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ff61273c01
This reverts commit 9e336ec0b8.
Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming.
Original change's description:
> Create new API for RtcEventLogParser.
>
> 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}
TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org
Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/72500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23026}
108 lines
3.7 KiB
C++
108 lines
3.7 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
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#include <assert.h>
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#include <string.h>
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#include <iostream>
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#include <limits>
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
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RtcEventLogSource* source = new RtcEventLogSource();
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RTC_CHECK(source->OpenFile(file_name));
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return source;
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}
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RtcEventLogSource::~RtcEventLogSource() {}
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bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) {
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RTC_CHECK(parser_.get());
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return parser_->RegisterRtpHeaderExtension(type, id);
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}
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std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
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for (; rtp_packet_index_ < parsed_stream_.GetNumberOfEvents();
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rtp_packet_index_++) {
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if (parsed_stream_.GetEventType(rtp_packet_index_) ==
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ParsedRtcEventLog::RTP_EVENT) {
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PacketDirection direction;
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size_t header_length;
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size_t packet_length;
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uint64_t timestamp_us = parsed_stream_.GetTimestamp(rtp_packet_index_);
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parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, nullptr,
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&header_length, &packet_length, nullptr);
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if (direction != kIncomingPacket) {
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continue;
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}
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uint8_t* packet_header = new uint8_t[header_length];
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parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, packet_header,
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nullptr, nullptr, nullptr);
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std::unique_ptr<Packet> packet(
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new Packet(packet_header, header_length, packet_length,
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static_cast<double>(timestamp_us) / 1000, *parser_.get()));
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if (!packet->valid_header()) {
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std::cout << "Warning: Packet with index " << rtp_packet_index_
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<< " has an invalid header and will be ignored." << std::endl;
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continue;
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}
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if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
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webrtc::ParsedRtcEventLog::MediaType::AUDIO) {
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continue;
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}
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// Check if the packet should not be filtered out.
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if (!filter_.test(packet->header().payloadType) &&
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!(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
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++rtp_packet_index_;
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return packet;
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}
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}
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}
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return nullptr;
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}
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int64_t RtcEventLogSource::NextAudioOutputEventMs() {
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while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
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if (parsed_stream_.GetEventType(audio_output_index_) ==
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ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
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uint64_t timestamp_us = parsed_stream_.GetTimestamp(audio_output_index_);
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// We call GetAudioPlayout only to check that the protobuf event is
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// well-formed.
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parsed_stream_.GetAudioPlayout(audio_output_index_, nullptr);
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audio_output_index_++;
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return timestamp_us / 1000;
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}
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audio_output_index_++;
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}
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return std::numeric_limits<int64_t>::max();
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}
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RtcEventLogSource::RtcEventLogSource()
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: PacketSource(), parser_(RtpHeaderParser::Create()) {}
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bool RtcEventLogSource::OpenFile(const std::string& file_name) {
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return parsed_stream_.ParseFile(file_name);
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}
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} // namespace test
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} // namespace webrtc
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