zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

Browse Source 
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.

BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc

Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
This commit is contained in:
perkj
2017-05-30 03:52:10 -07:00
committed by Commit Bot
parent 367aba92bf
commit 77cd58e140
21 changed files with 295 additions and 327 deletions
Download Patch File Download Diff File Expand all files Collapse all files

55
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
View File

@ -39,38 +39,45 @@ bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
}
std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
while (rtp_packet_index_ < parsed_stream_.GetNumberOfEvents()) {
for (; rtp_packet_index_ < parsed_stream_.GetNumberOfEvents();
rtp_packet_index_++) {
if (parsed_stream_.GetEventType(rtp_packet_index_) ==
ParsedRtcEventLog::RTP_EVENT) {
PacketDirection direction;
MediaType media_type;
size_t header_length;
size_t packet_length;
uint64_t timestamp_us = parsed_stream_.GetTimestamp(rtp_packet_index_);
parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, &media_type,
nullptr, &header_length, &packet_length);
if (direction == kIncomingPacket && media_type == MediaType::AUDIO) {
uint8_t* packet_header = new uint8_t[header_length];
parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, nullptr,
packet_header, nullptr, nullptr);
std::unique_ptr<Packet> packet(new Packet(
packet_header, header_length, packet_length,
static_cast<double>(timestamp_us) / 1000, *parser_.get()));
if (packet->valid_header()) {
// Check if the packet should not be filtered out.
if (!filter_.test(packet->header().payloadType) &&
!(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
rtp_packet_index_++;
return packet;
}
} else {
std::cout << "Warning: Packet with index " << rtp_packet_index_
<< " has an invalid header and will be ignored."
<< std::endl;
}
parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, nullptr,
&header_length, &packet_length);
if (direction != kIncomingPacket) {
continue;
}
uint8_t* packet_header = new uint8_t[header_length];
parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, packet_header,
nullptr, nullptr);
std::unique_ptr<Packet> packet(
new Packet(packet_header, header_length, packet_length,
static_cast<double>(timestamp_us) / 1000, *parser_.get()));
if (!packet->valid_header()) {
std::cout << "Warning: Packet with index " << rtp_packet_index_
<< " has an invalid header and will be ignored." << std::endl;
continue;
}
if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
webrtc::ParsedRtcEventLog::MediaType::AUDIO) {
continue;
}
// Check if the packet should not be filtered out.
if (!filter_.test(packet->header().payloadType) &&
!(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
return packet;
}
}
rtp_packet_index_++;
}
return nullptr;
}