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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
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19
logging/rtc_event_log/rtc_event_log_unittest.cc
View File

@ -34,7 +34,7 @@
#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_parser2.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "logging/rtc_event_log/rtc_event_log_unittest_helper.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
@ -750,16 +750,17 @@ TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
for (size_t i = 1; i < parsed_log.GetNumberOfEvents() - 1; i++) {
EXPECT_EQ(parsed_log.GetEventType(i),
ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT);
LoggedAudioPlayoutEvent playout_event = parsed_log.GetAudioPlayout(i);
EXPECT_LT(playout_event.ssrc, kNumEvents);
EXPECT_EQ(static_cast<int64_t>(kStartTime + 10000 * playout_event.ssrc),
playout_event.timestamp_us);
uint32_t ssrc;
parsed_log.GetAudioPlayout(i, &ssrc);
int64_t timestamp = parsed_log.GetTimestamp(i);
EXPECT_LT(ssrc, kNumEvents);
EXPECT_EQ(static_cast<int64_t>(kStartTime + 10000 * ssrc), timestamp);
if (last_ssrc)
EXPECT_EQ(playout_event.ssrc, *last_ssrc + 1);
EXPECT_EQ(ssrc, *last_ssrc + 1);
if (last_timestamp)
EXPECT_EQ(playout_event.timestamp_us, *last_timestamp + 10000);
last_ssrc = playout_event.ssrc;
last_timestamp = playout_event.timestamp_us;
EXPECT_EQ(timestamp, *last_timestamp + 10000);
last_ssrc = ssrc;
last_timestamp = timestamp;
}
RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log,
parsed_log.GetNumberOfEvents() - 1);