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The new API stores events gathered by event type. For example, it is possible to ask for 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: Ib244338c86a2c1a010c668a7aba440482023b512 Reviewed-on: https://webrtc-review.googlesource.com/73140 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23056}
65 lines
1.9 KiB
C++
65 lines
1.9 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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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
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#include <memory>
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#include <string>
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#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class RtpHeaderParser;
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namespace test {
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class Packet;
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class RtcEventLogSource : public PacketSource {
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public:
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// Creates an RtcEventLogSource reading from |file_name|. If the file cannot
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// be opened, or has the wrong format, NULL will be returned.
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static RtcEventLogSource* Create(const std::string& file_name);
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virtual ~RtcEventLogSource();
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// Registers an RTP header extension and binds it to |id|.
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virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
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std::unique_ptr<Packet> NextPacket() override;
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// Returns the timestamp of the next audio output event, in milliseconds. The
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// maximum value of int64_t is returned if there are no more audio output
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// events available.
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int64_t NextAudioOutputEventMs();
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private:
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RtcEventLogSource();
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bool OpenFile(const std::string& file_name);
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size_t rtp_packet_index_ = 0;
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size_t audio_output_index_ = 0;
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ParsedRtcEventLogNew parsed_stream_;
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std::unique_ptr<RtpHeaderParser> parser_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
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