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Move RtpPacketSender and merge it with RtpPacketPacer.

Browse Source 
This interface is intended to only handle packet-sending parts of the
paced sender.

See https://webrtc-review.googlesource.com/c/src/+/145212 for context

Bug: webrtc:10809
Change-Id: I93f0b40e1865665c2d436db67021350a0ed0687b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145216
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28662}
This commit is contained in:
Erik Språng
2019-07-24 14:52:55 +02:00
committed by Commit Bot
parent 55c4a42099
commit aa59eca891
17 changed files with 100 additions and 106 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
audio/audio_send_stream.cc
View File

@ -332,7 +332,7 @@ void AudioSendStream::Start() {
if (allocation_settings_.IncludeAudioInAllocationOnStart(
config_.min_bitrate_bps, config_.max_bitrate_bps, config_.has_dscp,
TransportSeqNumId(config_))) {
rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] {
@ -796,7 +796,7 @@ void AudioSendStream::ReconfigureBitrateObserver(
if (stream->allocation_settings_.IncludeAudioInAllocationOnReconfigure(
new_config.min_bitrate_bps, new_config.max_bitrate_bps,
new_config.has_dscp, TransportSeqNumId(new_config))) {
stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
stream->rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtc::Event thread_sync_event;
stream->worker_queue_->PostTask([&] {
RTC_DCHECK_RUN_ON(stream->worker_queue_);
@ -813,7 +813,7 @@ void AudioSendStream::ReconfigureBitrateObserver(
thread_sync_event.Wait(rtc::Event::kForever);
stream->rtp_rtcp_module_->SetAsPartOfAllocation(true);
} else {
stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(false);
stream->rtp_transport_->AccountForAudioPacketsInPacedSender(false);
stream->RemoveBitrateObserver();
stream->rtp_rtcp_module_->SetAsPartOfAllocation(false);
}

12
audio/channel_send.cc
View File

@ -365,11 +365,11 @@ class TransportSequenceNumberProxy : public TransportSequenceNumberAllocator {
TransportSequenceNumberAllocator* seq_num_allocator_ RTC_GUARDED_BY(&crit_);
};
class RtpPacketSenderProxy : public RtpPacketPacer {
class RtpPacketSenderProxy : public RtpPacketSender {
public:
RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {}
void SetPacketPacer(RtpPacketPacer* rtp_packet_pacer) {
void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) {
RTC_DCHECK(thread_checker_.IsCurrent());
rtc::CritScope lock(&crit_);
rtp_packet_pacer_ = rtp_packet_pacer;
@ -394,14 +394,10 @@ class RtpPacketSenderProxy : public RtpPacketPacer {
}
}
void SetAccountForAudioPackets(bool account_for_audio) override {
RTC_NOTREACHED();
}
private:
rtc::ThreadChecker thread_checker_;
rtc::CriticalSection crit_;
RtpPacketPacer* rtp_packet_pacer_ RTC_GUARDED_BY(&crit_);
RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&crit_);
};
class VoERtcpObserver : public RtcpBandwidthObserver {
@ -1005,7 +1001,7 @@ void ChannelSend::RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RtpPacketPacer* rtp_packet_pacer = transport->packet_sender();
RtpPacketSender* rtp_packet_pacer = transport->packet_sender();
TransportFeedbackObserver* transport_feedback_observer =
transport->transport_feedback_observer();
PacketRouter* packet_router = transport->packet_router();

9
call/call.cc
View File

@ -503,12 +503,13 @@ Call::~Call() {
call_stats_->DeregisterStatsObserver(&receive_side_cc_);
}
int64_t first_sent_packet_ms = transport_send_->GetFirstPacketTimeMs();
absl::optional<int64_t> first_sent_packet_ms =
transport_send_->GetFirstPacketTimeMs();
// Only update histograms after process threads have been shut down, so that
// they won't try to concurrently update stats.
{
if (first_sent_packet_ms) {
rtc::CritScope lock(&bitrate_crit_);
UpdateSendHistograms(first_sent_packet_ms);
UpdateSendHistograms(*first_sent_packet_ms);
}
UpdateReceiveHistograms();
UpdateHistograms();
@ -619,8 +620,6 @@ void Call::UpdateHistograms() {
}
void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
if (first_sent_packet_ms == -1)
return;
int64_t elapsed_sec =
(clock_->TimeInMilliseconds() - first_sent_packet_ms) / 1000;
if (elapsed_sec < metrics::kMinRunTimeInSeconds)

7
call/rtp_transport_controller_send.cc
View File

@ -167,7 +167,7 @@ RtpTransportControllerSend::transport_feedback_observer() {
return this;
}
RtpPacketPacer* RtpTransportControllerSend::packet_sender() {
RtpPacketSender* RtpTransportControllerSend::packet_sender() {
return &pacer_;
}
@ -398,6 +398,11 @@ void RtpTransportControllerSend::OnTransportOverheadChanged(
}
}
void RtpTransportControllerSend::AccountForAudioPacketsInPacedSender(
bool account_for_audio) {
pacer_.SetAccountForAudioPackets(account_for_audio);
}
void RtpTransportControllerSend::OnReceivedEstimatedBitrate(uint32_t bitrate) {
RemoteBitrateReport msg;
msg.receive_time = Timestamp::ms(clock_->TimeInMilliseconds());

4
call/rtp_transport_controller_send.h
View File

@ -76,7 +76,7 @@ class RtpTransportControllerSend final
NetworkStateEstimateObserver* network_state_estimate_observer() override;
TransportFeedbackObserver* transport_feedback_observer() override;
RtpPacketPacer* packet_sender() override;
RtpPacketSender* packet_sender() override;
void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
int max_padding_bitrate_bps,
@ -106,6 +106,8 @@ class RtpTransportControllerSend final
void OnTransportOverheadChanged(
size_t transport_overhead_per_packet) override;
void AccountForAudioPacketsInPacedSender(bool account_for_audio) override;
// Implements RtcpBandwidthObserver interface
void OnReceivedEstimatedBitrate(uint32_t bitrate) override;
void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,

6
call/rtp_transport_controller_send_interface.h
View File

@ -27,7 +27,7 @@
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
@ -119,7 +119,7 @@ class RtpTransportControllerSendInterface {
virtual NetworkStateEstimateObserver* network_state_estimate_observer() = 0;
virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
virtual RtpPacketPacer* packet_sender() = 0;
virtual RtpPacketSender* packet_sender() = 0;
// SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
// settings.
@ -160,6 +160,8 @@ class RtpTransportControllerSendInterface {
virtual void OnTransportOverheadChanged(
size_t transport_overhead_per_packet) = 0;
virtual void AccountForAudioPacketsInPacedSender(bool account_for_audio) = 0;
};
} // namespace webrtc

3
call/test/mock_rtp_transport_controller_send.h
View File

@ -48,7 +48,7 @@ class MockRtpTransportControllerSend
MOCK_METHOD0(network_state_estimate_observer,
NetworkStateEstimateObserver*());
MOCK_METHOD0(transport_feedback_observer, TransportFeedbackObserver*());
MOCK_METHOD0(packet_sender, RtpPacketPacer*());
MOCK_METHOD0(packet_sender, RtpPacketSender*());
MOCK_METHOD3(SetAllocatedSendBitrateLimits, void(int, int, int));
MOCK_METHOD1(SetPacingFactor, void(float));
MOCK_METHOD1(SetQueueTimeLimit, void(int));
@ -67,6 +67,7 @@ class MockRtpTransportControllerSend
MOCK_METHOD1(SetSdpBitrateParameters, void(const BitrateConstraints&));
MOCK_METHOD1(SetClientBitratePreferences, void(const BitrateSettings&));
MOCK_METHOD1(OnTransportOverheadChanged, void(size_t));
MOCK_METHOD1(AccountForAudioPacketsInPacedSender, void(bool));
MOCK_METHOD1(OnReceivedPacket, void(const ReceivedPacket&));
};
} // namespace webrtc

4
modules/pacing/paced_sender.cc
View File

@ -212,10 +212,10 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
RtpPacketToSend::Type type;
switch (priority) {
case RtpPacketPacer::kHighPriority:
case RtpPacketSender::kHighPriority:
type = RtpPacketToSend::Type::kAudio;
break;
case RtpPacketPacer::kNormalPriority:
case RtpPacketSender::kNormalPriority:
type = RtpPacketToSend::Type::kRetransmission;
break;
default:

7
modules/pacing/paced_sender.h
View File

@ -27,8 +27,7 @@
#include "modules/pacing/interval_budget.h"
#include "modules/pacing/packet_router.h"
#include "modules/pacing/round_robin_packet_queue.h"
#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/critical_section.h"
@ -39,7 +38,7 @@ namespace webrtc {
class Clock;
class RtcEventLog;
class PacedSender : public Module, public RtpPacketPacer {
class PacedSender : public Module, public RtpPacketSender {
public:
static constexpr int64_t kNoCongestionWindow = -1;
@ -98,7 +97,7 @@ class PacedSender : public Module, public RtpPacketPacer {
// With the introduction of audio BWE audio traffic will be accounted for
// the pacer budget calculation. The audio traffic still will be injected
// at high priority.
void SetAccountForAudioPackets(bool account_for_audio) override;
void SetAccountForAudioPackets(bool account_for_audio);
// Returns the time since the oldest queued packet was enqueued.
virtual int64_t QueueInMs() const;

2
modules/rtp_rtcp/BUILD.gn
View File

@ -15,7 +15,7 @@ rtc_source_set("rtp_rtcp_format") {
"include/rtcp_statistics.h",
"include/rtp_cvo.h",
"include/rtp_header_extension_map.h",
"include/rtp_packet_pacer.h",
"include/rtp_packet_sender.h",
"include/rtp_rtcp_defines.h",
"source/byte_io.h",
"source/rtcp_packet.h",

39
modules/rtp_rtcp/include/rtp_packet_pacer.h
View File

@ -1,39 +0,0 @@
/*
* Copyright (c) 2019 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 MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_PACER_H_
#define MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_PACER_H_
#include <memory>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
namespace webrtc {
// Interface for a paced sender, as implemented in the pacing module.
// This intended to replace the RtpPacketSender interface defined in
// rtp_rtcp_defines.h
// TODO(bugs.webrtc.org/10633): Add things missing to this interface so that we
// can use multiple different pacer implementations, and stop inheriting from
// RtpPacketSender.
class RtpPacketPacer : public RtpPacketSender {
public:
RtpPacketPacer() = default;
~RtpPacketPacer() override = default;
// Insert packet into queue, for eventual transmission. Based on the type of
// the packet, it will prioritized and scheduled relative to other packets and
// the current target send rate.
virtual void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) = 0;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_PACER_H_

51
modules/rtp_rtcp/include/rtp_packet_sender.h Normal file
View File

@ -0,0 +1,51 @@
/*
* Copyright (c) 2019 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 MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
#define MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
#include <memory>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
namespace webrtc {
// TODO(bugs.webrtc.org/10633): Remove Priority and InsertPacket when old pacer
// code path is gone.
class RtpPacketSender {
public:
virtual ~RtpPacketSender() = default;
// These are part of the legacy PacedSender interface and will be removed.
enum Priority {
kHighPriority = 0, // Pass through; will be sent immediately.
kNormalPriority = 2, // Put in back of the line.
kLowPriority = 3, // Put in back of the low priority line.
};
// Adds the packet information to the queue and call TimeToSendPacket when
// it's time to send.
virtual void InsertPacket(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) = 0;
// Insert packet into queue, for eventual transmission. Based on the type of
// the packet, it will be prioritized and scheduled relative to other packets
// and the current target send rate.
virtual void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) = 0;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_

4
modules/rtp_rtcp/include/rtp_rtcp.h
View File

@ -25,7 +25,7 @@
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/constructor_magic.h"
@ -89,7 +89,7 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
RemoteBitrateEstimator* remote_bitrate_estimator = nullptr;
// Spread any bursts of packets into smaller bursts to minimize packet loss.
RtpPacketPacer* paced_sender = nullptr;
RtpPacketSender* paced_sender = nullptr;
// Generate FlexFEC packets.
// TODO(brandtr): Remove when FlexfecSender is wired up to PacedSender.

31
modules/rtp_rtcp/include/rtp_rtcp_defines.h
View File

@ -340,37 +340,6 @@ class RtcpRttStats {
virtual ~RtcpRttStats() {}
};
// This class will be deprecated and replaced with RtpPacketPacer.
class RtpPacketSender {
public:
RtpPacketSender() {}
virtual ~RtpPacketSender() {}
// These are part of the legacy PacedSender interface and will be removed.
enum Priority {
kHighPriority = 0, // Pass through; will be sent immediately.
kNormalPriority = 2, // Put in back of the line.
kLowPriority = 3, // Put in back of the low priority line.
};
// Adds the packet information to the queue and call TimeToSendPacket when
// it's time to send.
virtual void InsertPacket(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) = 0;
// Currently audio traffic is not accounted by pacer and passed through.
// With the introduction of audio BWE audio traffic will be accounted for
// the pacer budget calculation. The audio traffic still will be injected
// at high priority.
// TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
// updated to support it.
virtual void SetAccountForAudioPackets(bool account_for_audio) {}
};
class TransportSequenceNumberAllocator {
public:
TransportSequenceNumberAllocator() {}

2
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -213,7 +213,7 @@ RTPSender::RTPSender(
bool audio,
Clock* clock,
Transport* transport,
RtpPacketPacer* paced_sender,
RtpPacketSender* paced_sender,
absl::optional<uint32_t> flexfec_ssrc,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_observer,

6
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -24,7 +24,7 @@
#include "api/transport/webrtc_key_value_config.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
@ -52,7 +52,7 @@ class RTPSender {
RTPSender(bool audio,
Clock* clock,
Transport* transport,
RtpPacketPacer* paced_sender,
RtpPacketSender* paced_sender,
absl::optional<uint32_t> flexfec_ssrc,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_callback,
@ -249,7 +249,7 @@ class RTPSender {
const absl::optional<uint32_t> flexfec_ssrc_;
RtpPacketPacer* const paced_sender_;
RtpPacketSender* const paced_sender_;
TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
TransportFeedbackObserver* const transport_feedback_observer_;
rtc::CriticalSection send_critsect_;

13
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -22,7 +22,7 @@
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
@ -166,7 +166,7 @@ std::string ToFieldTrialString(TestConfig config) {
} // namespace
class MockRtpPacketPacer : public RtpPacketPacer {
class MockRtpPacketPacer : public RtpPacketSender {
public:
MockRtpPacketPacer() {}
virtual ~MockRtpPacketPacer() {}
@ -180,6 +180,15 @@ class MockRtpPacketPacer : public RtpPacketPacer {
int64_t capture_time_ms,
size_t bytes,
bool retransmission));
MOCK_METHOD2(CreateProbeCluster, void(int bitrate_bps, int cluster_id));
MOCK_METHOD0(Pause, void());
MOCK_METHOD0(Resume, void());
MOCK_METHOD1(SetCongestionWindow,
void(absl::optional<int64_t> congestion_window_bytes));
MOCK_METHOD1(UpdateOutstandingData, void(int64_t outstanding_bytes));
MOCK_METHOD1(SetAccountForAudioPackets, void(bool account_for_audio));
};
class MockTransportSequenceNumberAllocator