| Index: webrtc/call/rtc_event_log_parser.cc
|
| diff --git a/webrtc/call/rtc_event_log_parser.cc b/webrtc/call/rtc_event_log_parser.cc
|
| deleted file mode 100644
|
| index a2f95d0a75e2c833b695233489651a1cb57c08b2..0000000000000000000000000000000000000000
|
| --- a/webrtc/call/rtc_event_log_parser.cc
|
| +++ /dev/null
|
| @@ -1,416 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
|
| - *
|
| - * Use of this source code is governed by a BSD-style license
|
| - * that can be found in the LICENSE file in the root of the source
|
| - * tree. An additional intellectual property rights grant can be found
|
| - * in the file PATENTS. All contributing project authors may
|
| - * be found in the AUTHORS file in the root of the source tree.
|
| - */
|
| -
|
| -#include "webrtc/call/rtc_event_log_parser.h"
|
| -
|
| -#include <string.h>
|
| -
|
| -#include <fstream>
|
| -#include <istream>
|
| -#include <utility>
|
| -
|
| -#include "webrtc/base/checks.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/call.h"
|
| -#include "webrtc/call/rtc_event_log.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
| -#include "webrtc/system_wrappers/include/file_wrapper.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -namespace {
|
| -MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
|
| - switch (media_type) {
|
| - case rtclog::MediaType::ANY:
|
| - return MediaType::ANY;
|
| - case rtclog::MediaType::AUDIO:
|
| - return MediaType::AUDIO;
|
| - case rtclog::MediaType::VIDEO:
|
| - return MediaType::VIDEO;
|
| - case rtclog::MediaType::DATA:
|
| - return MediaType::DATA;
|
| - }
|
| - RTC_NOTREACHED();
|
| - return MediaType::ANY;
|
| -}
|
| -
|
| -RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
|
| - switch (rtcp_mode) {
|
| - case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
|
| - return RtcpMode::kCompound;
|
| - case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
|
| - return RtcpMode::kReducedSize;
|
| - }
|
| - RTC_NOTREACHED();
|
| - return RtcpMode::kOff;
|
| -}
|
| -
|
| -ParsedRtcEventLog::EventType GetRuntimeEventType(
|
| - rtclog::Event::EventType event_type) {
|
| - switch (event_type) {
|
| - case rtclog::Event::UNKNOWN_EVENT:
|
| - return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
|
| - case rtclog::Event::LOG_START:
|
| - return ParsedRtcEventLog::EventType::LOG_START;
|
| - case rtclog::Event::LOG_END:
|
| - return ParsedRtcEventLog::EventType::LOG_END;
|
| - case rtclog::Event::RTP_EVENT:
|
| - return ParsedRtcEventLog::EventType::RTP_EVENT;
|
| - case rtclog::Event::RTCP_EVENT:
|
| - return ParsedRtcEventLog::EventType::RTCP_EVENT;
|
| - case rtclog::Event::AUDIO_PLAYOUT_EVENT:
|
| - return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
|
| - case rtclog::Event::BWE_PACKET_LOSS_EVENT:
|
| - return ParsedRtcEventLog::EventType::BWE_PACKET_LOSS_EVENT;
|
| - case rtclog::Event::BWE_PACKET_DELAY_EVENT:
|
| - return ParsedRtcEventLog::EventType::BWE_PACKET_DELAY_EVENT;
|
| - case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
|
| - return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
|
| - case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
|
| - return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
|
| - case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
|
| - return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
|
| - case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
|
| - return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
|
| - }
|
| - RTC_NOTREACHED();
|
| - return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
|
| -}
|
| -
|
| -std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
|
| - uint64_t varint = 0;
|
| - for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
|
| - // The most significant bit of each byte is 0 if it is the last byte in
|
| - // the varint and 1 otherwise. Thus, we take the 7 least significant bits
|
| - // of each byte and shift them 7 bits for each byte read previously to get
|
| - // the (unsigned) integer.
|
| - int byte = stream.get();
|
| - if (stream.eof()) {
|
| - return std::make_pair(varint, false);
|
| - }
|
| - RTC_DCHECK(0 <= byte && byte <= 255);
|
| - varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
|
| - if ((byte & 0x80) == 0) {
|
| - return std::make_pair(varint, true);
|
| - }
|
| - }
|
| - return std::make_pair(varint, false);
|
| -}
|
| -
|
| -} // namespace
|
| -
|
| -bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
|
| - std::ifstream file(filename, std::ios_base::in | std::ios_base::binary);
|
| - if (!file.good() || !file.is_open()) {
|
| - LOG(LS_WARNING) << "Could not open file for reading.";
|
| - return false;
|
| - }
|
| -
|
| - return ParseStream(file);
|
| -}
|
| -
|
| -bool ParsedRtcEventLog::ParseString(const std::string& s) {
|
| - std::istringstream stream(s, std::ios_base::in | std::ios_base::binary);
|
| - return ParseStream(stream);
|
| -}
|
| -
|
| -bool ParsedRtcEventLog::ParseStream(std::istream& stream) {
|
| - events_.clear();
|
| - const size_t kMaxEventSize = (1u << 16) - 1;
|
| - std::vector<char> tmp_buffer(kMaxEventSize);
|
| - uint64_t tag;
|
| - uint64_t message_length;
|
| - bool success;
|
| -
|
| - RTC_DCHECK(stream.good());
|
| -
|
| - while (1) {
|
| - // Check whether we have reached end of file.
|
| - stream.peek();
|
| - if (stream.eof()) {
|
| - return true;
|
| - }
|
| -
|
| - // Read the next message tag. The tag number is defined as
|
| - // (fieldnumber << 3) | wire_type. In our case, the field number is
|
| - // supposed to be 1 and the wire type for an length-delimited field is 2.
|
| - const uint64_t kExpectedTag = (1 << 3) | 2;
|
| - std::tie(tag, success) = ParseVarInt(stream);
|
| - if (!success) {
|
| - LOG(LS_WARNING) << "Missing field tag from beginning of protobuf event.";
|
| - return false;
|
| - } else if (tag != kExpectedTag) {
|
| - LOG(LS_WARNING) << "Unexpected field tag at beginning of protobuf event.";
|
| - return false;
|
| - }
|
| -
|
| - // Read the length field.
|
| - std::tie(message_length, success) = ParseVarInt(stream);
|
| - if (!success) {
|
| - LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
|
| - return false;
|
| - } else if (message_length > kMaxEventSize) {
|
| - LOG(LS_WARNING) << "Protobuf message length is too large.";
|
| - return false;
|
| - }
|
| -
|
| - // Read the next protobuf event to a temporary char buffer.
|
| - stream.read(tmp_buffer.data(), message_length);
|
| - if (stream.gcount() != static_cast<int>(message_length)) {
|
| - LOG(LS_WARNING) << "Failed to read protobuf message from file.";
|
| - return false;
|
| - }
|
| -
|
| - // Parse the protobuf event from the buffer.
|
| - rtclog::Event event;
|
| - if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
|
| - LOG(LS_WARNING) << "Failed to parse protobuf message.";
|
| - return false;
|
| - }
|
| - events_.push_back(event);
|
| - }
|
| -}
|
| -
|
| -size_t ParsedRtcEventLog::GetNumberOfEvents() const {
|
| - return events_.size();
|
| -}
|
| -
|
| -int64_t ParsedRtcEventLog::GetTimestamp(size_t index) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(event.has_timestamp_us());
|
| - return event.timestamp_us();
|
| -}
|
| -
|
| -ParsedRtcEventLog::EventType ParsedRtcEventLog::GetEventType(
|
| - size_t index) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(event.has_type());
|
| - return GetRuntimeEventType(event.type());
|
| -}
|
| -
|
| -// The header must have space for at least IP_PACKET_SIZE bytes.
|
| -void ParsedRtcEventLog::GetRtpHeader(size_t index,
|
| - PacketDirection* incoming,
|
| - MediaType* media_type,
|
| - uint8_t* header,
|
| - size_t* header_length,
|
| - size_t* total_length) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(event.has_type());
|
| - RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
|
| - RTC_CHECK(event.has_rtp_packet());
|
| - const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
|
| - // Get direction of packet.
|
| - RTC_CHECK(rtp_packet.has_incoming());
|
| - if (incoming != nullptr) {
|
| - *incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
|
| - }
|
| - // Get media type.
|
| - RTC_CHECK(rtp_packet.has_type());
|
| - if (media_type != nullptr) {
|
| - *media_type = GetRuntimeMediaType(rtp_packet.type());
|
| - }
|
| - // Get packet length.
|
| - RTC_CHECK(rtp_packet.has_packet_length());
|
| - if (total_length != nullptr) {
|
| - *total_length = rtp_packet.packet_length();
|
| - }
|
| - // Get header length.
|
| - RTC_CHECK(rtp_packet.has_header());
|
| - if (header_length != nullptr) {
|
| - *header_length = rtp_packet.header().size();
|
| - }
|
| - // Get header contents.
|
| - if (header != nullptr) {
|
| - const size_t kMinRtpHeaderSize = 12;
|
| - RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
|
| - RTC_CHECK_LE(rtp_packet.header().size(),
|
| - static_cast<size_t>(IP_PACKET_SIZE));
|
| - memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
|
| - }
|
| -}
|
| -
|
| -// The packet must have space for at least IP_PACKET_SIZE bytes.
|
| -void ParsedRtcEventLog::GetRtcpPacket(size_t index,
|
| - PacketDirection* incoming,
|
| - MediaType* media_type,
|
| - uint8_t* packet,
|
| - size_t* length) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(event.has_type());
|
| - RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
|
| - RTC_CHECK(event.has_rtcp_packet());
|
| - const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
|
| - // Get direction of packet.
|
| - RTC_CHECK(rtcp_packet.has_incoming());
|
| - if (incoming != nullptr) {
|
| - *incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
|
| - }
|
| - // Get media type.
|
| - RTC_CHECK(rtcp_packet.has_type());
|
| - if (media_type != nullptr) {
|
| - *media_type = GetRuntimeMediaType(rtcp_packet.type());
|
| - }
|
| - // Get packet length.
|
| - RTC_CHECK(rtcp_packet.has_packet_data());
|
| - if (length != nullptr) {
|
| - *length = rtcp_packet.packet_data().size();
|
| - }
|
| - // Get packet contents.
|
| - if (packet != nullptr) {
|
| - RTC_CHECK_LE(rtcp_packet.packet_data().size(),
|
| - static_cast<unsigned>(IP_PACKET_SIZE));
|
| - memcpy(packet, rtcp_packet.packet_data().data(),
|
| - rtcp_packet.packet_data().size());
|
| - }
|
| -}
|
| -
|
| -void ParsedRtcEventLog::GetVideoReceiveConfig(
|
| - size_t index,
|
| - VideoReceiveStream::Config* config) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(config != nullptr);
|
| - RTC_CHECK(event.has_type());
|
| - RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
|
| - RTC_CHECK(event.has_video_receiver_config());
|
| - const rtclog::VideoReceiveConfig& receiver_config =
|
| - event.video_receiver_config();
|
| - // Get SSRCs.
|
| - RTC_CHECK(receiver_config.has_remote_ssrc());
|
| - config->rtp.remote_ssrc = receiver_config.remote_ssrc();
|
| - RTC_CHECK(receiver_config.has_local_ssrc());
|
| - config->rtp.local_ssrc = receiver_config.local_ssrc();
|
| - // Get RTCP settings.
|
| - RTC_CHECK(receiver_config.has_rtcp_mode());
|
| - config->rtp.rtcp_mode = GetRuntimeRtcpMode(receiver_config.rtcp_mode());
|
| - RTC_CHECK(receiver_config.has_remb());
|
| - config->rtp.remb = receiver_config.remb();
|
| - // Get RTX map.
|
| - config->rtp.rtx.clear();
|
| - for (int i = 0; i < receiver_config.rtx_map_size(); i++) {
|
| - const rtclog::RtxMap& map = receiver_config.rtx_map(i);
|
| - RTC_CHECK(map.has_payload_type());
|
| - RTC_CHECK(map.has_config());
|
| - RTC_CHECK(map.config().has_rtx_ssrc());
|
| - RTC_CHECK(map.config().has_rtx_payload_type());
|
| - webrtc::VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
|
| - rtx_pair.ssrc = map.config().rtx_ssrc();
|
| - rtx_pair.payload_type = map.config().rtx_payload_type();
|
| - config->rtp.rtx.insert(std::make_pair(map.payload_type(), rtx_pair));
|
| - }
|
| - // Get header extensions.
|
| - config->rtp.extensions.clear();
|
| - for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
|
| - RTC_CHECK(receiver_config.header_extensions(i).has_name());
|
| - RTC_CHECK(receiver_config.header_extensions(i).has_id());
|
| - const std::string& name = receiver_config.header_extensions(i).name();
|
| - int id = receiver_config.header_extensions(i).id();
|
| - config->rtp.extensions.push_back(RtpExtension(name, id));
|
| - }
|
| - // Get decoders.
|
| - config->decoders.clear();
|
| - for (int i = 0; i < receiver_config.decoders_size(); i++) {
|
| - RTC_CHECK(receiver_config.decoders(i).has_name());
|
| - RTC_CHECK(receiver_config.decoders(i).has_payload_type());
|
| - VideoReceiveStream::Decoder decoder;
|
| - decoder.payload_name = receiver_config.decoders(i).name();
|
| - decoder.payload_type = receiver_config.decoders(i).payload_type();
|
| - config->decoders.push_back(decoder);
|
| - }
|
| -}
|
| -
|
| -void ParsedRtcEventLog::GetVideoSendConfig(
|
| - size_t index,
|
| - VideoSendStream::Config* config) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(config != nullptr);
|
| - RTC_CHECK(event.has_type());
|
| - RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
|
| - RTC_CHECK(event.has_video_sender_config());
|
| - const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
|
| - // Get SSRCs.
|
| - config->rtp.ssrcs.clear();
|
| - for (int i = 0; i < sender_config.ssrcs_size(); i++) {
|
| - config->rtp.ssrcs.push_back(sender_config.ssrcs(i));
|
| - }
|
| - // Get header extensions.
|
| - config->rtp.extensions.clear();
|
| - for (int i = 0; i < sender_config.header_extensions_size(); i++) {
|
| - RTC_CHECK(sender_config.header_extensions(i).has_name());
|
| - RTC_CHECK(sender_config.header_extensions(i).has_id());
|
| - const std::string& name = sender_config.header_extensions(i).name();
|
| - int id = sender_config.header_extensions(i).id();
|
| - config->rtp.extensions.push_back(RtpExtension(name, id));
|
| - }
|
| - // Get RTX settings.
|
| - config->rtp.rtx.ssrcs.clear();
|
| - for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
|
| - config->rtp.rtx.ssrcs.push_back(sender_config.rtx_ssrcs(i));
|
| - }
|
| - if (sender_config.rtx_ssrcs_size() > 0) {
|
| - RTC_CHECK(sender_config.has_rtx_payload_type());
|
| - config->rtp.rtx.payload_type = sender_config.rtx_payload_type();
|
| - } else {
|
| - // Reset RTX payload type default value if no RTX SSRCs are used.
|
| - config->rtp.rtx.payload_type = -1;
|
| - }
|
| - // Get encoder.
|
| - RTC_CHECK(sender_config.has_encoder());
|
| - RTC_CHECK(sender_config.encoder().has_name());
|
| - RTC_CHECK(sender_config.encoder().has_payload_type());
|
| - config->encoder_settings.payload_name = sender_config.encoder().name();
|
| - config->encoder_settings.payload_type =
|
| - sender_config.encoder().payload_type();
|
| -}
|
| -
|
| -void ParsedRtcEventLog::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(event.has_type());
|
| - RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
|
| - RTC_CHECK(event.has_audio_playout_event());
|
| - const rtclog::AudioPlayoutEvent& loss_event = event.audio_playout_event();
|
| - RTC_CHECK(loss_event.has_local_ssrc());
|
| - if (ssrc != nullptr) {
|
| - *ssrc = loss_event.local_ssrc();
|
| - }
|
| -}
|
| -
|
| -void ParsedRtcEventLog::GetBwePacketLossEvent(size_t index,
|
| - int32_t* bitrate,
|
| - uint8_t* fraction_loss,
|
| - int32_t* total_packets) const {
|
| - RTC_CHECK_LT(index, GetNumberOfEvents());
|
| - const rtclog::Event& event = events_[index];
|
| - RTC_CHECK(event.has_type());
|
| - RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PACKET_LOSS_EVENT);
|
| - RTC_CHECK(event.has_bwe_packet_loss_event());
|
| - const rtclog::BwePacketLossEvent& loss_event = event.bwe_packet_loss_event();
|
| - RTC_CHECK(loss_event.has_bitrate());
|
| - if (bitrate != nullptr) {
|
| - *bitrate = loss_event.bitrate();
|
| - }
|
| - RTC_CHECK(loss_event.has_fraction_loss());
|
| - if (fraction_loss != nullptr) {
|
| - *fraction_loss = loss_event.fraction_loss();
|
| - }
|
| - RTC_CHECK(loss_event.has_total_packets());
|
| - if (total_packets != nullptr) {
|
| - *total_packets = loss_event.total_packets();
|
| - }
|
| -}
|
| -
|
| -} // namespace webrtc
|
|
|