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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/rtc_tools/event_log_visualizer/analyzer.h" | 11 #include "webrtc/rtc_tools/event_log_visualizer/analyzer.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 #include <limits> | 14 #include <limits> |
15 #include <map> | 15 #include <map> |
16 #include <sstream> | 16 #include <sstream> |
17 #include <string> | 17 #include <string> |
18 #include <utility> | 18 #include <utility> |
19 | 19 |
20 #include "webrtc/call/audio_receive_stream.h" | 20 #include "webrtc/call/audio_receive_stream.h" |
21 #include "webrtc/call/audio_send_stream.h" | 21 #include "webrtc/call/audio_send_stream.h" |
22 #include "webrtc/call/call.h" | 22 #include "webrtc/call/call.h" |
23 #include "webrtc/call/video_receive_stream.h" | 23 #include "webrtc/call/video_receive_stream.h" |
24 #include "webrtc/call/video_send_stream.h" | 24 #include "webrtc/call/video_send_stream.h" |
25 #include "webrtc/common_types.h" | 25 #include "webrtc/common_types.h" |
| 26 #include "webrtc/logging/rtc_event_log/rtc_stream_config.h" |
26 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" | 27 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" |
27 #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" | 28 #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" |
28 #include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" | 29 #include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" |
29 #include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h" | 30 #include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h" |
30 #include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h" | 31 #include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h" |
31 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" | 32 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" |
32 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" | 33 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" |
33 #include "webrtc/modules/include/module_common_types.h" | 34 #include "webrtc/modules/include/module_common_types.h" |
34 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 35 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
35 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 36 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
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1728 plot->AppendTimeSeries(std::move(series.second)); | 1729 plot->AppendTimeSeries(std::move(series.second)); |
1729 } | 1730 } |
1730 | 1731 |
1731 plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin); | 1732 plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin); |
1732 plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin, | 1733 plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin, |
1733 kTopMargin); | 1734 kTopMargin); |
1734 plot->SetTitle("NetEq timing"); | 1735 plot->SetTitle("NetEq timing"); |
1735 } | 1736 } |
1736 } // namespace plotting | 1737 } // namespace plotting |
1737 } // namespace webrtc | 1738 } // namespace webrtc |
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