mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
9e336ec0b8
The new API stores events gathered by event type. For example, it is possible to ask fo a list of all incoming RTCP messages or all audio playout events. The new API is experimental and may change over next few weeks. Once it has stabilized and all unit tests and existing tools have been ported to the new API, the old one will be removed. This CL also updates the event_log_visualizer tool to use the new parser API. This is not a funcional change except for: - Incoming and outgoing audio level are now drawn in two separate plots. - Incoming and outgoing timstamps are now drawn in two separate plots. - RTCP count is no longer split into Video and Audio. It also counts all RTCP packets rather than only specific message types. - Slight timing difference in sendside BWE simulation due to only iterating over transport feedbacks and not over all RTCP packets. This timing changes are not visible in the plots. Media type for RTCP messages might not be identified correctly by rtc_event_log2text anymore. On the other hand, assigning a specific media type to an RTCP packet was a bit hacky to begin with. Bug: webrtc:8111 Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b Reviewed-on: https://webrtc-review.googlesource.com/60865 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23015}
106 lines
3.5 KiB
C++
106 lines
3.5 KiB
C++
/*
|
|
* Copyright (c) 2015 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 "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
|
|
|
|
#include <assert.h>
|
|
#include <string.h>
|
|
#include <iostream>
|
|
#include <limits>
|
|
|
|
#include "modules/audio_coding/neteq/tools/packet.h"
|
|
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
|
|
RtcEventLogSource* source = new RtcEventLogSource();
|
|
RTC_CHECK(source->OpenFile(file_name));
|
|
return source;
|
|
}
|
|
|
|
RtcEventLogSource::~RtcEventLogSource() {}
|
|
|
|
bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
|
|
uint8_t id) {
|
|
RTC_CHECK(parser_.get());
|
|
return parser_->RegisterRtpHeaderExtension(type, id);
|
|
}
|
|
|
|
std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
|
|
for (; rtp_packet_index_ < parsed_stream_.GetNumberOfEvents();
|
|
rtp_packet_index_++) {
|
|
if (parsed_stream_.GetEventType(rtp_packet_index_) ==
|
|
ParsedRtcEventLog::RTP_EVENT) {
|
|
PacketDirection direction;
|
|
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, nullptr,
|
|
&header_length, &packet_length, nullptr);
|
|
|
|
if (direction != kIncomingPacket) {
|
|
continue;
|
|
}
|
|
|
|
uint8_t* packet_header = new uint8_t[header_length];
|
|
parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, packet_header,
|
|
nullptr, 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) !=
|
|
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_)) {
|
|
++rtp_packet_index_;
|
|
return packet;
|
|
}
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
int64_t RtcEventLogSource::NextAudioOutputEventMs() {
|
|
while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
|
|
if (parsed_stream_.GetEventType(audio_output_index_) ==
|
|
ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
|
|
LoggedAudioPlayoutEvent playout_event =
|
|
parsed_stream_.GetAudioPlayout(audio_output_index_);
|
|
audio_output_index_++;
|
|
return playout_event.timestamp_us / 1000;
|
|
}
|
|
audio_output_index_++;
|
|
}
|
|
return std::numeric_limits<int64_t>::max();
|
|
}
|
|
|
|
RtcEventLogSource::RtcEventLogSource()
|
|
: PacketSource(), parser_(RtpHeaderParser::Create()) {}
|
|
|
|
bool RtcEventLogSource::OpenFile(const std::string& file_name) {
|
|
return parsed_stream_.ParseFile(file_name);
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|