Add decoder-timing stats to VideoReceiveStream.
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667, 1788
Review URL: https://webrtc-codereview.appspot.com/40819004
Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index cbd6eea..f2643c3 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -1766,21 +1766,20 @@
info.send_frame_width = 0;
info.send_frame_height = 0;
- for (std::map<uint32_t, webrtc::SsrcStats>::iterator it =
+ for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
stats.substreams.begin();
- it != stats.substreams.end();
- ++it) {
+ it != stats.substreams.end(); ++it) {
// TODO(pbos): Wire up additional stats, such as padding bytes.
- webrtc::SsrcStats stream_stats = it->second;
+ webrtc::VideoSendStream::StreamStats stream_stats = it->second;
info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
stream_stats.rtp_stats.transmitted.header_bytes +
stream_stats.rtp_stats.transmitted.padding_bytes;
info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
info.packets_lost += stream_stats.rtcp_stats.cumulative_lost;
- if (stream_stats.sent_width > info.send_frame_width)
- info.send_frame_width = stream_stats.sent_width;
- if (stream_stats.sent_height > info.send_frame_height)
- info.send_frame_height = stream_stats.sent_height;
+ if (stream_stats.width > info.send_frame_width)
+ info.send_frame_width = stream_stats.width;
+ if (stream_stats.height > info.send_frame_height)
+ info.send_frame_height = stream_stats.height;
info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets;
info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets;
info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets;
@@ -1788,7 +1787,8 @@
if (!stats.substreams.empty()) {
// TODO(pbos): Report fraction lost per SSRC.
- webrtc::SsrcStats first_stream_stats = stats.substreams.begin()->second;
+ webrtc::VideoSendStream::StreamStats first_stream_stats =
+ stats.substreams.begin()->second;
info.fraction_lost =
static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
(1 << 8);
@@ -1804,10 +1804,9 @@
return;
}
webrtc::VideoSendStream::Stats stats = stream_->GetStats();
- for (std::map<uint32_t, webrtc::SsrcStats>::iterator it =
+ for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
stats.substreams.begin();
- it != stats.substreams.end();
- ++it) {
+ it != stats.substreams.end(); ++it) {
bwe_info->transmit_bitrate += it->second.total_bitrate_bps;
bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps;
}
@@ -2043,6 +2042,14 @@
info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_;
}
+ info.decode_ms = stats.decode_ms;
+ info.max_decode_ms = stats.max_decode_ms;
+ info.current_delay_ms = stats.current_delay_ms;
+ info.target_delay_ms = stats.target_delay_ms;
+ info.jitter_buffer_ms = stats.jitter_buffer_ms;
+ info.min_playout_delay_ms = stats.min_playout_delay_ms;
+ info.render_delay_ms = stats.render_delay_ms;
+
info.firs_sent = stats.rtcp_packet_type_counts.fir_packets;
info.plis_sent = stats.rtcp_packet_type_counts.pli_packets;
info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets;
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index deb0e29..b4f58e2 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -2041,12 +2041,12 @@
TEST_F(WebRtcVideoChannel2Test, GetStatsReportsUpperResolution) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
- stats.substreams[17].sent_width = 123;
- stats.substreams[17].sent_height = 40;
- stats.substreams[42].sent_width = 80;
- stats.substreams[42].sent_height = 31;
- stats.substreams[11].sent_width = 20;
- stats.substreams[11].sent_height = 90;
+ stats.substreams[17].width = 123;
+ stats.substreams[17].height = 40;
+ stats.substreams[42].width = 80;
+ stats.substreams[42].height = 31;
+ stats.substreams[11].width = 20;
+ stats.substreams[11].height = 90;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
@@ -2096,6 +2096,29 @@
info.receivers[0].plis_sent);
}
+TEST_F(WebRtcVideoChannel2Test, GetStatsTranslatesDecodeStatsCorrectly) {
+ FakeVideoReceiveStream* stream = AddRecvStream();
+ webrtc::VideoReceiveStream::Stats stats;
+ stats.decode_ms = 2;
+ stats.max_decode_ms = 3;
+ stats.current_delay_ms = 4;
+ stats.target_delay_ms = 5;
+ stats.jitter_buffer_ms = 6;
+ stats.min_playout_delay_ms = 7;
+ stats.render_delay_ms = 8;
+ stream->SetStats(stats);
+
+ cricket::VideoMediaInfo info;
+ ASSERT_TRUE(channel_->GetStats(cricket::StatsOptions(), &info));
+ EXPECT_EQ(stats.decode_ms, info.receivers[0].decode_ms);
+ EXPECT_EQ(stats.max_decode_ms, info.receivers[0].max_decode_ms);
+ EXPECT_EQ(stats.current_delay_ms, info.receivers[0].current_delay_ms);
+ EXPECT_EQ(stats.target_delay_ms, info.receivers[0].target_delay_ms);
+ EXPECT_EQ(stats.jitter_buffer_ms, info.receivers[0].jitter_buffer_ms);
+ EXPECT_EQ(stats.min_playout_delay_ms, info.receivers[0].min_playout_delay_ms);
+ EXPECT_EQ(stats.render_delay_ms, info.receivers[0].render_delay_ms);
+}
+
TEST_F(WebRtcVideoChannel2Test, TranslatesCallStatsCorrectly) {
AddSendStream();
AddSendStream();
diff --git a/webrtc/config.h b/webrtc/config.h
index c699170..9f2a7a3 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -21,27 +21,6 @@
namespace webrtc {
-struct SsrcStats {
- SsrcStats()
- : sent_width(0),
- sent_height(0),
- total_bitrate_bps(0),
- retransmit_bitrate_bps(0),
- avg_delay_ms(0),
- max_delay_ms(0) {}
- FrameCounts frame_counts;
- int sent_width;
- int sent_height;
- // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
- int total_bitrate_bps;
- int retransmit_bitrate_bps;
- int avg_delay_ms;
- int max_delay_ms;
- StreamDataCounters rtp_stats;
- RtcpPacketTypeCounter rtcp_packet_type_counts;
- RtcpStatistics rtcp_stats;
-};
-
// Settings for NACK, see RFC 4585 for details.
struct NackConfig {
NackConfig() : rtp_history_ms(0) {}
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 7b349a9..66cd74f 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1630,7 +1630,7 @@
stats.rtp_stats.retransmitted.packets != 0;
receive_stats_filled_["CodecStats"] |=
- stats.avg_delay_ms != 0 || stats.discarded_packets != 0;
+ stats.target_delay_ms != 0 || stats.discarded_packets != 0;
receive_stats_filled_["FrameCounts"] |=
stats.frame_counts.key_frames != 0 ||
@@ -1656,17 +1656,16 @@
send_stats_filled_["NumStreams"] |=
stats.substreams.size() == expected_send_ssrcs_.size();
- for (std::map<uint32_t, SsrcStats>::const_iterator it =
+ for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
stats.substreams.begin();
- it != stats.substreams.end();
- ++it) {
+ it != stats.substreams.end(); ++it) {
EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
expected_send_ssrcs_.end());
send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
stats.input_frame_rate != 0;
- const SsrcStats& stream_stats = it->second;
+ const VideoSendStream::StreamStats& stream_stats = it->second;
send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
stream_stats.rtcp_stats.cumulative_lost != 0 ||
diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc
index 444f7c1..6e9505c 100644
--- a/webrtc/video/receive_statistics_proxy.cc
+++ b/webrtc/video/receive_statistics_proxy.cc
@@ -47,7 +47,13 @@
int min_playout_delay_ms,
int render_delay_ms) {
CriticalSectionScoped lock(crit_.get());
- stats_.avg_delay_ms = target_delay_ms;
+ stats_.decode_ms = decode_ms;
+ stats_.max_decode_ms = max_decode_ms;
+ stats_.current_delay_ms = current_delay_ms;
+ stats_.target_delay_ms = target_delay_ms;
+ stats_.jitter_buffer_ms = jitter_buffer_ms;
+ stats_.min_playout_delay_ms = min_playout_delay_ms;
+ stats_.render_delay_ms = render_delay_ms;
}
void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
diff --git a/webrtc/video/send_statistics_proxy.cc b/webrtc/video/send_statistics_proxy.cc
index 00a1f0d..64126d2 100644
--- a/webrtc/video/send_statistics_proxy.cc
+++ b/webrtc/video/send_statistics_proxy.cc
@@ -57,20 +57,23 @@
void SendStatisticsProxy::PurgeOldStats() {
int64_t current_time_ms = clock_->TimeInMilliseconds();
- for (std::map<uint32_t, SsrcStats>::iterator it = stats_.substreams.begin();
+ for (std::map<uint32_t, VideoSendStream::StreamStats>::iterator it =
+ stats_.substreams.begin();
it != stats_.substreams.end(); ++it) {
uint32_t ssrc = it->first;
if (update_times_[ssrc].resolution_update_ms + kStatsTimeoutMs >
current_time_ms)
continue;
- it->second.sent_width = 0;
- it->second.sent_height = 0;
+ it->second.width = 0;
+ it->second.height = 0;
}
}
-SsrcStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
- std::map<uint32_t, SsrcStats>::iterator it = stats_.substreams.find(ssrc);
+VideoSendStream::StreamStats* SendStatisticsProxy::GetStatsEntry(
+ uint32_t ssrc) {
+ std::map<uint32_t, VideoSendStream::StreamStats>::iterator it =
+ stats_.substreams.find(ssrc);
if (it != stats_.substreams.end())
return &it->second;
@@ -98,12 +101,12 @@
uint32_t ssrc = config_.rtp.ssrcs[simulcast_idx];
CriticalSectionScoped lock(crit_.get());
- SsrcStats* stats = GetStatsEntry(ssrc);
+ VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
- stats->sent_width = encoded_image._encodedWidth;
- stats->sent_height = encoded_image._encodedHeight;
+ stats->width = encoded_image._encodedWidth;
+ stats->height = encoded_image._encodedHeight;
update_times_[ssrc].resolution_update_ms = clock_->TimeInMilliseconds();
}
@@ -111,7 +114,7 @@
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) {
CriticalSectionScoped lock(crit_.get());
- SsrcStats* stats = GetStatsEntry(ssrc);
+ VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
@@ -121,7 +124,7 @@
void SendStatisticsProxy::StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) {
CriticalSectionScoped lock(crit_.get());
- SsrcStats* stats = GetStatsEntry(ssrc);
+ VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
@@ -135,11 +138,9 @@
const StreamDataCounters& counters,
uint32_t ssrc) {
CriticalSectionScoped lock(crit_.get());
- SsrcStats* stats = GetStatsEntry(ssrc);
+ VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
DCHECK(stats != NULL) << "DataCountersUpdated reported for unknown ssrc: "
<< ssrc;
- if (stats == NULL)
- return;
stats->rtp_stats = counters;
}
@@ -148,7 +149,7 @@
const BitrateStatistics& retransmit_stats,
uint32_t ssrc) {
CriticalSectionScoped lock(crit_.get());
- SsrcStats* stats = GetStatsEntry(ssrc);
+ VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
@@ -159,7 +160,7 @@
void SendStatisticsProxy::FrameCountUpdated(const FrameCounts& frame_counts,
uint32_t ssrc) {
CriticalSectionScoped lock(crit_.get());
- SsrcStats* stats = GetStatsEntry(ssrc);
+ VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
@@ -170,7 +171,7 @@
int max_delay_ms,
uint32_t ssrc) {
CriticalSectionScoped lock(crit_.get());
- SsrcStats* stats = GetStatsEntry(ssrc);
+ VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
stats->avg_delay_ms = avg_delay_ms;
diff --git a/webrtc/video/send_statistics_proxy.h b/webrtc/video/send_statistics_proxy.h
index 568ffc4..5d4edcd 100644
--- a/webrtc/video/send_statistics_proxy.h
+++ b/webrtc/video/send_statistics_proxy.h
@@ -94,7 +94,8 @@
int64_t resolution_update_ms;
};
void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
- SsrcStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+ VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_);
Clock* const clock_;
const VideoSendStream::Config config_;
diff --git a/webrtc/video/send_statistics_proxy_unittest.cc b/webrtc/video/send_statistics_proxy_unittest.cc
index 2a154f3..ffbdc6e 100644
--- a/webrtc/video/send_statistics_proxy_unittest.cc
+++ b/webrtc/video/send_statistics_proxy_unittest.cc
@@ -49,15 +49,14 @@
EXPECT_EQ(one.suspended, other.suspended);
EXPECT_EQ(one.substreams.size(), other.substreams.size());
- for (std::map<uint32_t, SsrcStats>::const_iterator it =
+ for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
one.substreams.begin();
- it != one.substreams.end();
- ++it) {
- std::map<uint32_t, SsrcStats>::const_iterator corresponding_it =
- other.substreams.find(it->first);
+ it != one.substreams.end(); ++it) {
+ std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator
+ corresponding_it = other.substreams.find(it->first);
ASSERT_TRUE(corresponding_it != other.substreams.end());
- const SsrcStats& a = it->second;
- const SsrcStats& b = corresponding_it->second;
+ const VideoSendStream::StreamStats& a = it->second;
+ const VideoSendStream::StreamStats& b = corresponding_it->second;
EXPECT_EQ(a.frame_counts.key_frames, b.frame_counts.key_frames);
EXPECT_EQ(a.frame_counts.delta_frames, b.frame_counts.delta_frames);
@@ -91,7 +90,8 @@
int avg_delay_ms_;
int max_delay_ms_;
VideoSendStream::Stats expected_;
- typedef std::map<uint32_t, SsrcStats>::const_iterator StreamIterator;
+ typedef std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator
+ StreamIterator;
};
TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
@@ -100,7 +100,7 @@
it != config_.rtp.ssrcs.end();
++it) {
const uint32_t ssrc = *it;
- SsrcStats& ssrc_stats = expected_.substreams[ssrc];
+ VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc];
// Add statistics with some arbitrary, but unique, numbers.
uint32_t offset = ssrc * sizeof(RtcpStatistics);
@@ -114,7 +114,7 @@
it != config_.rtp.rtx.ssrcs.end();
++it) {
const uint32_t ssrc = *it;
- SsrcStats& ssrc_stats = expected_.substreams[ssrc];
+ VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc];
// Add statistics with some arbitrary, but unique, numbers.
uint32_t offset = ssrc * sizeof(RtcpStatistics);
@@ -171,8 +171,8 @@
++it) {
const uint32_t ssrc = *it;
// Add statistics with some arbitrary, but unique, numbers.
- SsrcStats& stats = expected_.substreams[ssrc];
- uint32_t offset = ssrc * sizeof(SsrcStats);
+ VideoSendStream::StreamStats& stats = expected_.substreams[ssrc];
+ uint32_t offset = ssrc * sizeof(VideoSendStream::StreamStats);
FrameCounts frame_counts;
frame_counts.key_frames = offset;
frame_counts.delta_frames = offset + 1;
@@ -184,8 +184,8 @@
++it) {
const uint32_t ssrc = *it;
// Add statistics with some arbitrary, but unique, numbers.
- SsrcStats& stats = expected_.substreams[ssrc];
- uint32_t offset = ssrc * sizeof(SsrcStats);
+ VideoSendStream::StreamStats& stats = expected_.substreams[ssrc];
+ uint32_t offset = ssrc * sizeof(VideoSendStream::StreamStats);
FrameCounts frame_counts;
frame_counts.key_frames = offset;
frame_counts.delta_frames = offset + 1;
@@ -342,16 +342,16 @@
statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
VideoSendStream::Stats stats = statistics_proxy_->GetStats();
- EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].sent_width);
- EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[0]].sent_height);
- EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[1]].sent_width);
- EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[1]].sent_height);
+ EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].width);
+ EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[0]].height);
+ EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[1]].width);
+ EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[1]].height);
// Forward almost to timeout, this should not have removed stats.
fake_clock_.AdvanceTimeMilliseconds(SendStatisticsProxy::kStatsTimeoutMs - 1);
stats = statistics_proxy_->GetStats();
- EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].sent_width);
- EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[0]].sent_height);
+ EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].width);
+ EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[0]].height);
// Update the first SSRC with bogus RTCP stats to make sure that encoded
// resolution still times out (no global timeout for all stats).
@@ -368,10 +368,10 @@
// reported, but substream 1 should.
fake_clock_.AdvanceTimeMilliseconds(1);
stats = statistics_proxy_->GetStats();
- EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[0]].sent_width);
- EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[0]].sent_height);
- EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[1]].sent_width);
- EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[1]].sent_height);
+ EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[0]].width);
+ EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[0]].height);
+ EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[1]].width);
+ EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[1]].height);
}
} // namespace webrtc
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index f5ec59b..e9ae9a7 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1673,9 +1673,10 @@
stats.substreams.end())
<< "No stats for SSRC: " << kSendSsrcs[i]
<< ", stats should exist as soon as frames have been encoded.";
- SsrcStats ssrc_stats = stats.substreams[kSendSsrcs[i]];
- EXPECT_EQ(kEncodedResolution[i].width, ssrc_stats.sent_width);
- EXPECT_EQ(kEncodedResolution[i].height, ssrc_stats.sent_height);
+ VideoSendStream::StreamStats ssrc_stats =
+ stats.substreams[kSendSsrcs[i]];
+ EXPECT_EQ(kEncodedResolution[i].width, ssrc_stats.width);
+ EXPECT_EQ(kEncodedResolution[i].height, ssrc_stats.height);
}
}
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index 5722c33..0f2151d 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -64,22 +64,29 @@
int expected_delay_ms;
};
- struct Stats : public SsrcStats {
- Stats()
- : network_frame_rate(0),
- decode_frame_rate(0),
- render_frame_rate(0),
- avg_delay_ms(0),
- discarded_packets(0),
- ssrc(0) {}
+ struct Stats {
+ int network_frame_rate = 0;
+ int decode_frame_rate = 0;
+ int render_frame_rate = 0;
- int network_frame_rate;
- int decode_frame_rate;
- int render_frame_rate;
- int avg_delay_ms;
- int discarded_packets;
- uint32_t ssrc;
+ // Decoder stats.
+ FrameCounts frame_counts;
+ int decode_ms = 0;
+ int max_decode_ms = 0;
+ int current_delay_ms = 0;
+ int target_delay_ms = 0;
+ int jitter_buffer_ms = 0;
+ int min_playout_delay_ms = 0;
+ int render_delay_ms = 0;
+
+ int total_bitrate_bps = 0;
+ int discarded_packets = 0;
+
+ uint32_t ssrc = 0;
std::string c_name;
+ StreamDataCounters rtp_stats;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
+ RtcpStatistics rtcp_stats;
};
struct Config {
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index 7483ceb..0e41cc5 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -37,6 +37,20 @@
class VideoSendStream {
public:
+ struct StreamStats {
+ FrameCounts frame_counts;
+ int width = 0;
+ int height = 0;
+ // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
+ int total_bitrate_bps = 0;
+ int retransmit_bitrate_bps = 0;
+ int avg_delay_ms = 0;
+ int max_delay_ms = 0;
+ StreamDataCounters rtp_stats;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
+ RtcpStatistics rtcp_stats;
+ };
+
struct Stats {
Stats()
: input_frame_rate(0),
@@ -47,7 +61,7 @@
int encode_frame_rate;
int media_bitrate_bps;
bool suspended;
- std::map<uint32_t, SsrcStats> substreams;
+ std::map<uint32_t, StreamStats> substreams;
};
struct Config {
