Add callbacks for receive channel RTP statistics

This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.

TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
sprang@webrtc.org 2014-01-23 10:00:39 +00:00
parent 91db93d24f
commit 0e93257cee
9 changed files with 300 additions and 110 deletions

View File

@ -51,9 +51,13 @@ class ReceiveStatistics : public Module {
static ReceiveStatistics* Create(Clock* clock);
// Updates the receive statistics with this packet.
virtual void IncomingPacket(const RTPHeader& rtp_header, size_t bytes,
virtual void IncomingPacket(const RTPHeader& rtp_header,
size_t bytes,
bool retransmitted) = 0;
// Increment counter for number of FEC packets received.
virtual void FecPacketReceived(uint32_t ssrc) = 0;
// Returns a map of all statisticians which have seen an incoming packet
// during the last two seconds.
virtual StatisticianMap GetActiveStatisticians() const = 0;
@ -67,12 +71,18 @@ class ReceiveStatistics : public Module {
// Called on new RTCP stats creation.
virtual void RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) = 0;
// Called on new RTP stats creation.
virtual void RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) = 0;
};
class NullReceiveStatistics : public ReceiveStatistics {
public:
virtual void IncomingPacket(const RTPHeader& rtp_header, size_t bytes,
virtual void IncomingPacket(const RTPHeader& rtp_header,
size_t bytes,
bool retransmitted) OVERRIDE;
virtual void FecPacketReceived(uint32_t ssrc) OVERRIDE;
virtual StatisticianMap GetActiveStatisticians() const OVERRIDE;
virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE;
virtual int32_t TimeUntilNextProcess() OVERRIDE;
@ -80,6 +90,8 @@ class NullReceiveStatistics : public ReceiveStatistics {
virtual void SetMaxReorderingThreshold(int max_reordering_threshold) OVERRIDE;
virtual void RegisterRtcpStatisticsCallback(RtcpStatisticsCallback* callback)
OVERRIDE;
virtual void RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) OVERRIDE;
};
} // namespace webrtc

View File

@ -26,7 +26,8 @@ StreamStatistician::~StreamStatistician() {}
StreamStatisticianImpl::StreamStatisticianImpl(
Clock* clock,
RtcpStatisticsCallback* rtcp_callback)
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback)
: clock_(clock),
stream_lock_(CriticalSectionWrapper::CreateCriticalSection()),
incoming_bitrate_(clock, NULL),
@ -43,33 +44,26 @@ StreamStatisticianImpl::StreamStatisticianImpl(
received_seq_first_(0),
received_seq_max_(0),
received_seq_wraps_(0),
first_packet_(true),
received_packet_overhead_(12),
received_byte_count_(0),
received_retransmitted_packets_(0),
received_inorder_packet_count_(0),
last_report_inorder_packets_(0),
last_report_old_packets_(0),
last_report_seq_max_(0),
last_reported_statistics_(),
rtcp_callback_(rtcp_callback) {}
rtcp_callback_(rtcp_callback),
rtp_callback_(rtp_callback) {}
void StreamStatisticianImpl::ResetStatistics() {
CriticalSectionScoped cs(stream_lock_.get());
last_report_inorder_packets_ = 0;
last_report_old_packets_ = 0;
last_report_seq_max_ = 0;
memset(&last_reported_statistics_, 0, sizeof(last_reported_statistics_));
last_reported_statistics_ = RtcpStatistics();
jitter_q4_ = 0;
cumulative_loss_ = 0;
jitter_q4_transmission_time_offset_ = 0;
received_seq_wraps_ = 0;
received_seq_max_ = 0;
received_seq_first_ = 0;
received_byte_count_ = 0;
received_retransmitted_packets_ = 0;
received_inorder_packet_count_ = 0;
first_packet_ = true;
receive_counters_ = StreamDataCounters();
}
void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
@ -79,17 +73,17 @@ void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
bool in_order = InOrderPacketInternal(header.sequenceNumber);
ssrc_ = header.ssrc;
incoming_bitrate_.Update(bytes);
received_byte_count_ += bytes;
receive_counters_.bytes +=
bytes - (header.paddingLength + header.headerLength);
receive_counters_.header_bytes += header.headerLength;
receive_counters_.padding_bytes += header.paddingLength;
++receive_counters_.packets;
if (!in_order && retransmitted) {
++receive_counters_.retransmitted_packets;
}
if (first_packet_) {
first_packet_ = false;
// This is the first received report.
if (receive_counters_.packets == 1) {
received_seq_first_ = header.sequenceNumber;
received_seq_max_ = header.sequenceNumber;
received_inorder_packet_count_ = 1;
clock_->CurrentNtp(last_receive_time_secs_, last_receive_time_frac_);
last_receive_time_ms_ = clock_->TimeInMilliseconds();
return;
}
// Count only the new packets received. That is, if packets 1, 2, 3, 5, 4, 6
@ -99,66 +93,27 @@ void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
uint32_t receive_time_secs;
uint32_t receive_time_frac;
clock_->CurrentNtp(receive_time_secs, receive_time_frac);
received_inorder_packet_count_++;
// Wrong if we use RetransmitOfOldPacket.
int32_t seq_diff = header.sequenceNumber - received_seq_max_;
if (seq_diff < 0) {
if (receive_counters_.packets > 1 &&
received_seq_max_ > header.sequenceNumber) {
// Wrap around detected.
received_seq_wraps_++;
}
// New max.
received_seq_max_ = header.sequenceNumber;
// If new time stamp and more than one in-order packet received, calculate
// new jitter statistics.
if (header.timestamp != last_received_timestamp_ &&
received_inorder_packet_count_ > 1) {
uint32_t receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP(
receive_time_secs, receive_time_frac, header.payload_type_frequency);
uint32_t last_receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP(
last_receive_time_secs_, last_receive_time_frac_,
header.payload_type_frequency);
int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
(header.timestamp - last_received_timestamp_);
time_diff_samples = abs(time_diff_samples);
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the threshold.
if (time_diff_samples < 450000) {
// Note we calculate in Q4 to avoid using float.
int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
// Extended jitter report, RFC 5450.
// Actual network jitter, excluding the source-introduced jitter.
int32_t time_diff_samples_ext =
(receive_time_rtp - last_receive_time_rtp) -
((header.timestamp +
header.extension.transmissionTimeOffset) -
(last_received_timestamp_ +
last_received_transmission_time_offset_));
time_diff_samples_ext = abs(time_diff_samples_ext);
if (time_diff_samples_ext < 450000) {
int32_t jitter_diffQ4TransmissionTimeOffset =
(time_diff_samples_ext << 4) - jitter_q4_transmission_time_offset_;
jitter_q4_transmission_time_offset_ +=
((jitter_diffQ4TransmissionTimeOffset + 8) >> 4);
}
(receive_counters_.packets - receive_counters_.retransmitted_packets) >
1) {
UpdateJitter(header, receive_time_secs, receive_time_frac);
}
last_received_timestamp_ = header.timestamp;
last_receive_time_secs_ = receive_time_secs;
last_receive_time_frac_ = receive_time_frac;
last_receive_time_ms_ = clock_->TimeInMilliseconds();
} else {
if (retransmitted) {
received_retransmitted_packets_++;
} else {
received_inorder_packet_count_++;
}
}
uint16_t packet_oh = header.headerLength + header.paddingLength;
@ -166,6 +121,55 @@ void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
// Our measured overhead. Filter from RFC 5104 4.2.1.2:
// avg_OH (new) = 15/16*avg_OH (old) + 1/16*pckt_OH,
received_packet_overhead_ = (15 * received_packet_overhead_ + packet_oh) >> 4;
rtp_callback_->DataCountersUpdated(receive_counters_, ssrc_);
}
void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
uint32_t receive_time_secs,
uint32_t receive_time_frac) {
uint32_t receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP(
receive_time_secs, receive_time_frac, header.payload_type_frequency);
uint32_t last_receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP(
last_receive_time_secs_, last_receive_time_frac_,
header.payload_type_frequency);
int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
(header.timestamp - last_received_timestamp_);
time_diff_samples = abs(time_diff_samples);
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the threshold.
if (time_diff_samples < 450000) {
// Note we calculate in Q4 to avoid using float.
int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
// Extended jitter report, RFC 5450.
// Actual network jitter, excluding the source-introduced jitter.
int32_t time_diff_samples_ext =
(receive_time_rtp - last_receive_time_rtp) -
((header.timestamp +
header.extension.transmissionTimeOffset) -
(last_received_timestamp_ +
last_received_transmission_time_offset_));
time_diff_samples_ext = abs(time_diff_samples_ext);
if (time_diff_samples_ext < 450000) {
int32_t jitter_diffQ4TransmissionTimeOffset =
(time_diff_samples_ext << 4) - jitter_q4_transmission_time_offset_;
jitter_q4_transmission_time_offset_ +=
((jitter_diffQ4TransmissionTimeOffset + 8) >> 4);
}
}
void StreamStatisticianImpl::FecPacketReceived() {
CriticalSectionScoped cs(stream_lock_.get());
++receive_counters_.fec_packets;
rtp_callback_->DataCountersUpdated(receive_counters_, ssrc_);
}
void StreamStatisticianImpl::SetMaxReorderingThreshold(
@ -178,7 +182,7 @@ bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
bool reset) {
{
CriticalSectionScoped cs(stream_lock_.get());
if (received_seq_first_ == 0 && received_byte_count_ == 0) {
if (received_seq_first_ == 0 && receive_counters_.bytes == 0) {
// We have not received anything.
return false;
}
@ -197,6 +201,8 @@ bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
}
rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
rtp_callback_->DataCountersUpdated(receive_counters_, ssrc_);
return true;
}
@ -219,7 +225,8 @@ RtcpStatistics StreamStatisticianImpl::CalculateStatistics() {
// Number of received RTP packets since last report, counts all packets but
// not re-transmissions.
uint32_t rec_since_last =
received_inorder_packet_count_ - last_report_inorder_packets_;
(receive_counters_.packets - receive_counters_.retransmitted_packets) -
last_report_inorder_packets_;
// With NACK we don't know the expected retransmissions during the last
// second. We know how many "old" packets we have received. We just count
@ -231,7 +238,7 @@ RtcpStatistics StreamStatisticianImpl::CalculateStatistics() {
// re-transmitted. We use RTT to decide if a packet is re-ordered or
// re-transmitted.
uint32_t retransmitted_packets =
received_retransmitted_packets_ - last_report_old_packets_;
receive_counters_.retransmitted_packets - last_report_old_packets_;
rec_since_last += retransmitted_packets;
int32_t missing = 0;
@ -258,8 +265,9 @@ RtcpStatistics StreamStatisticianImpl::CalculateStatistics() {
last_reported_statistics_ = stats;
// Only for report blocks in RTCP SR and RR.
last_report_inorder_packets_ = received_inorder_packet_count_;
last_report_old_packets_ = received_retransmitted_packets_;
last_report_inorder_packets_ =
receive_counters_.packets - receive_counters_.retransmitted_packets;
last_report_old_packets_ = receive_counters_.retransmitted_packets;
last_report_seq_max_ = received_seq_max_;
return stats;
@ -269,11 +277,11 @@ void StreamStatisticianImpl::GetDataCounters(
uint32_t* bytes_received, uint32_t* packets_received) const {
CriticalSectionScoped cs(stream_lock_.get());
if (bytes_received) {
*bytes_received = received_byte_count_;
*bytes_received = receive_counters_.bytes + receive_counters_.header_bytes +
receive_counters_.padding_bytes;
}
if (packets_received) {
*packets_received =
received_retransmitted_packets_ + received_inorder_packet_count_;
*packets_received = receive_counters_.packets;
}
}
@ -358,7 +366,8 @@ ReceiveStatisticsImpl::ReceiveStatisticsImpl(Clock* clock)
: clock_(clock),
receive_statistics_lock_(CriticalSectionWrapper::CreateCriticalSection()),
last_rate_update_ms_(0),
rtcp_stats_callback_(NULL) {}
rtcp_stats_callback_(NULL),
rtp_stats_callback_(NULL) {}
ReceiveStatisticsImpl::~ReceiveStatisticsImpl() {
while (!statisticians_.empty()) {
@ -368,7 +377,8 @@ ReceiveStatisticsImpl::~ReceiveStatisticsImpl() {
}
void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header,
size_t bytes, bool old_packet) {
size_t bytes,
bool retransmitted) {
StatisticianImplMap::iterator it;
{
CriticalSectionScoped cs(receive_statistics_lock_.get());
@ -376,11 +386,18 @@ void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header,
if (it == statisticians_.end()) {
std::pair<StatisticianImplMap::iterator, uint32_t> insert_result =
statisticians_.insert(std::make_pair(
header.ssrc, new StreamStatisticianImpl(clock_, this)));
header.ssrc, new StreamStatisticianImpl(clock_, this, this)));
it = insert_result.first;
}
}
it->second->IncomingPacket(header, bytes, old_packet);
it->second->IncomingPacket(header, bytes, retransmitted);
}
void ReceiveStatisticsImpl::FecPacketReceived(uint32_t ssrc) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
StatisticianImplMap::iterator it = statisticians_.find(ssrc);
assert(it != statisticians_.end());
it->second->FecPacketReceived();
}
void ReceiveStatisticsImpl::ChangeSsrc(uint32_t from_ssrc, uint32_t to_ssrc) {
@ -461,10 +478,28 @@ void ReceiveStatisticsImpl::StatisticsUpdated(const RtcpStatistics& statistics,
}
}
void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
if (callback != NULL)
assert(rtp_stats_callback_ == NULL);
rtp_stats_callback_ = callback;
}
void ReceiveStatisticsImpl::DataCountersUpdated(const StreamDataCounters& stats,
uint32_t ssrc) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(stats, ssrc);
}
}
void NullReceiveStatistics::IncomingPacket(const RTPHeader& rtp_header,
size_t bytes,
bool retransmitted) {}
void NullReceiveStatistics::FecPacketReceived(uint32_t ssrc) {}
StatisticianMap NullReceiveStatistics::GetActiveStatisticians() const {
return StatisticianMap();
}
@ -484,4 +519,7 @@ int32_t NullReceiveStatistics::Process() { return 0; }
void NullReceiveStatistics::RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {}
void NullReceiveStatistics::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {}
} // namespace webrtc

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@ -25,7 +25,9 @@ class CriticalSectionWrapper;
class StreamStatisticianImpl : public StreamStatistician {
public:
StreamStatisticianImpl(Clock* clock, RtcpStatisticsCallback* rtcp_callback);
StreamStatisticianImpl(Clock* clock,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback);
virtual ~StreamStatisticianImpl() {}
virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) OVERRIDE;
@ -34,11 +36,13 @@ class StreamStatisticianImpl : public StreamStatistician {
virtual uint32_t BitrateReceived() const OVERRIDE;
virtual void ResetStatistics() OVERRIDE;
virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
int min_rtt) const OVERRIDE;
int min_rtt) const OVERRIDE;
virtual bool IsPacketInOrder(uint16_t sequence_number) const OVERRIDE;
void IncomingPacket(const RTPHeader& rtp_header, size_t bytes,
void IncomingPacket(const RTPHeader& rtp_header,
size_t bytes,
bool retransmitted);
void FecPacketReceived();
void SetMaxReorderingThreshold(int max_reordering_threshold);
void ProcessBitrate();
virtual void LastReceiveTimeNtp(uint32_t* secs, uint32_t* frac) const;
@ -46,6 +50,9 @@ class StreamStatisticianImpl : public StreamStatistician {
private:
bool InOrderPacketInternal(uint16_t sequence_number) const;
RtcpStatistics CalculateStatistics();
void UpdateJitter(const RTPHeader& header,
uint32_t receive_time_secs,
uint32_t receive_time_frac);
Clock* clock_;
scoped_ptr<CriticalSectionWrapper> stream_lock_;
@ -66,13 +73,10 @@ class StreamStatisticianImpl : public StreamStatistician {
uint16_t received_seq_first_;
uint16_t received_seq_max_;
uint16_t received_seq_wraps_;
bool first_packet_;
// Current counter values.
uint16_t received_packet_overhead_;
uint32_t received_byte_count_;
uint32_t received_retransmitted_packets_;
uint32_t received_inorder_packet_count_;
StreamDataCounters receive_counters_;
// Counter values when we sent the last report.
uint32_t last_report_inorder_packets_;
@ -81,18 +85,22 @@ class StreamStatisticianImpl : public StreamStatistician {
RtcpStatistics last_reported_statistics_;
RtcpStatisticsCallback* const rtcp_callback_;
StreamDataCountersCallback* const rtp_callback_;
};
class ReceiveStatisticsImpl : public ReceiveStatistics,
public RtcpStatisticsCallback {
public RtcpStatisticsCallback,
public StreamDataCountersCallback {
public:
explicit ReceiveStatisticsImpl(Clock* clock);
~ReceiveStatisticsImpl();
// Implement ReceiveStatistics.
virtual void IncomingPacket(const RTPHeader& header, size_t bytes,
bool old_packet) OVERRIDE;
virtual void IncomingPacket(const RTPHeader& header,
size_t bytes,
bool retransmitted) OVERRIDE;
virtual void FecPacketReceived(uint32_t ssrc) OVERRIDE;
virtual StatisticianMap GetActiveStatisticians() const OVERRIDE;
virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE;
virtual void SetMaxReorderingThreshold(int max_reordering_threshold) OVERRIDE;
@ -106,10 +114,15 @@ class ReceiveStatisticsImpl : public ReceiveStatistics,
virtual void RegisterRtcpStatisticsCallback(RtcpStatisticsCallback* callback)
OVERRIDE;
virtual void StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) OVERRIDE;
virtual void RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) OVERRIDE;
private:
virtual void StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) OVERRIDE;
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) OVERRIDE;
typedef std::map<uint32_t, StreamStatisticianImpl*> StatisticianImplMap;
Clock* clock_;
@ -118,6 +131,7 @@ class ReceiveStatisticsImpl : public ReceiveStatistics,
StatisticianImplMap statisticians_;
RtcpStatisticsCallback* rtcp_stats_callback_;
StreamDataCountersCallback* rtp_stats_callback_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_

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@ -28,10 +28,10 @@ class ReceiveStatisticsTest : public ::testing::Test {
receive_statistics_(ReceiveStatistics::Create(&clock_)) {
memset(&header1_, 0, sizeof(header1_));
header1_.ssrc = kSsrc1;
header1_.sequenceNumber = 0;
header1_.sequenceNumber = 100;
memset(&header2_, 0, sizeof(header2_));
header2_.ssrc = kSsrc2;
header2_.sequenceNumber = 0;
header2_.sequenceNumber = 100;
}
protected:
@ -132,7 +132,7 @@ TEST_F(ReceiveStatisticsTest, ActiveStatisticians) {
EXPECT_EQ(2u, packets_received);
}
TEST_F(ReceiveStatisticsTest, Callbacks) {
TEST_F(ReceiveStatisticsTest, RtcpCallbacks) {
class TestCallback : public RtcpStatisticsCallback {
public:
TestCallback()
@ -186,6 +186,10 @@ TEST_F(ReceiveStatisticsTest, Callbacks) {
callback.stats_.extended_max_sequence_number);
EXPECT_EQ(statistics.fraction_lost, callback.stats_.fraction_lost);
EXPECT_EQ(statistics.jitter, callback.stats_.jitter);
EXPECT_EQ(51, statistics.fraction_lost);
EXPECT_EQ(1u, statistics.cumulative_lost);
EXPECT_EQ(5u, statistics.extended_max_sequence_number);
EXPECT_EQ(4u, statistics.jitter);
receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
@ -214,4 +218,86 @@ TEST_F(ReceiveStatisticsTest, Callbacks) {
// Should not have been called after deregister.
EXPECT_EQ(1u, callback.num_calls_);
}
TEST_F(ReceiveStatisticsTest, RtpCallbacks) {
class TestCallback : public StreamDataCountersCallback {
public:
TestCallback()
: StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {}
virtual ~TestCallback() {}
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) {
ssrc_ = ssrc;
stats_ = counters;
++num_calls_;
}
void ExpectMatches(uint32_t num_calls,
uint32_t ssrc,
uint32_t bytes,
uint32_t padding,
uint32_t packets,
uint32_t retransmits,
uint32_t fec) {
EXPECT_EQ(num_calls, num_calls_);
EXPECT_EQ(ssrc, ssrc_);
EXPECT_EQ(bytes, stats_.bytes);
EXPECT_EQ(padding, stats_.padding_bytes);
EXPECT_EQ(packets, stats_.packets);
EXPECT_EQ(retransmits, stats_.retransmitted_packets);
EXPECT_EQ(fec, stats_.fec_packets);
}
uint32_t num_calls_;
uint32_t ssrc_;
StreamDataCounters stats_;
} callback;
receive_statistics_->RegisterRtpStatisticsCallback(&callback);
const uint32_t kHeaderLength = 20;
const uint32_t kPaddingLength = 9;
// One packet of size kPacketSize1.
header1_.headerLength = kHeaderLength;
receive_statistics_->IncomingPacket(
header1_, kPacketSize1 + kHeaderLength, false);
callback.ExpectMatches(1, kSsrc1, kPacketSize1, 0, 1, 0, 0);
++header1_.sequenceNumber;
clock_.AdvanceTimeMilliseconds(5);
header1_.paddingLength = 9;
// Another packet of size kPacketSize1 with 9 bytes padding.
receive_statistics_->IncomingPacket(
header1_, kPacketSize1 + kHeaderLength + kPaddingLength, false);
callback.ExpectMatches(2, kSsrc1, 2 * kPacketSize1, kPaddingLength, 2, 0, 0);
clock_.AdvanceTimeMilliseconds(5);
// Retransmit last packet.
receive_statistics_->IncomingPacket(
header1_, kPacketSize1 + kHeaderLength + kPaddingLength, true);
callback.ExpectMatches(
3, kSsrc1, 3 * kPacketSize1, kPaddingLength * 2, 3, 1, 0);
header1_.paddingLength = 0;
++header1_.sequenceNumber;
clock_.AdvanceTimeMilliseconds(5);
// One recovered packet.
receive_statistics_->IncomingPacket(
header1_, kPacketSize1 + kHeaderLength, false);
receive_statistics_->FecPacketReceived(kSsrc1);
callback.ExpectMatches(
5, kSsrc1, 4 * kPacketSize1, kPaddingLength * 2, 4, 1, 1);
receive_statistics_->RegisterRtpStatisticsCallback(NULL);
// New stats, but callback should not be called.
++header1_.sequenceNumber;
clock_.AdvanceTimeMilliseconds(5);
receive_statistics_->IncomingPacket(
header1_, kPacketSize1 + kHeaderLength, true);
callback.ExpectMatches(
5, kSsrc1, 4 * kPacketSize1, kPaddingLength * 2, 4, 1, 1);
}
} // namespace webrtc

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@ -1372,6 +1372,16 @@ void ViEChannel::RegisterSendChannelRtpStatisticsCallback(
}
}
void ViEChannel::RegisterReceiveChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
WEBRTC_TRACE(kTraceInfo,
kTraceVideo,
ViEId(engine_id_, channel_id_),
"%s",
__FUNCTION__);
vie_receiver_.GetReceiveStatistics()->RegisterRtpStatisticsCallback(callback);
}
void ViEChannel::GetBandwidthUsage(uint32_t* total_bitrate_sent,
uint32_t* video_bitrate_sent,
uint32_t* fec_bitrate_sent,

View File

@ -199,6 +199,10 @@ class ViEChannel
void RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback);
// Called on update of RTP statistics.
void RegisterReceiveChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback);
void GetBandwidthUsage(uint32_t* total_bitrate_sent,
uint32_t* video_bitrate_sent,
uint32_t* fec_bitrate_sent,

View File

@ -269,11 +269,13 @@ int ViEReceiver::InsertRTPPacket(const int8_t* rtp_packet,
header.payload_type_frequency = kVideoPayloadTypeFrequency;
bool in_order = IsPacketInOrder(header);
rtp_receive_statistics_->IncomingPacket(header, received_packet_length,
IsPacketRetransmitted(header, in_order));
rtp_receive_statistics_->IncomingPacket(
header, received_packet_length, IsPacketRetransmitted(header, in_order));
rtp_payload_registry_->SetIncomingPayloadType(header);
return ReceivePacket(received_packet, received_packet_length, header,
in_order) ? 0 : -1;
return ReceivePacket(
received_packet, received_packet_length, header, in_order)
? 0
: -1;
}
bool ViEReceiver::ReceivePacket(const uint8_t* packet,
@ -299,9 +301,11 @@ bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
int packet_length,
const RTPHeader& header) {
if (rtp_payload_registry_->IsRed(header)) {
int8_t ulpfec_pt = rtp_payload_registry_->ulpfec_payload_type();
if (packet[header.headerLength] == ulpfec_pt)
rtp_receive_statistics_->FecPacketReceived(header.ssrc);
if (fec_receiver_->AddReceivedRedPacket(
header, packet, packet_length,
rtp_payload_registry_->ulpfec_payload_type()) != 0) {
header, packet, packet_length, ulpfec_pt) != 0) {
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_,
"Incoming RED packet error");
return false;

View File

@ -90,8 +90,10 @@ class ViEReceiver : public RtpData {
private:
int InsertRTPPacket(const int8_t* rtp_packet, int rtp_packet_length,
const PacketTime& packet_time);
bool ReceivePacket(const uint8_t* packet, int packet_length,
const RTPHeader& header, bool in_order);
bool ReceivePacket(const uint8_t* packet,
int packet_length,
const RTPHeader& header,
bool in_order);
// Parses and handles for instance RTX and RED headers.
// This function assumes that it's being called from only one thread.
bool ParseAndHandleEncapsulatingHeader(const uint8_t* packet,

View File

@ -1224,15 +1224,35 @@ int ViERTP_RTCPImpl::DeregisterSendChannelRtpStatisticsCallback(
}
int ViERTP_RTCPImpl::RegisterReceiveChannelRtpStatisticsCallback(
int channel, StreamDataCountersCallback* callback) {
// TODO(sprang): Implement
return -1;
const 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->RegisterReceiveChannelRtpStatisticsCallback(callback);
return 0;
}
int ViERTP_RTCPImpl::DeregisterReceiveChannelRtpStatisticsCallback(
int channel, StreamDataCountersCallback* callback) {
// TODO(sprang): Implement
return -1;
const 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->RegisterReceiveChannelRtpStatisticsCallback(NULL);
return 0;
}
// Called whenever the send bitrate is updated.