Wire up rtcp xr target bitrate on receive side.

BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2540363003
Cr-Commit-Position: refs/heads/master@{#15388}
This commit is contained in:
sprang
2016-12-02 07:29:44 -08:00
committed by Commit bot
parent 9afbfc00d8
commit a790d834c9
7 changed files with 131 additions and 66 deletions

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@ -34,6 +34,7 @@ class RemoteBitrateEstimator;
class RtcEventLog;
class RtpReceiver;
class Transport;
class VideoBitrateAllocationObserver;
RTPExtensionType StringToRtpExtensionType(const std::string& extension);
@ -68,6 +69,7 @@ class RtpRtcp : public Module {
RtcpBandwidthObserver* bandwidth_callback = nullptr;
TransportFeedbackObserver* transport_feedback_callback = nullptr;
VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr;
RtcpRttStats* rtt_stats = nullptr;
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;

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@ -22,6 +22,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/common_video/include/video_bitrate_allocator.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
@ -67,6 +68,7 @@ struct RTCPReceiver::PacketInformation {
uint64_t rpsi_picture_id = 0;
uint32_t receiver_estimated_max_bitrate_bps = 0;
std::unique_ptr<rtcp::TransportFeedback> transport_feedback;
rtc::Optional<BitrateAllocation> target_bitrate_allocation;
};
struct RTCPReceiver::ReceiveInformation {
@ -103,13 +105,15 @@ RTCPReceiver::RTCPReceiver(
RtcpBandwidthObserver* rtcp_bandwidth_observer,
RtcpIntraFrameObserver* rtcp_intra_frame_observer,
TransportFeedbackObserver* transport_feedback_observer,
VideoBitrateAllocationObserver* bitrate_allocation_observer,
ModuleRtpRtcp* owner)
: _clock(clock),
receiver_only_(receiver_only),
_rtpRtcp(*owner),
_cbRtcpBandwidthObserver(rtcp_bandwidth_observer),
_cbRtcpIntraFrameObserver(rtcp_intra_frame_observer),
_cbTransportFeedbackObserver(transport_feedback_observer),
rtcp_bandwidth_observer_(rtcp_bandwidth_observer),
rtcp_intra_frame_observer_(rtcp_intra_frame_observer),
transport_feedback_observer_(transport_feedback_observer),
bitrate_allocation_observer_(bitrate_allocation_observer),
main_ssrc_(0),
_remoteSSRC(0),
_remoteSenderInfo(),
@ -117,7 +121,7 @@ RTCPReceiver::RTCPReceiver(
xr_rr_rtt_ms_(0),
_lastReceivedRrMs(0),
_lastIncreasedSequenceNumberMs(0),
stats_callback_(NULL),
stats_callback_(nullptr),
packet_type_counter_observer_(packet_type_counter_observer),
num_skipped_packets_(0),
last_skipped_packets_warning_(clock->TimeInMilliseconds()) {
@ -627,7 +631,7 @@ void RTCPReceiver::HandleSDES(const CommonHeader& rtcp_block,
for (const rtcp::Sdes::Chunk& chunk : sdes.chunks()) {
received_cnames_[chunk.ssrc] = chunk.cname;
{
rtc::CritScope lock(&_criticalSectionFeedbacks);
rtc::CritScope lock(&feedbacks_lock_);
if (stats_callback_)
stats_callback_->CNameChanged(chunk.cname.c_str(), chunk.ssrc);
}
@ -693,6 +697,9 @@ void RTCPReceiver::HandleXr(const CommonHeader& rtcp_block,
for (const rtcp::ReceiveTimeInfo& time_info : xr.dlrr().sub_blocks())
HandleXrDlrrReportBlock(time_info);
if (xr.target_bitrate())
HandleXrTargetBitrate(*xr.target_bitrate(), packet_information);
}
void RTCPReceiver::HandleXrReceiveReferenceTime(
@ -725,6 +732,17 @@ void RTCPReceiver::HandleXrDlrrReportBlock(const rtcp::ReceiveTimeInfo& rti) {
xr_rr_rtt_ms_ = CompactNtpRttToMs(rtt_ntp);
}
void RTCPReceiver::HandleXrTargetBitrate(
const rtcp::TargetBitrate& target_bitrate,
PacketInformation* packet_information) {
BitrateAllocation bitrate_allocation;
for (const auto& item : target_bitrate.GetTargetBitrates()) {
bitrate_allocation.SetBitrate(item.spatial_layer, item.temporal_layer,
item.target_bitrate_kbps * 1000);
}
packet_information->target_bitrate_allocation.emplace(bitrate_allocation);
}
void RTCPReceiver::HandlePLI(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::Pli pli;
@ -898,11 +916,11 @@ void RTCPReceiver::UpdateTmmbr() {
std::vector<rtcp::TmmbItem> bounding =
TMMBRHelp::FindBoundingSet(TmmbrReceived());
if (!bounding.empty() && _cbRtcpBandwidthObserver) {
if (!bounding.empty() && rtcp_bandwidth_observer_) {
// We have a new bandwidth estimate on this channel.
uint64_t bitrate_bps = TMMBRHelp::CalcMinBitrateBps(bounding);
if (bitrate_bps <= std::numeric_limits<uint32_t>::max())
_cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate_bps);
rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate_bps);
}
// Set bounding set: inform remote clients about the new bandwidth.
@ -911,12 +929,12 @@ void RTCPReceiver::UpdateTmmbr() {
void RTCPReceiver::RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {
rtc::CritScope cs(&_criticalSectionFeedbacks);
rtc::CritScope cs(&feedbacks_lock_);
stats_callback_ = callback;
}
RtcpStatisticsCallback* RTCPReceiver::GetRtcpStatisticsCallback() {
rtc::CritScope cs(&_criticalSectionFeedbacks);
rtc::CritScope cs(&feedbacks_lock_);
return stats_callback_;
}
@ -947,12 +965,12 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
_rtpRtcp.OnReceivedNack(packet_information.nack_sequence_numbers);
}
}
{
// We need feedback that we have received a report block(s) so that we
// can generate a new packet in a conference relay scenario, one received
// report can generate several RTCP packets, based on number relayed/mixed
// a send report block should go out to all receivers.
if (_cbRtcpIntraFrameObserver) {
if (rtcp_intra_frame_observer_) {
RTC_DCHECK(!receiver_only_);
if ((packet_information.packet_type_flags & kRtcpPli) ||
(packet_information.packet_type_flags & kRtcpFir)) {
@ -963,30 +981,29 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
LOG(LS_VERBOSE) << "Incoming FIR from SSRC "
<< packet_information.remote_ssrc;
}
_cbRtcpIntraFrameObserver->OnReceivedIntraFrameRequest(local_ssrc);
rtcp_intra_frame_observer_->OnReceivedIntraFrameRequest(local_ssrc);
}
if (packet_information.packet_type_flags & kRtcpSli) {
_cbRtcpIntraFrameObserver->OnReceivedSLI(
rtcp_intra_frame_observer_->OnReceivedSLI(
local_ssrc, packet_information.sli_picture_id);
}
if (packet_information.packet_type_flags & kRtcpRpsi) {
_cbRtcpIntraFrameObserver->OnReceivedRPSI(
rtcp_intra_frame_observer_->OnReceivedRPSI(
local_ssrc, packet_information.rpsi_picture_id);
}
}
if (_cbRtcpBandwidthObserver) {
if (rtcp_bandwidth_observer_) {
RTC_DCHECK(!receiver_only_);
if (packet_information.packet_type_flags & kRtcpRemb) {
LOG(LS_VERBOSE)
<< "Incoming REMB: "
LOG(LS_VERBOSE) << "Incoming REMB: "
<< packet_information.receiver_estimated_max_bitrate_bps;
_cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(
rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(
packet_information.receiver_estimated_max_bitrate_bps);
}
if ((packet_information.packet_type_flags & kRtcpSr) ||
(packet_information.packet_type_flags & kRtcpRr)) {
int64_t now = _clock->TimeInMilliseconds();
_cbRtcpBandwidthObserver->OnReceivedRtcpReceiverReport(
rtcp_bandwidth_observer_->OnReceivedRtcpReceiverReport(
packet_information.report_blocks, packet_information.rtt_ms, now);
}
}
@ -995,20 +1012,25 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
_rtpRtcp.OnReceivedRtcpReportBlocks(packet_information.report_blocks);
}
if (_cbTransportFeedbackObserver &&
if (transport_feedback_observer_ &&
(packet_information.packet_type_flags & kRtcpTransportFeedback)) {
uint32_t media_source_ssrc =
packet_information.transport_feedback->media_ssrc();
if (media_source_ssrc == local_ssrc ||
registered_ssrcs.find(media_source_ssrc) != registered_ssrcs.end()) {
_cbTransportFeedbackObserver->OnTransportFeedback(
transport_feedback_observer_->OnTransportFeedback(
*packet_information.transport_feedback);
}
}
if (bitrate_allocation_observer_ &&
packet_information.target_bitrate_allocation) {
bitrate_allocation_observer_->OnBitrateAllocationUpdated(
*packet_information.target_bitrate_allocation);
}
if (!receiver_only_) {
rtc::CritScope cs(&_criticalSectionFeedbacks);
rtc::CritScope cs(&feedbacks_lock_);
if (stats_callback_) {
for (const auto& report_block : packet_information.report_blocks) {
RtcpStatistics stats;

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@ -25,10 +25,12 @@
#include "webrtc/typedefs.h"
namespace webrtc {
class VideoBitrateAllocationObserver;
namespace rtcp {
class CommonHeader;
class ReportBlock;
class Rrtr;
class TargetBitrate;
class TmmbItem;
} // namespace rtcp
@ -53,6 +55,7 @@ class RTCPReceiver {
RtcpBandwidthObserver* rtcp_bandwidth_observer,
RtcpIntraFrameObserver* rtcp_intra_frame_observer,
TransportFeedbackObserver* transport_feedback_observer,
VideoBitrateAllocationObserver* bitrate_allocation_observer,
ModuleRtpRtcp* owner);
virtual ~RTCPReceiver();
@ -163,6 +166,10 @@ class RTCPReceiver {
void HandleXrDlrrReportBlock(const rtcp::ReceiveTimeInfo& rti)
EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
void HandleXrTargetBitrate(const rtcp::TargetBitrate& target_bitrate,
PacketInformation* packet_information)
EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
void HandleNACK(const rtcp::CommonHeader& rtcp_block,
PacketInformation* packet_information)
EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
@ -210,10 +217,11 @@ class RTCPReceiver {
const bool receiver_only_;
ModuleRtpRtcp& _rtpRtcp;
rtc::CriticalSection _criticalSectionFeedbacks;
RtcpBandwidthObserver* const _cbRtcpBandwidthObserver;
RtcpIntraFrameObserver* const _cbRtcpIntraFrameObserver;
TransportFeedbackObserver* const _cbTransportFeedbackObserver;
rtc::CriticalSection feedbacks_lock_;
RtcpBandwidthObserver* const rtcp_bandwidth_observer_;
RtcpIntraFrameObserver* const rtcp_intra_frame_observer_;
TransportFeedbackObserver* const transport_feedback_observer_;
VideoBitrateAllocationObserver* const bitrate_allocation_observer_;
rtc::CriticalSection _criticalSectionRTCPReceiver;
uint32_t main_ssrc_ GUARDED_BY(_criticalSectionRTCPReceiver);
@ -247,7 +255,7 @@ class RTCPReceiver {
// delivered RTP packet to the remote side.
int64_t _lastIncreasedSequenceNumberMs;
RtcpStatisticsCallback* stats_callback_ GUARDED_BY(_criticalSectionFeedbacks);
RtcpStatisticsCallback* stats_callback_ GUARDED_BY(feedbacks_lock_);
RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
RtcpPacketTypeCounter packet_type_counter_;

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@ -14,6 +14,7 @@
#include "webrtc/base/array_view.h"
#include "webrtc/base/random.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/include/video_bitrate_allocator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
@ -96,6 +97,13 @@ class MockModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
MOCK_METHOD1(OnReceivedRtcpReportBlocks, void(const ReportBlockList&));
};
class MockVideoBitrateAllocationObserver
: public VideoBitrateAllocationObserver {
public:
MOCK_METHOD1(OnBitrateAllocationUpdated,
void(const BitrateAllocation& allocation));
};
// SSRC of remote peer, that sends rtcp packet to the rtcp receiver under test.
constexpr uint32_t kSenderSsrc = 0x10203;
// SSRCs of local peer, that rtcp packet addressed to.
@ -118,6 +126,7 @@ class RtcpReceiverTest : public ::testing::Test {
&bandwidth_observer_,
&intra_frame_observer_,
&transport_feedback_observer_,
&bitrate_allocation_observer_,
&rtp_rtcp_impl_) {}
void SetUp() {
std::set<uint32_t> ssrcs = {kReceiverMainSsrc, kReceiverExtraSsrc};
@ -142,6 +151,7 @@ class RtcpReceiverTest : public ::testing::Test {
StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_;
StrictMock<MockRtcpIntraFrameObserver> intra_frame_observer_;
StrictMock<MockTransportFeedbackObserver> transport_feedback_observer_;
StrictMock<MockVideoBitrateAllocationObserver> bitrate_allocation_observer_;
StrictMock<MockModuleRtpRtcp> rtp_rtcp_impl_;
RTCPReceiver rtcp_receiver_;
@ -1266,4 +1276,25 @@ TEST_F(RtcpReceiverTest, ForceSenderReport) {
InjectRtcpPacket(rr);
}
TEST_F(RtcpReceiverTest, ReceivesTargetBitrate) {
BitrateAllocation expected_allocation;
expected_allocation.SetBitrate(0, 0, 10000);
expected_allocation.SetBitrate(0, 1, 20000);
expected_allocation.SetBitrate(1, 0, 40000);
expected_allocation.SetBitrate(1, 1, 80000);
rtcp::TargetBitrate bitrate;
bitrate.AddTargetBitrate(0, 0, expected_allocation.GetBitrate(0, 0) / 1000);
bitrate.AddTargetBitrate(0, 1, expected_allocation.GetBitrate(0, 1) / 1000);
bitrate.AddTargetBitrate(1, 0, expected_allocation.GetBitrate(1, 0) / 1000);
bitrate.AddTargetBitrate(1, 1, expected_allocation.GetBitrate(1, 1) / 1000);
rtcp::ExtendedReports xr;
xr.SetTargetBitrate(bitrate);
EXPECT_CALL(bitrate_allocation_observer_,
OnBitrateAllocationUpdated(expected_allocation));
InjectRtcpPacket(xr);
}
} // namespace webrtc

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@ -964,6 +964,7 @@ int32_t RTCPSender::SetApplicationSpecificData(uint8_t subType,
return 0;
}
// TODO(sprang): Remove support for VoIP metrics? (Not used in receiver.)
int32_t RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) {
rtc::CritScope lock(&critical_section_rtcp_sender_);
xr_voip_metric_.emplace(*VoIPMetric);

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@ -96,6 +96,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
configuration.bandwidth_callback,
configuration.intra_frame_callback,
configuration.transport_feedback_callback,
configuration.bitrate_allocation_observer,
this),
clock_(configuration.clock),
audio_(configuration.audio),

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@ -29,7 +29,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
SimulatedClock clock(1234);
RTCPReceiver receiver(&clock, false, nullptr, nullptr, nullptr, nullptr,
&rtp_rtcp_module);
nullptr, &rtp_rtcp_module);
receiver.IncomingPacket(data, size);
}