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/*
* Copyright (c) 2016 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.
*/
#ifndef RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
#define RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
#include "rtc_tools/event_log_visualizer/plot_base.h"
namespace webrtc {
Revert "Create new API for RtcEventLogParser." 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}
2018-04-25 14:23:01 +00:00
namespace plotting {
class PythonPlot final : public Plot {
public:
PythonPlot();
~PythonPlot() override;
void Draw() override;
};
class PythonPlotCollection final : public PlotCollection {
public:
PythonPlotCollection();
~PythonPlotCollection() override;
void Draw() override;
Plot* AppendNewPlot() override;
};
Revert "Create new API for RtcEventLogParser." 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}
2018-04-25 14:23:01 +00:00
} // namespace plotting
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_