public ::testing::Test {
RTPSenderVideo::Config video_config;
video_config.clock = &clock_;
video_config.rtp_sender = sender_.impl_->RtpSender();
video_config.playout_delay_oracle = &playout_delay_oracle_;
video_config.field_trials = &field_trials;
sender_video_ = std::make_unique<RTPSenderVideo>(video_config);
@ -199,6 +201,7 @@ class RtpRtcpImplTest : public ::testing::Test {
SimulatedClock clock_;
RtpRtcpModule sender_;
PlayoutDelayOracle playout_delay_oracle_;
std::unique_ptr<RTPSenderVideo> sender_video_;
RtpRtcpModule receiver_;
VideoCodec codec_;

18
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -649,10 +649,12 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
config.event_log = &mock_rtc_event_log_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
@ -1151,10 +1153,12 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) {
TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
const uint8_t kPayloadType = 127;
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
uint8_t payload[] = {47, 11, 32, 93, 89};
@ -1193,10 +1197,12 @@ TEST_P(RtpSenderTestWithoutPacer, SendRawVideo) {
const uint8_t kPayloadType = 111;
const uint8_t payload[] = {11, 22, 33, 44, 55};
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
@ -1238,11 +1244,13 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
rtp_sender_context_->packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull, 10);
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.flexfec_sender = &flexfec_sender;
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
@ -1322,11 +1330,13 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
rtp_sender()->SetSequenceNumber(kSeqNum);
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.flexfec_sender = &flexfec_sender;
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
@ -1594,11 +1604,13 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
rtp_sender()->SetSequenceNumber(kSeqNum);
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.flexfec_sender = &flexfec_sender;
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
// Parameters selected to generate a single FEC packet per media packet.
@ -1668,10 +1680,12 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_context_ = std::make_unique<RtpSenderContext>(config);
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
@ -1724,10 +1738,12 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
const uint8_t kPayloadType = 127;
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
RTPSenderVideo rtp_sender_video(video_config);
uint8_t payload[] = {47, 11, 32, 93, 89};
@ -1779,10 +1795,12 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacksUlpfec) {
const uint8_t kUlpfecPayloadType = 97;
const uint8_t kPayloadType = 127;
const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
PlayoutDelayOracle playout_delay_oracle;
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock_;
video_config.rtp_sender = rtp_sender();
video_config.playout_delay_oracle = &playout_delay_oracle;
video_config.field_trials = &field_trials;
video_config.red_payload_type = kRedPayloadType;
video_config.ulpfec_payload_type = kUlpfecPayloadType;

84
modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -13,7 +13,6 @@
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <limits>
#include <memory>
#include <string>
@ -241,10 +240,6 @@ const char* FrameTypeToString(VideoFrameType frame_type) {
}
#endif
bool IsNoopDelay(const PlayoutDelay& delay) {
return delay.min_ms == -1 && delay.max_ms == -1;
}
} // namespace
RTPSenderVideo::RTPSenderVideo(Clock* clock,
@ -261,6 +256,7 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
config.clock = clock;
config.rtp_sender = rtp_sender;
config.flexfec_sender = flexfec_sender;
config.playout_delay_oracle = playout_delay_oracle;
config.frame_encryptor = frame_encryptor;
config.require_frame_encryption = require_frame_encryption;
config.need_rtp_packet_infos = need_rtp_packet_infos;
@ -278,8 +274,7 @@ RTPSenderVideo::RTPSenderVideo(const Config& config)
: (kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers)),
last_rotation_(kVideoRotation_0),
transmit_color_space_next_frame_(false),
current_playout_delay_{-1, -1},
playout_delay_pending_(false),
playout_delay_oracle_(config.playout_delay_oracle),
rtp_sequence_number_map_(config.need_rtp_packet_infos
? std::make_unique<RtpSequenceNumberMap>(
kRtpSequenceNumberMapMaxEntries)
@ -301,7 +296,9 @@ RTPSenderVideo::RTPSenderVideo(const Config& config)
config.field_trials
->Lookup(kExcludeTransportSequenceNumberFromFecFieldTrial)
.find("Enabled") == 0),
absolute_capture_time_sender_(config.clock) {}
absolute_capture_time_sender_(config.clock) {
RTC_DCHECK(playout_delay_oracle_);
}
RTPSenderVideo::~RTPSenderVideo() {}
@ -524,16 +521,8 @@ bool RTPSenderVideo::SendVideo(
video_header.codec == kVideoCodecH264 &&
video_header.frame_marking.temporal_id != kNoTemporalIdx;
MaybeUpdateCurrentPlayoutDelay(video_header);
if (video_header.frame_type == VideoFrameType::kVideoFrameKey &&
!IsNoopDelay(current_playout_delay_)) {
// Force playout delay on key-frames, if set.
playout_delay_pending_ = true;
}
const absl::optional<PlayoutDelay> playout_delay =
playout_delay_pending_
? absl::optional<PlayoutDelay>(current_playout_delay_)
: absl::nullopt;
playout_delay_oracle_->PlayoutDelayToSend(video_header.playout_delay);
// According to
// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
@ -662,15 +651,6 @@ bool RTPSenderVideo::SendVideo(
MinimizeDescriptor(&video_header);
}
if (video_header.frame_type == VideoFrameType::kVideoFrameKey ||
(IsBaseLayer(video_header) &&
!(video_header.generic.has_value() ? video_header.generic->discardable
: false))) {
// This frame has guaranteed delivery, no need to populate playout
// delay extensions until it changes again.
playout_delay_pending_ = false;
}
// TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
rtc::Buffer encrypted_video_payload;
if (frame_encryptor_ != nullptr) {
@ -765,6 +745,10 @@ bool RTPSenderVideo::SendVideo(
first_sequence_number = packet->SequenceNumber();
}
if (i == 0) {
playout_delay_oracle_->OnSentPacket(packet->SequenceNumber(),
playout_delay);
}
// No FEC protection for upper temporal layers, if used.
bool protect_packet = temporal_id == 0 || temporal_id == kNoTemporalIdx;
@ -958,52 +942,4 @@ bool RTPSenderVideo::UpdateConditionalRetransmit(
return false;
}
void RTPSenderVideo::MaybeUpdateCurrentPlayoutDelay(
const RTPVideoHeader& header) {
if (IsNoopDelay(header.playout_delay)) {
return;
}
PlayoutDelay requested_delay = header.playout_delay;
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;
}
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;
}
if (!playout_delay_pending_) {
current_playout_delay_ = requested_delay;
playout_delay_pending_ = true;
return;
}
if ((requested_delay.min_ms == -1 ||
requested_delay.min_ms == current_playout_delay_.min_ms) &&
(requested_delay.max_ms == -1 ||
requested_delay.max_ms == current_playout_delay_.max_ms)) {
// No change, ignore.
return;
}
if (requested_delay.min_ms == -1) {
RTC_DCHECK_GE(requested_delay.max_ms, 0);
requested_delay.min_ms =
std::min(current_playout_delay_.min_ms, requested_delay.max_ms);
}
if (requested_delay.max_ms == -1) {
requested_delay.max_ms =
std::max(current_playout_delay_.max_ms, requested_delay.min_ms);
}
current_playout_delay_ = requested_delay;
playout_delay_pending_ = true;
}
} // namespace webrtc

13
modules/rtp_rtcp/source/rtp_sender_video.h
View File

@ -70,6 +70,7 @@ class RTPSenderVideo {
Clock* clock = nullptr;
RTPSender* rtp_sender = nullptr;
FlexfecSender* flexfec_sender = nullptr;
PlayoutDelayOracle* playout_delay_oracle = nullptr;
FrameEncryptorInterface* frame_encryptor = nullptr;
bool require_frame_encryption = false;
bool need_rtp_packet_infos = false;
@ -180,9 +181,6 @@ class RTPSenderVideo {
int64_t expected_retransmission_time_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(stats_crit_);
void MaybeUpdateCurrentPlayoutDelay(const RTPVideoHeader& header)
RTC_EXCLUSIVE_LOCKS_REQUIRED(send_checker_);
RTPSender* const rtp_sender_;
Clock* const clock_;
@ -197,11 +195,10 @@ class RTPSenderVideo {
std::unique_ptr<FrameDependencyStructure> video_structure_
RTC_GUARDED_BY(send_checker_);
// Current target playout delay.
PlayoutDelay current_playout_delay_ RTC_GUARDED_BY(send_checker_);
// Flag indicating if we need to propagate |current_playout_delay_| in order
// to guarantee it gets delivered.
bool playout_delay_pending_;
// Tracks the current request for playout delay limits from application
// and decides whether the current RTP frame should include the playout
// delay extension on header.
PlayoutDelayOracle* const playout_delay_oracle_;
// Should never be held when calling out of this class.
rtc::CriticalSection crit_;

62
modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
View File

@ -54,7 +54,6 @@ enum : int { // The first valid value is 1.
kVideoRotationExtensionId,
kVideoTimingExtensionId,
kAbsoluteCaptureTimeExtensionId,
kPlayoutDelayExtensionId
};
constexpr int kPayload = 100;
@ -88,8 +87,6 @@ class LoopbackTransportTest : public webrtc::Transport {
kFrameMarkingExtensionId);
receivers_extensions_.Register<AbsoluteCaptureTimeExtension>(
kAbsoluteCaptureTimeExtensionId);
receivers_extensions_.Register<PlayoutDelayLimits>(
kPlayoutDelayExtensionId);
}
bool SendRtp(const uint8_t* data,
@ -124,6 +121,7 @@ class TestRtpSenderVideo : public RTPSenderVideo {
config.clock = clock;
config.rtp_sender = rtp_sender;
config.flexfec_sender = flexfec_sender;
config.playout_delay_oracle = &playout_delay_oracle_;
config.field_trials = &field_trials;
return config;
}()) {}
@ -136,6 +134,7 @@ class TestRtpSenderVideo : public RTPSenderVideo {
retransmission_settings,
expected_retransmission_time_ms);
}
PlayoutDelayOracle playout_delay_oracle_;
};
class FieldTrials : public WebRtcKeyValueConfig {
@ -793,63 +792,6 @@ TEST_P(RtpSenderVideoTest, AbsoluteCaptureTime) {
EXPECT_EQ(packets_with_abs_capture_time, 1);
}
TEST_P(RtpSenderVideoTest, PopulatesPlayoutDelay) {
// Single packet frames.
constexpr size_t kPacketSize = 123;
uint8_t kFrame[kPacketSize];
rtp_module_->RegisterRtpHeaderExtension(PlayoutDelayLimits::kUri,
kPlayoutDelayExtensionId);
const PlayoutDelay kExpectedDelay = {10, 20};
// Send initial key-frame without playout delay.
RTPVideoHeader hdr;
hdr.frame_type = VideoFrameType::kVideoFrameKey;
hdr.codec = VideoCodecType::kVideoCodecVP8;
auto& vp8_header = hdr.video_type_header.emplace<RTPVideoHeaderVP8>();
vp8_header.temporalIdx = 0;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
EXPECT_FALSE(
transport_.last_sent_packet().HasExtension<PlayoutDelayLimits>());
// Set playout delay on a discardable frame.
hdr.playout_delay = kExpectedDelay;
hdr.frame_type = VideoFrameType::kVideoFrameDelta;
vp8_header.temporalIdx = 1;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
PlayoutDelay received_delay = PlayoutDelay::Noop();
ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
&received_delay));
EXPECT_EQ(received_delay, kExpectedDelay);
// Set playout delay on a non-discardable frame, the extension should still
// be populated since dilvery wasn't guaranteed on the last one.
hdr.playout_delay = PlayoutDelay::Noop(); // Inidcates "no change".
vp8_header.temporalIdx = 0;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
&received_delay));
EXPECT_EQ(received_delay, kExpectedDelay);
// The next frame does not need the extensions since it's delivery has
// already been guaranteed.
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
EXPECT_FALSE(
transport_.last_sent_packet().HasExtension<PlayoutDelayLimits>());
// Insert key-frame, we need to refresh the state here.
hdr.frame_type = VideoFrameType::kVideoFrameKey;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
&received_delay));
EXPECT_EQ(received_delay, kExpectedDelay);
}
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderVideoTest,
::testing::Bool());

5
test/fuzzers/rtp_packet_fuzzer.cc
View File

@ -99,11 +99,10 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
&feedback_request);
break;
}
case kRtpExtensionPlayoutDelay: {
PlayoutDelay playout = PlayoutDelay::Noop();
case kRtpExtensionPlayoutDelay:
PlayoutDelay playout;
packet.GetExtension<PlayoutDelayLimits>(&playout);
break;
}
case kRtpExtensionVideoContentType:
VideoContentType content_type;
packet.GetExtension<VideoContentTypeExtension>(&content_type);