| Index: webrtc/tools/event_log_visualizer/analyzer.cc
|
| diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
|
| index 05d94ee0633dc316ff3f7ede094b5c1e62a7314e..ec569990224d2a9d7e951e3e06d24793e98c06e3 100644
|
| --- a/webrtc/tools/event_log_visualizer/analyzer.cc
|
| +++ b/webrtc/tools/event_log_visualizer/analyzer.cc
|
| @@ -76,43 +76,6 @@ int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
|
| return difference;
|
| }
|
|
|
| -class StreamId {
|
| - public:
|
| - StreamId(uint32_t ssrc,
|
| - webrtc::PacketDirection direction,
|
| - webrtc::MediaType media_type)
|
| - : ssrc_(ssrc), direction_(direction), media_type_(media_type) {}
|
| -
|
| - bool operator<(const StreamId& other) const {
|
| - if (ssrc_ < other.ssrc_) {
|
| - return true;
|
| - }
|
| - if (ssrc_ == other.ssrc_) {
|
| - if (media_type_ < other.media_type_) {
|
| - return true;
|
| - }
|
| - if (media_type_ == other.media_type_) {
|
| - if (direction_ < other.direction_) {
|
| - return true;
|
| - }
|
| - }
|
| - }
|
| - return false;
|
| - }
|
| -
|
| - bool operator==(const StreamId& other) const {
|
| - return ssrc_ == other.ssrc_ && direction_ == other.direction_ &&
|
| - media_type_ == other.media_type_;
|
| - }
|
| -
|
| - uint32_t GetSsrc() const { return ssrc_; }
|
| -
|
| - private:
|
| - uint32_t ssrc_;
|
| - webrtc::PacketDirection direction_;
|
| - webrtc::MediaType media_type_;
|
| -};
|
| -
|
| const double kXMargin = 1.02;
|
| const double kYMargin = 1.1;
|
| const double kDefaultXMin = -1;
|
| @@ -123,24 +86,138 @@ const double kDefaultYMin = -1;
|
| namespace webrtc {
|
| namespace plotting {
|
|
|
| +
|
| +bool EventLogAnalyzer::StreamId::operator<(const StreamId& other) const {
|
| + if (ssrc_ < other.ssrc_) {
|
| + return true;
|
| + }
|
| + if (ssrc_ == other.ssrc_) {
|
| + if (media_type_ < other.media_type_) {
|
| + return true;
|
| + }
|
| + if (media_type_ == other.media_type_) {
|
| + if (direction_ < other.direction_) {
|
| + return true;
|
| + }
|
| + }
|
| + }
|
| + return false;
|
| +}
|
| +
|
| +bool EventLogAnalyzer::StreamId::operator==(const StreamId& other) const {
|
| + return ssrc_ == other.ssrc_ && direction_ == other.direction_ &&
|
| + media_type_ == other.media_type_;
|
| +}
|
| +
|
| +
|
| EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
|
| : parsed_log_(log), window_duration_(250000), step_(10000) {
|
| uint64_t first_timestamp = std::numeric_limits<uint64_t>::max();
|
| uint64_t last_timestamp = std::numeric_limits<uint64_t>::min();
|
| +
|
| + // Maps a stream identifier consisting of ssrc, direction and MediaType
|
| + // to the header extensions used by that stream,
|
| + std::map<StreamId, RtpHeaderExtensionMap> extension_maps;
|
| +
|
| + PacketDirection direction;
|
| + MediaType media_type;
|
| + uint8_t header[IP_PACKET_SIZE];
|
| + size_t header_length;
|
| + size_t total_length;
|
| +
|
| for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
|
| ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
|
| - if (event_type == ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT)
|
| - continue;
|
| - if (event_type == ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT)
|
| - continue;
|
| - if (event_type == ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT)
|
| - continue;
|
| - if (event_type == ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT)
|
| - continue;
|
| - uint64_t timestamp = parsed_log_.GetTimestamp(i);
|
| - first_timestamp = std::min(first_timestamp, timestamp);
|
| - last_timestamp = std::max(last_timestamp, timestamp);
|
| + if (event_type != ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT &&
|
| + event_type != ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT &&
|
| + event_type != ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT &&
|
| + event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
|
| + uint64_t timestamp = parsed_log_.GetTimestamp(i);
|
| + first_timestamp = std::min(first_timestamp, timestamp);
|
| + last_timestamp = std::max(last_timestamp, timestamp);
|
| + }
|
| +
|
| + switch (parsed_log_.GetEventType(i)) {
|
| + case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
|
| + VideoReceiveStream::Config config(nullptr);
|
| + parsed_log_.GetVideoReceiveConfig(i, &config);
|
| + StreamId stream(config.rtp.remote_ssrc, kIncomingPacket,
|
| + MediaType::VIDEO);
|
| + extension_maps[stream].Erase();
|
| + for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
|
| + const std::string& extension = config.rtp.extensions[j].uri;
|
| + int id = config.rtp.extensions[j].id;
|
| + extension_maps[stream].Register(StringToRtpExtensionType(extension),
|
| + id);
|
| + }
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
|
| + VideoSendStream::Config config(nullptr);
|
| + parsed_log_.GetVideoSendConfig(i, &config);
|
| + for (auto ssrc : config.rtp.ssrcs) {
|
| + StreamId stream(ssrc, kOutgoingPacket, MediaType::VIDEO);
|
| + extension_maps[stream].Erase();
|
| + for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
|
| + const std::string& extension = config.rtp.extensions[j].uri;
|
| + int id = config.rtp.extensions[j].id;
|
| + extension_maps[stream].Register(StringToRtpExtensionType(extension),
|
| + id);
|
| + }
|
| + }
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
|
| + AudioReceiveStream::Config config;
|
| + // TODO(terelius): Parse the audio configs once we have them.
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
|
| + AudioSendStream::Config config(nullptr);
|
| + // TODO(terelius): Parse the audio configs once we have them.
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::RTP_EVENT: {
|
| + parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
|
| + &header_length, &total_length);
|
| + // Parse header to get SSRC.
|
| + RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
|
| + RTPHeader parsed_header;
|
| + rtp_parser.Parse(&parsed_header);
|
| + StreamId stream(parsed_header.ssrc, direction, media_type);
|
| + // Look up the extension_map and parse it again to get the extensions.
|
| + if (extension_maps.count(stream) == 1) {
|
| + RtpHeaderExtensionMap* extension_map = &extension_maps[stream];
|
| + rtp_parser.Parse(&parsed_header, extension_map);
|
| + }
|
| + uint64_t timestamp = parsed_log_.GetTimestamp(i);
|
| + rtp_packets_[stream].push_back(
|
| + LoggedRtpPacket(timestamp, parsed_header));
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::RTCP_EVENT: {
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::LOG_START: {
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::LOG_END: {
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::BWE_PACKET_LOSS_EVENT: {
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::BWE_PACKET_DELAY_EVENT: {
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
|
| + break;
|
| + }
|
| + case ParsedRtcEventLog::UNKNOWN_EVENT: {
|
| + break;
|
| + }
|
| + }
|
| }
|
| +
|
| if (last_timestamp < first_timestamp) {
|
| // No useful events in the log.
|
| first_timestamp = last_timestamp = 0;
|
| @@ -307,111 +384,50 @@ void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) {
|
| }
|
|
|
| void EventLogAnalyzer::CreateDelayChangeGraph(Plot* plot) {
|
| - // Maps a stream identifier consisting of ssrc, direction and MediaType
|
| - // to the header extensions used by that stream,
|
| - std::map<StreamId, RtpHeaderExtensionMap> extension_maps;
|
| -
|
| - struct SendReceiveTime {
|
| - SendReceiveTime() = default;
|
| - SendReceiveTime(uint32_t send_time, uint64_t recv_time)
|
| - : absolute_send_time(send_time), receive_timestamp(recv_time) {}
|
| - uint32_t absolute_send_time; // 24-bit value in units of 2^-18 seconds.
|
| - uint64_t receive_timestamp; // In microseconds.
|
| - };
|
| - std::map<StreamId, SendReceiveTime> last_packet;
|
| - std::map<StreamId, TimeSeries> time_series;
|
| -
|
| - PacketDirection direction;
|
| - MediaType media_type;
|
| - uint8_t header[IP_PACKET_SIZE];
|
| - size_t header_length, total_length;
|
| -
|
| double max_y = 10;
|
| double min_y = 0;
|
|
|
| - for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
|
| - ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
|
| - if (event_type == ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
|
| - VideoReceiveStream::Config config(nullptr);
|
| - parsed_log_.GetVideoReceiveConfig(i, &config);
|
| - StreamId stream(config.rtp.remote_ssrc, kIncomingPacket,
|
| - MediaType::VIDEO);
|
| - extension_maps[stream].Erase();
|
| - for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
|
| - const std::string& extension = config.rtp.extensions[j].uri;
|
| - int id = config.rtp.extensions[j].id;
|
| - extension_maps[stream].Register(StringToRtpExtensionType(extension),
|
| - id);
|
| - }
|
| - } else if (event_type == ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
|
| - VideoSendStream::Config config(nullptr);
|
| - parsed_log_.GetVideoSendConfig(i, &config);
|
| - for (auto ssrc : config.rtp.ssrcs) {
|
| - StreamId stream(ssrc, kIncomingPacket, MediaType::VIDEO);
|
| - extension_maps[stream].Erase();
|
| - for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
|
| - const std::string& extension = config.rtp.extensions[j].uri;
|
| - int id = config.rtp.extensions[j].id;
|
| - extension_maps[stream].Register(StringToRtpExtensionType(extension),
|
| - id);
|
| - }
|
| - }
|
| - } else if (event_type == ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
|
| - AudioReceiveStream::Config config;
|
| - // TODO(terelius): Parse the audio configs once we have them
|
| - } else if (event_type == ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
|
| - AudioSendStream::Config config(nullptr);
|
| - // TODO(terelius): Parse the audio configs once we have them
|
| - } else if (event_type == ParsedRtcEventLog::RTP_EVENT) {
|
| - parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
|
| - &header_length, &total_length);
|
| - if (direction == kIncomingPacket) {
|
| - // Parse header to get SSRC.
|
| - RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
|
| - RTPHeader parsed_header;
|
| - rtp_parser.Parse(&parsed_header);
|
| - // Filter on SSRC.
|
| - if (MatchingSsrc(parsed_header.ssrc, desired_ssrc_)) {
|
| - StreamId stream(parsed_header.ssrc, direction, media_type);
|
| - // Look up the extension_map and parse it again to get the extensions.
|
| - if (extension_maps.count(stream) == 1) {
|
| - RtpHeaderExtensionMap* extension_map = &extension_maps[stream];
|
| - rtp_parser.Parse(&parsed_header, extension_map);
|
| - if (parsed_header.extension.hasAbsoluteSendTime) {
|
| - uint64_t timestamp = parsed_log_.GetTimestamp(i);
|
| - int64_t send_time_diff = WrappingDifference(
|
| - parsed_header.extension.absoluteSendTime,
|
| - last_packet[stream].absolute_send_time, 1ul << 24);
|
| - int64_t recv_time_diff =
|
| - timestamp - last_packet[stream].receive_timestamp;
|
| -
|
| - float x = static_cast<float>(timestamp - begin_time_) / 1000000;
|
| - double y = static_cast<double>(
|
| - recv_time_diff -
|
| - AbsSendTimeToMicroseconds(send_time_diff)) /
|
| - 1000;
|
| - if (time_series[stream].points.size() == 0) {
|
| - // There were no previusly logged playout for this SSRC.
|
| - // Generate a point, but place it on the x-axis.
|
| - y = 0;
|
| - }
|
| - max_y = std::max(max_y, y);
|
| - min_y = std::min(min_y, y);
|
| - time_series[stream].points.push_back(TimeSeriesPoint(x, y));
|
| - last_packet[stream] = SendReceiveTime(
|
| - parsed_header.extension.absoluteSendTime, timestamp);
|
| - }
|
| - }
|
| + for (auto& kv : rtp_packets_) {
|
| + StreamId stream_id = kv.first;
|
| + // Filter on direction and SSRC.
|
| + if (stream_id.GetDirection() != kIncomingPacket ||
|
| + !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
|
| + continue;
|
| + }
|
| +
|
| + TimeSeries time_series;
|
| + time_series.label = SsrcToString(stream_id.GetSsrc());
|
| + time_series.style = BAR_GRAPH;
|
| + const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
|
| + int64_t last_abs_send_time = 0;
|
| + int64_t last_timestamp = 0;
|
| + for (const LoggedRtpPacket& packet : packet_stream) {
|
| + if (packet.header.extension.hasAbsoluteSendTime) {
|
| + int64_t send_time_diff =
|
| + WrappingDifference(packet.header.extension.absoluteSendTime,
|
| + last_abs_send_time, 1ul << 24);
|
| + int64_t recv_time_diff = packet.timestamp - last_timestamp;
|
| +
|
| + last_abs_send_time = packet.header.extension.absoluteSendTime;
|
| + last_timestamp = packet.timestamp;
|
| +
|
| + float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
|
| + double y =
|
| + static_cast<double>(recv_time_diff -
|
| + AbsSendTimeToMicroseconds(send_time_diff)) /
|
| + 1000;
|
| + if (time_series.points.size() == 0) {
|
| + // There were no previously logged packets for this SSRC.
|
| + // Generate a point, but place it on the x-axis.
|
| + y = 0;
|
| }
|
| + max_y = std::max(max_y, y);
|
| + min_y = std::min(min_y, y);
|
| + time_series.points.emplace_back(x, y);
|
| }
|
| }
|
| - }
|
| -
|
| - // Set labels and put in graph.
|
| - for (auto& kv : time_series) {
|
| - kv.second.label = SsrcToString(kv.first.GetSsrc());
|
| - kv.second.style = BAR_GRAPH;
|
| - plot->series.push_back(std::move(kv.second));
|
| + // Add the data set to the plot.
|
| + plot->series.push_back(std::move(time_series));
|
| }
|
|
|
| plot->xaxis_min = kDefaultXMin;
|
| @@ -424,117 +440,50 @@ void EventLogAnalyzer::CreateDelayChangeGraph(Plot* plot) {
|
| }
|
|
|
| void EventLogAnalyzer::CreateAccumulatedDelayChangeGraph(Plot* plot) {
|
| - // TODO(terelius): Refactor
|
| -
|
| - // Maps a stream identifier consisting of ssrc, direction and MediaType
|
| - // to the header extensions used by that stream.
|
| - std::map<StreamId, RtpHeaderExtensionMap> extension_maps;
|
| -
|
| - struct SendReceiveTime {
|
| - SendReceiveTime() = default;
|
| - SendReceiveTime(uint32_t send_time, uint64_t recv_time, double accumulated)
|
| - : absolute_send_time(send_time),
|
| - receive_timestamp(recv_time),
|
| - accumulated_delay(accumulated) {}
|
| - uint32_t absolute_send_time; // 24-bit value in units of 2^-18 seconds.
|
| - uint64_t receive_timestamp; // In microseconds.
|
| - double accumulated_delay; // In milliseconds.
|
| - };
|
| - std::map<StreamId, SendReceiveTime> last_packet;
|
| - std::map<StreamId, TimeSeries> time_series;
|
| -
|
| - PacketDirection direction;
|
| - MediaType media_type;
|
| - uint8_t header[IP_PACKET_SIZE];
|
| - size_t header_length, total_length;
|
| -
|
| double max_y = 10;
|
| double min_y = 0;
|
|
|
| - for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
|
| - ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
|
| - if (event_type == ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
|
| - VideoReceiveStream::Config config(nullptr);
|
| - parsed_log_.GetVideoReceiveConfig(i, &config);
|
| - StreamId stream(config.rtp.remote_ssrc, kIncomingPacket,
|
| - MediaType::VIDEO);
|
| - extension_maps[stream].Erase();
|
| - for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
|
| - const std::string& extension = config.rtp.extensions[j].uri;
|
| - int id = config.rtp.extensions[j].id;
|
| - extension_maps[stream].Register(StringToRtpExtensionType(extension),
|
| - id);
|
| - }
|
| - } else if (event_type == ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
|
| - VideoSendStream::Config config(nullptr);
|
| - parsed_log_.GetVideoSendConfig(i, &config);
|
| - for (auto ssrc : config.rtp.ssrcs) {
|
| - StreamId stream(ssrc, kIncomingPacket, MediaType::VIDEO);
|
| - extension_maps[stream].Erase();
|
| - for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
|
| - const std::string& extension = config.rtp.extensions[j].uri;
|
| - int id = config.rtp.extensions[j].id;
|
| - extension_maps[stream].Register(StringToRtpExtensionType(extension),
|
| - id);
|
| - }
|
| - }
|
| - } else if (event_type == ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
|
| - AudioReceiveStream::Config config;
|
| - // TODO(terelius): Parse the audio configs once we have them
|
| - } else if (event_type == ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
|
| - AudioSendStream::Config config(nullptr);
|
| - // TODO(terelius): Parse the audio configs once we have them
|
| - } else if (event_type == ParsedRtcEventLog::RTP_EVENT) {
|
| - parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
|
| - &header_length, &total_length);
|
| - if (direction == kIncomingPacket) {
|
| - // Parse header to get SSRC.
|
| - RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
|
| - RTPHeader parsed_header;
|
| - rtp_parser.Parse(&parsed_header);
|
| - // Filter on SSRC.
|
| - if (MatchingSsrc(parsed_header.ssrc, desired_ssrc_)) {
|
| - StreamId stream(parsed_header.ssrc, direction, media_type);
|
| - // Look up the extension_map and parse it again to get the extensions.
|
| - if (extension_maps.count(stream) == 1) {
|
| - RtpHeaderExtensionMap* extension_map = &extension_maps[stream];
|
| - rtp_parser.Parse(&parsed_header, extension_map);
|
| - if (parsed_header.extension.hasAbsoluteSendTime) {
|
| - uint64_t timestamp = parsed_log_.GetTimestamp(i);
|
| - int64_t send_time_diff = WrappingDifference(
|
| - parsed_header.extension.absoluteSendTime,
|
| - last_packet[stream].absolute_send_time, 1ul << 24);
|
| - int64_t recv_time_diff =
|
| - timestamp - last_packet[stream].receive_timestamp;
|
| -
|
| - float x = static_cast<float>(timestamp - begin_time_) / 1000000;
|
| - double y = last_packet[stream].accumulated_delay +
|
| - static_cast<double>(
|
| - recv_time_diff -
|
| - AbsSendTimeToMicroseconds(send_time_diff)) /
|
| - 1000;
|
| - if (time_series[stream].points.size() == 0) {
|
| - // There were no previusly logged playout for this SSRC.
|
| - // Generate a point, but place it on the x-axis.
|
| - y = 0;
|
| - }
|
| - max_y = std::max(max_y, y);
|
| - min_y = std::min(min_y, y);
|
| - time_series[stream].points.push_back(TimeSeriesPoint(x, y));
|
| - last_packet[stream] = SendReceiveTime(
|
| - parsed_header.extension.absoluteSendTime, timestamp, y);
|
| - }
|
| - }
|
| + for (auto& kv : rtp_packets_) {
|
| + StreamId stream_id = kv.first;
|
| + // Filter on direction and SSRC.
|
| + if (stream_id.GetDirection() != kIncomingPacket ||
|
| + !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
|
| + continue;
|
| + }
|
| + TimeSeries time_series;
|
| + time_series.label = SsrcToString(stream_id.GetSsrc());
|
| + time_series.style = LINE_GRAPH;
|
| + const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
|
| + int64_t last_abs_send_time = 0;
|
| + int64_t last_timestamp = 0;
|
| + double accumulated_delay_ms = 0;
|
| + for (const LoggedRtpPacket& packet : packet_stream) {
|
| + if (packet.header.extension.hasAbsoluteSendTime) {
|
| + int64_t send_time_diff =
|
| + WrappingDifference(packet.header.extension.absoluteSendTime,
|
| + last_abs_send_time, 1ul << 24);
|
| + int64_t recv_time_diff = packet.timestamp - last_timestamp;
|
| +
|
| + last_abs_send_time = packet.header.extension.absoluteSendTime;
|
| + last_timestamp = packet.timestamp;
|
| +
|
| + float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
|
| + accumulated_delay_ms +=
|
| + static_cast<double>(recv_time_diff -
|
| + AbsSendTimeToMicroseconds(send_time_diff)) /
|
| + 1000;
|
| + if (time_series.points.size() == 0) {
|
| + // There were no previously logged packets for this SSRC.
|
| + // Generate a point, but place it on the x-axis.
|
| + accumulated_delay_ms = 0;
|
| }
|
| + max_y = std::max(max_y, accumulated_delay_ms);
|
| + min_y = std::min(min_y, accumulated_delay_ms);
|
| + time_series.points.emplace_back(x, accumulated_delay_ms);
|
| }
|
| }
|
| - }
|
| -
|
| - // Set labels and put in graph.
|
| - for (auto& kv : time_series) {
|
| - kv.second.label = SsrcToString(kv.first.GetSsrc());
|
| - kv.second.style = LINE_GRAPH;
|
| - plot->series.push_back(std::move(kv.second));
|
| + // Add the data set to the plot.
|
| + plot->series.push_back(std::move(time_series));
|
| }
|
|
|
| plot->xaxis_min = kDefaultXMin;
|
| @@ -630,7 +579,7 @@ void EventLogAnalyzer::CreateStreamBitrateGraph(
|
| uint64_t timestamp;
|
| size_t size;
|
| };
|
| - std::map<uint32_t, std::vector<TimestampSize> > packets;
|
| + std::map<uint32_t, std::vector<TimestampSize>> packets;
|
|
|
| PacketDirection direction;
|
| MediaType media_type;
|
|
|