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Revert "Create new API for RtcEventLogParser."

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This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa.

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}
This commit is contained in:
Björn Terelius
2018-04-25 14:23:01 +00:00
committed by Commit Bot
parent 65fb4049c1
commit ff61273c01
23 changed files with 1422 additions and 3279 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
modules/audio_coding/neteq/tools/rtc_event_log_source.cc
View File

@ -66,7 +66,7 @@ std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
}
if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
ParsedRtcEventLog::MediaType::AUDIO) {
webrtc::ParsedRtcEventLog::MediaType::AUDIO) {
continue;
}
@ -85,10 +85,12 @@ 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_);
uint64_t timestamp_us = parsed_stream_.GetTimestamp(audio_output_index_);
// We call GetAudioPlayout only to check that the protobuf event is
// well-formed.
parsed_stream_.GetAudioPlayout(audio_output_index_, nullptr);
audio_output_index_++;
return playout_event.timestamp_us / 1000;
return timestamp_us / 1000;
}
audio_output_index_++;
}

2
modules/audio_coding/neteq/tools/rtc_event_log_source.h
View File

@ -14,7 +14,7 @@
#include <memory>
#include <string>
#include "logging/rtc_event_log/rtc_event_log_parser2.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/constructormagic.h"