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Add callbacks for send channel rtp statistics

Browse Source 
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
sprang@webrtc.org 2013-12-05 14:29:02 +00:00
parent 5cea89f3e1
commit ebad765ee0
11 changed files with 253 additions and 47 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

15
webrtc/common_types.h
View File

@ -235,7 +235,6 @@ protected:
// Statistics for an RTCP channel
struct RtcpStatistics {
public:
RtcpStatistics()
: fraction_lost(0),
cumulative_lost(0),
@ -259,19 +258,20 @@ class RtcpStatisticsCallback {
// Data usage statistics for a (rtp) stream
struct StreamDataCounters {
public:
StreamDataCounters()
: bytes(0),
header_bytes(0),
padding_bytes(0),
packets(0),
retransmitted_packets(0),
fec_packets(0) {}
uint32_t bytes;
uint32_t padding_bytes;
uint32_t packets;
uint32_t retransmitted_packets;
uint32_t fec_packets;
uint32_t bytes; // Payload bytes, excluding RTP headers and padding.
uint32_t header_bytes; // Number of bytes used by RTP headers.
uint32_t padding_bytes; // Number of padding bytes.
uint32_t packets; // Number of packets.
uint32_t retransmitted_packets; // Number of retransmitted packets.
uint32_t fec_packets; // Number of redundancy packets.
};
// Callback, called whenever byte/packet counts have been updated.
@ -285,7 +285,6 @@ class StreamDataCountersCallback {
// Rate statistics for a stream
struct BitrateStatistics {
public:
BitrateStatistics()
: bitrate_(0),
packet_rate(0),

6
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
View File

@ -338,6 +338,12 @@ class RtpRtcp : public Module {
virtual bool GetSendSideDelay(int* avg_send_delay_ms,
int* max_send_delay_ms) const = 0;
// Called on generation of new statistics after an RTP send.
virtual void RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback) = 0;
virtual StreamDataCountersCallback*
GetSendChannelRtpStatisticsCallback() const = 0;
/**************************************************************************
*
* RTCP

4
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -250,6 +250,10 @@ class MockRtpRtcp : public RtpRtcp {
void(FrameCountObserver*));
MOCK_CONST_METHOD0(GetSendFrameCountObserver,
FrameCountObserver*(void));
MOCK_METHOD1(RegisterSendChannelRtpStatisticsCallback,
void(StreamDataCountersCallback*));
MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
StreamDataCountersCallback*(void));
// Members.
unsigned int remote_ssrc_;
};

10
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -1667,6 +1667,16 @@ uint32_t ModuleRtpRtcpImpl::rtt_ms() const {
return rtt_ms_;
}
void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
rtp_sender_.RegisterRtpStatisticsCallback(callback);
}
StreamDataCountersCallback*
ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
return rtp_sender_.GetRtpStatisticsCallback();
}
void ModuleRtpRtcpImpl::RegisterSendFrameCountObserver(
FrameCountObserver* observer) {
rtp_sender_.RegisterFrameCountObserver(observer);

5
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -357,6 +357,11 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
virtual int32_t SendRTCPReferencePictureSelection(
const uint64_t picture_id) OVERRIDE;
virtual void RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback);
virtual StreamDataCountersCallback*
GetSendChannelRtpStatisticsCallback() const;
void OnReceivedTMMBR();
// Bad state of RTP receiver request a keyframe.

113
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -55,13 +55,14 @@ RTPSender::RTPSender(const int32_t id, const bool audio, Clock *clock,
packet_history_(clock),
// Statistics
statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
packets_sent_(0), payload_bytes_sent_(0), start_time_stamp_forced_(false),
frame_count_observer_(NULL), rtp_stats_callback_(NULL),
// RTP variables
start_time_stamp_forced_(false),
start_time_stamp_(0), ssrc_db_(*SSRCDatabase::GetSSRCDatabase()),
remote_ssrc_(0), sequence_number_forced_(false), ssrc_forced_(false),
timestamp_(0), capture_time_ms_(0), last_timestamp_time_ms_(0),
last_packet_marker_bit_(false), num_csrcs_(0), csrcs_(),
include_csrcs_(true), rtx_(kRtxOff), payload_type_rtx_(-1),
frame_counts_(), frame_count_observer_(NULL) {
include_csrcs_(true), rtx_(kRtxOff), payload_type_rtx_(-1) {
memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
memset(csrcs_, 0, sizeof(csrcs_));
@ -578,16 +579,6 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
return 0;
}
{
// Update send statistics prior to pacer.
CriticalSectionScoped lock(statistics_crit_.get());
Bitrate::Update(length);
++packets_sent_;
// We on purpose don't add to payload_bytes_sent_ since this is a
// re-transmit and not new payload data.
}
ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length);
RTPHeader header;
if (!rtp_parser.Parse(header)) {
@ -620,6 +611,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
}
if (SendPacketToNetwork(buffer_to_send_ptr, length)) {
UpdateRtpStats(buffer_to_send_ptr, length, header, rtx_ != kRtxOff, true);
return length;
}
return -1;
@ -813,7 +805,59 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
rtp_header.sequenceNumber,
rtp_header.headerLength);
}
return SendPacketToNetwork(buffer_to_send_ptr, length);
bool ret = SendPacketToNetwork(buffer_to_send_ptr, length);
UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, false, false);
return ret;
}
void RTPSender::UpdateRtpStats(const uint8_t* buffer,
uint32_t size,
const RTPHeader& header,
bool is_rtx,
bool is_retransmit) {
CriticalSectionScoped lock(statistics_crit_.get());
StreamDataCounters* counters;
uint32_t ssrc;
if (is_rtx) {
counters = &rtx_rtp_stats_;
ssrc = ssrc_rtx_;
} else {
counters = &rtp_stats_;
ssrc = ssrc_;
}
Bitrate::Update(size);
++counters->packets;
if (IsFecPacket(buffer, header)) {
++counters->fec_packets;
}
if (is_retransmit) {
++counters->retransmitted_packets;
} else {
counters->bytes += size - (header.headerLength + header.paddingLength);
counters->header_bytes += header.headerLength;
counters->padding_bytes += header.paddingLength;
}
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(*counters, ssrc);
}
}
bool RTPSender::IsFecPacket(const uint8_t* buffer,
const RTPHeader& header) const {
if (!video_) {
return false;
}
bool fec_enabled;
uint8_t pt_red;
uint8_t pt_fec;
video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
return fec_enabled &&
header.payloadType == pt_red &&
buffer[header.headerLength] == pt_fec;
}
int RTPSender::TimeToSendPadding(int bytes) {
@ -875,13 +919,6 @@ int32_t RTPSender::SendToNetwork(
storage) != 0) {
return -1;
}
{
// Update send statistics prior to pacer.
CriticalSectionScoped lock(statistics_crit_.get());
Bitrate::Update(payload_length + rtp_header_length);
++packets_sent_;
payload_bytes_sent_ += payload_length;
}
if (paced_sender_ && storage != kDontStore) {
if (!paced_sender_->SendPacket(priority, rtp_header.ssrc,
@ -895,10 +932,11 @@ int32_t RTPSender::SendToNetwork(
if (capture_time_ms > 0) {
UpdateDelayStatistics(capture_time_ms, now_ms);
}
if (SendPacketToNetwork(buffer, payload_length + rtp_header_length)) {
return 0;
}
return -1;
uint32_t length = payload_length + rtp_header_length;
if (!SendPacketToNetwork(buffer, length))
return -1;
UpdateRtpStats(buffer, length, rtp_header, false, false);
return 0;
}
void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
@ -934,19 +972,23 @@ uint16_t RTPSender::IncrementSequenceNumber() {
void RTPSender::ResetDataCounters() {
CriticalSectionScoped lock(statistics_crit_.get());
packets_sent_ = 0;
payload_bytes_sent_ = 0;
rtp_stats_ = StreamDataCounters();
rtx_rtp_stats_ = StreamDataCounters();
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc_);
rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx_);
}
}
uint32_t RTPSender::Packets() const {
CriticalSectionScoped lock(statistics_crit_.get());
return packets_sent_;
return rtp_stats_.packets + rtx_rtp_stats_.packets;
}
// Number of sent RTP bytes.
uint32_t RTPSender::Bytes() const {
CriticalSectionScoped lock(statistics_crit_.get());
return payload_bytes_sent_;
return rtp_stats_.bytes + rtx_rtp_stats_.bytes;
}
int RTPSender::CreateRTPHeader(
@ -1533,4 +1575,17 @@ FrameCountObserver* RTPSender::GetFrameCountObserver() const {
return frame_count_observer_;
}
void RTPSender::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
CriticalSectionScoped cs(statistics_crit_.get());
if (callback != NULL)
assert(rtp_stats_callback_ == NULL);
rtp_stats_callback_ = callback;
}
StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
CriticalSectionScoped cs(statistics_crit_.get());
return rtp_stats_callback_;
}
} // namespace webrtc

20
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -271,6 +271,10 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
int32_t bytes, StorageType store,
bool force_full_size_packets, bool only_pad_after_markerbit);
// Called on update of RTP statistics.
void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
StreamDataCountersCallback* GetRtpStatisticsCallback() const;
protected:
int32_t CheckPayloadType(const int8_t payload_type,
RtpVideoCodecTypes *video_type);
@ -307,6 +311,13 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
void UpdateRtpStats(const uint8_t* buffer,
uint32_t size,
const RTPHeader& header,
bool is_rtx,
bool is_retransmit);
bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
int32_t id_;
const bool audio_configured_;
RTPSenderAudio *audio_;
@ -338,9 +349,12 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
// Statistics
scoped_ptr<CriticalSectionWrapper> statistics_crit_;
uint32_t packets_sent_;
uint32_t payload_bytes_sent_;
SendDelayMap send_delays_;
std::map<FrameType, uint32_t> frame_counts_;
FrameCountObserver* frame_count_observer_;
StreamDataCounters rtp_stats_;
StreamDataCounters rtx_rtp_stats_;
StreamDataCountersCallback* rtp_stats_callback_;
// RTP variables
bool start_time_stamp_forced_;
@ -362,8 +376,6 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
int rtx_;
uint32_t ssrc_rtx_;
int payload_type_rtx_;
std::map<FrameType, uint32_t> frame_counts_;
FrameCountObserver* frame_count_observer_;
};
} // namespace webrtc

79
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -791,6 +791,85 @@ class RtpSenderAudioTest : public RtpSenderTest {
}
};
TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
class TestCallback : public StreamDataCountersCallback {
public:
TestCallback()
: StreamDataCountersCallback(), ssrc_(0), counters_() {}
virtual ~TestCallback() {}
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) {
ssrc_ = ssrc;
counters_ = counters;
}
uint32_t ssrc_;
StreamDataCounters counters_;
bool Matches(uint32_t ssrc, uint32_t bytes, uint32_t header_bytes,
uint32_t padding, uint32_t packets, uint32_t retransmits,
uint32_t fec) {
return ssrc_ == ssrc &&
counters_.bytes == bytes &&
counters_.header_bytes == header_bytes &&
counters_.padding_bytes == padding &&
counters_.packets == packets &&
counters_.retransmitted_packets == retransmits &&
counters_.fec_packets == fec;
}
} callback;
const uint8_t kRedPayloadType = 96;
const uint8_t kUlpfecPayloadType = 97;
const uint32_t kMaxPaddingSize = 224;
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
4321, payload, sizeof(payload),
NULL));
// {bytes = 6, header = 12, padding = 0, packets = 1, retrans = 0, fec = 0}
EXPECT_TRUE(callback.Matches(ssrc, 6, 12, 0, 1, 0, 0));
// Retransmit a frame.
uint16_t seqno = rtp_sender_->SequenceNumber() - 1;
rtp_sender_->ReSendPacket(seqno, 0);
// bytes = 6, header = 12, padding = 0, packets = 2, retrans = 1, fec = 0}
EXPECT_TRUE(callback.Matches(ssrc, 6, 12, 0, 2, 1, 0));
// Send padding.
rtp_sender_->TimeToSendPadding(kMaxPaddingSize);
// {bytes = 6, header = 24, padding = 224, packets = 3, retrans = 1, fec = 0}
EXPECT_TRUE(callback.Matches(ssrc, 6, 24, 224, 3, 1, 0));
// Send FEC.
rtp_sender_->SetGenericFECStatus(true, kRedPayloadType, kUlpfecPayloadType);
FecProtectionParams fec_params;
fec_params.fec_mask_type = kFecMaskRandom;
fec_params.fec_rate = 1;
fec_params.max_fec_frames = 1;
rtp_sender_->SetFecParameters(&fec_params, &fec_params);
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
1234, 4321, payload,
sizeof(payload), NULL));
// {bytes = 34, header = 48, padding = 224, packets = 5, retrans = 1, fec = 1}
EXPECT_TRUE(callback.Matches(ssrc, 34, 48, 224, 5, 1, 1));
rtp_sender_->RegisterRtpStatisticsCallback(NULL);
}
TEST_F(RtpSenderAudioTest, BuildRTPPacketWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->SetAudioLevelIndicationStatus(true,
kAudioLevelExtensionId));

19
webrtc/video_engine/vie_channel.cc
View File

@ -358,6 +358,7 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->RegisterSendFrameCountObserver(NULL);
rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
simulcast_rtp_rtcp_.pop_back();
removed_rtp_rtcp_.push_front(rtp_rtcp);
}
@ -416,6 +417,8 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
rtp_rtcp_->GetSendFrameCountObserver());
rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(
rtp_rtcp_->GetSendChannelRtcpStatisticsCallback());
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
rtp_rtcp_->GetSendChannelRtpStatisticsCallback());
}
// |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
// modules can be deleted after this step.
@ -428,6 +431,7 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->RegisterSendFrameCountObserver(NULL);
rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
simulcast_rtp_rtcp_.pop_back();
removed_rtp_rtcp_.push_front(rtp_rtcp);
}
@ -1349,6 +1353,21 @@ int32_t ViEChannel::GetRtpStatistics(uint32_t* bytes_sent,
return 0;
}
void ViEChannel::RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
__FUNCTION__);
rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(callback);
{
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end();
it++) {
(*it)->RegisterSendChannelRtpStatisticsCallback(callback);
}
}
}
void ViEChannel::GetBandwidthUsage(uint32_t* total_bitrate_sent,
uint32_t* video_bitrate_sent,
uint32_t* fec_bitrate_sent,

5
webrtc/video_engine/vie_channel.h
View File

@ -190,6 +190,11 @@ class ViEChannel
uint32_t* packets_sent,
uint32_t* bytes_received,
uint32_t* packets_received) const;
// Called on update of RTP statistics.
void RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback);
void GetBandwidthUsage(uint32_t* total_bitrate_sent,
uint32_t* video_bitrate_sent,
uint32_t* fec_bitrate_sent,

24
webrtc/video_engine/vie_rtp_rtcp_impl.cc
View File

@ -1167,15 +1167,27 @@ int ViERTP_RTCPImpl::DeregisterReceiveChannelRtcpStatisticsCallback(
}
int ViERTP_RTCPImpl::RegisterSendChannelRtpStatisticsCallback(
int channel, StreamDataCountersCallback* callback) {
// TODO(sprang): Implement
return -1;
int video_channel, StreamDataCountersCallback* callback) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
"%s(channel: %d)", __FUNCTION__, video_channel);
ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
ViEChannel* vie_channel = cs.Channel(video_channel);
assert(vie_channel != NULL);
vie_channel->RegisterSendChannelRtpStatisticsCallback(callback);
return 0;
}
int ViERTP_RTCPImpl::DeregisterSendChannelRtpStatisticsCallback(
int channel, StreamDataCountersCallback* callback) {
// TODO(sprang): Implement
return -1;
int video_channel, StreamDataCountersCallback* callback) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
"%s(channel: %d)", __FUNCTION__, video_channel);
ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
ViEChannel* vie_channel = cs.Channel(video_channel);
assert(vie_channel != NULL);
vie_channel->RegisterSendChannelRtpStatisticsCallback(NULL);
return 0;
}
int ViERTP_RTCPImpl::RegisterReceiveChannelRtpStatisticsCallback(