Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(367)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/logging/rtc_event_log/rtc_event_log2text.cc

Issue 2855143002: Removed RtcEventLog deps to call:call_interfaces. (Closed)
Patch Set: Rebased Created 3 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc ('K') | « webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc ('k') | webrtc/logging/rtc_event_log/rtc_event_log_parser.h » ('j') | webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/logging/rtc_event_log/rtc_event_log2text.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
index fab04c9c5751c3883e22169c9d788d5a52a66334..6d38e5b2ce0fbabfc079c3feb37dc57332e7fe68 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
@@ -14,7 +14,6 @@
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
-#include "webrtc/call/call.h"
#include "webrtc/common_types.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
@@ -52,6 +51,8 @@ DEFINE_string(ssrc,
"Print only packets with this SSRC (decimal or hex, the latter "
"starting with 0x).");
+enum class MediaType { ANY, AUDIO, VIDEO, DATA };
+
static uint32_t filtered_ssrc = 0;
// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
@@ -75,12 +76,10 @@ bool ParseSsrc(std::string str) {
// Struct used for storing SSRCs used in a Stream.
struct Stream {
- Stream(uint32_t ssrc,
- webrtc::MediaType media_type,
- webrtc::PacketDirection direction)
+ Stream(uint32_t ssrc, MediaType media_type, webrtc::PacketDirection direction)
: ssrc(ssrc), media_type(media_type), direction(direction) {}
uint32_t ssrc;
- webrtc::MediaType media_type;
+ MediaType media_type;
webrtc::PacketDirection direction;
};
@@ -89,28 +88,27 @@ std::vector<Stream> global_streams;
// Returns the MediaType for registered SSRCs. Search from the end to use last
// registered types first.
-webrtc::MediaType GetMediaType(uint32_t ssrc,
- webrtc::PacketDirection direction) {
+MediaType GetMediaType(uint32_t ssrc, webrtc::PacketDirection direction) {
for (auto rit = global_streams.rbegin(); rit != global_streams.rend();
++rit) {
if (rit->ssrc == ssrc && rit->direction == direction)
return rit->media_type;
}
- return webrtc::MediaType::ANY;
+ return MediaType::ANY;
}
bool ExcludePacket(webrtc::PacketDirection direction,
- webrtc::MediaType media_type,
+ MediaType media_type,
uint32_t packet_ssrc) {
if (FLAGS_nooutgoing && direction == webrtc::kOutgoingPacket)
return true;
if (FLAGS_noincoming && direction == webrtc::kIncomingPacket)
return true;
- if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
+ if (FLAGS_noaudio && media_type == MediaType::AUDIO)
return true;
- if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+ if (FLAGS_novideo && media_type == MediaType::VIDEO)
return true;
- if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+ if (FLAGS_nodata && media_type == MediaType::DATA)
return true;
if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
return true;
@@ -118,23 +116,23 @@ bool ExcludePacket(webrtc::PacketDirection direction,
}
const char* StreamInfo(webrtc::PacketDirection direction,
terelius 2017/05/23 11:57:58 This could be a toString() on the Stream object.
perkj_webrtc 2017/05/24 12:32:21 Prefer not to. I moved the Stream into the parser.
- webrtc::MediaType media_type) {
+ MediaType media_type) {
if (direction == webrtc::kOutgoingPacket) {
- if (media_type == webrtc::MediaType::AUDIO)
+ if (media_type == MediaType::AUDIO)
return "(out,audio)";
- else if (media_type == webrtc::MediaType::VIDEO)
+ else if (media_type == MediaType::VIDEO)
return "(out,video)";
- else if (media_type == webrtc::MediaType::DATA)
+ else if (media_type == MediaType::DATA)
return "(out,data)";
else
return "(out)";
}
if (direction == webrtc::kIncomingPacket) {
- if (media_type == webrtc::MediaType::AUDIO)
+ if (media_type == MediaType::AUDIO)
return "(in,audio)";
- else if (media_type == webrtc::MediaType::VIDEO)
+ else if (media_type == MediaType::VIDEO)
return "(in,video)";
- else if (media_type == webrtc::MediaType::DATA)
+ else if (media_type == MediaType::DATA)
return "(in,data)";
else
return "(in)";
@@ -148,7 +146,7 @@ void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::SenderReport sr;
if (!sr.Parse(rtcp_block))
return;
- webrtc::MediaType media_type = GetMediaType(sr.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(sr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -163,7 +161,7 @@ void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::ReceiverReport rr;
if (!rr.Parse(rtcp_block))
return;
- webrtc::MediaType media_type = GetMediaType(rr.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(rr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -177,7 +175,7 @@ void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::ExtendedReports xr;
if (!xr.Parse(rtcp_block))
return;
- webrtc::MediaType media_type = GetMediaType(xr.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(xr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -189,7 +187,7 @@ void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
std::cout << log_timestamp << "\t"
- << "RTCP_SDES" << StreamInfo(direction, webrtc::MediaType::ANY)
+ << "RTCP_SDES" << StreamInfo(direction, MediaType::ANY)
<< std::endl;
RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
}
@@ -200,7 +198,7 @@ void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Bye bye;
if (!bye.Parse(rtcp_block))
return;
- webrtc::MediaType media_type = GetMediaType(bye.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(bye.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -216,8 +214,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Nack nack;
if (!nack.Parse(rtcp_block))
return;
- webrtc::MediaType media_type =
- GetMediaType(nack.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(nack.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -229,8 +226,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Tmmbr tmmbr;
if (!tmmbr.Parse(rtcp_block))
return;
- webrtc::MediaType media_type =
- GetMediaType(tmmbr.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(tmmbr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -242,8 +238,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Tmmbn tmmbn;
if (!tmmbn.Parse(rtcp_block))
return;
- webrtc::MediaType media_type =
- GetMediaType(tmmbn.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(tmmbn.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -255,8 +250,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::RapidResyncRequest sr_req;
if (!sr_req.Parse(rtcp_block))
return;
- webrtc::MediaType media_type =
- GetMediaType(sr_req.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(sr_req.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -268,7 +262,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::TransportFeedback transport_feedback;
if (!transport_feedback.Parse(rtcp_block))
return;
- webrtc::MediaType media_type =
+ MediaType media_type =
GetMediaType(transport_feedback.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type,
transport_feedback.sender_ssrc()))
@@ -291,7 +285,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Pli pli;
if (!pli.Parse(rtcp_block))
return;
- webrtc::MediaType media_type = GetMediaType(pli.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(pli.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -303,7 +297,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Fir fir;
if (!fir.Parse(rtcp_block))
return;
- webrtc::MediaType media_type = GetMediaType(fir.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(fir.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -315,8 +309,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Remb remb;
if (!remb.Parse(rtcp_block))
return;
- webrtc::MediaType media_type =
- GetMediaType(remb.sender_ssrc(), direction);
+ MediaType media_type = GetMediaType(remb.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -367,11 +360,9 @@ int main(int argc, char* argv[]) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetVideoReceiveConfig(i, &config);
- global_streams.emplace_back(config.remote_ssrc,
- webrtc::MediaType::VIDEO,
+ global_streams.emplace_back(config.remote_ssrc, MediaType::VIDEO,
webrtc::kIncomingPacket);
- global_streams.emplace_back(config.local_ssrc,
- webrtc::MediaType::VIDEO,
+ global_streams.emplace_back(config.local_ssrc, MediaType::VIDEO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming) {
@@ -384,10 +375,10 @@ int main(int argc, char* argv[]) {
webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetVideoSendConfig(i, &config);
- global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::VIDEO,
+ global_streams.emplace_back(config.local_ssrc, MediaType::VIDEO,
webrtc::kOutgoingPacket);
- global_streams.emplace_back(config.rtx_ssrc, webrtc::MediaType::VIDEO,
+ global_streams.emplace_back(config.rtx_ssrc, MediaType::VIDEO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
@@ -401,11 +392,9 @@ int main(int argc, char* argv[]) {
webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetAudioReceiveConfig(i, &config);
- global_streams.emplace_back(config.remote_ssrc,
- webrtc::MediaType::AUDIO,
+ global_streams.emplace_back(config.remote_ssrc, MediaType::AUDIO,
webrtc::kIncomingPacket);
- global_streams.emplace_back(config.local_ssrc,
- webrtc::MediaType::AUDIO,
+ global_streams.emplace_back(config.local_ssrc, MediaType::AUDIO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
@@ -417,7 +406,7 @@ int main(int argc, char* argv[]) {
webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetAudioSendConfig(i, &config);
- global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::AUDIO,
+ global_streams.emplace_back(config.local_ssrc, MediaType::AUDIO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
@@ -430,15 +419,15 @@ int main(int argc, char* argv[]) {
size_t total_length;
uint8_t header[IP_PACKET_SIZE];
webrtc::PacketDirection direction;
- webrtc::MediaType media_type;
- parsed_stream.GetRtpHeader(i, &direction, &media_type, header,
- &header_length, &total_length);
+
+ parsed_stream.GetRtpHeader(i, &direction, header, &header_length,
+ &total_length);
// Parse header to get SSRC and RTP time.
webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
webrtc::RTPHeader parsed_header;
rtp_parser.Parse(&parsed_header);
- media_type = GetMediaType(parsed_header.ssrc, direction);
+ MediaType media_type = GetMediaType(parsed_header.ssrc, direction);
if (ExcludePacket(direction, media_type, parsed_header.ssrc))
continue;
@@ -454,8 +443,7 @@ int main(int argc, char* argv[]) {
size_t length;
uint8_t packet[IP_PACKET_SIZE];
webrtc::PacketDirection direction;
- webrtc::MediaType media_type;
- parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet, &length);
+ parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
webrtc::rtcp::CommonHeader rtcp_block;
const uint8_t* packet_end = packet + length;
« webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc ('K') | « webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc ('k') | webrtc/logging/rtc_event_log/rtc_event_log_parser.h » ('j') | webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h » ('J')

Powered by Google App Engine
This is Rietveld 408576698