OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 #include <stdio.h> | 10 #include <stdio.h> |
11 | 11 |
12 #include <algorithm> | 12 #include <algorithm> |
13 #include <deque> | 13 #include <deque> |
14 #include <map> | 14 #include <map> |
15 #include <sstream> | 15 #include <sstream> |
16 #include <string> | 16 #include <string> |
17 #include <vector> | 17 #include <vector> |
18 | 18 |
19 #include "testing/gtest/include/gtest/gtest.h" | 19 #include "testing/gtest/include/gtest/gtest.h" |
20 | 20 |
21 #include "webrtc/base/checks.h" | 21 #include "webrtc/base/checks.h" |
22 #include "webrtc/base/event.h" | 22 #include "webrtc/base/event.h" |
23 #include "webrtc/base/format_macros.h" | 23 #include "webrtc/base/format_macros.h" |
24 #include "webrtc/base/scoped_ptr.h" | 24 #include "webrtc/base/scoped_ptr.h" |
25 #include "webrtc/call.h" | 25 #include "webrtc/call.h" |
26 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" | 26 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
27 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 27 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
28 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 28 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
29 #include "webrtc/system_wrappers/include/cpu_info.h" | 29 #include "webrtc/system_wrappers/include/cpu_info.h" |
| 30 #include "webrtc/system_wrappers/include/tick_util.h" |
30 #include "webrtc/test/layer_filtering_transport.h" | 31 #include "webrtc/test/layer_filtering_transport.h" |
31 #include "webrtc/test/run_loop.h" | 32 #include "webrtc/test/run_loop.h" |
32 #include "webrtc/test/statistics.h" | 33 #include "webrtc/test/statistics.h" |
33 #include "webrtc/test/testsupport/fileutils.h" | 34 #include "webrtc/test/testsupport/fileutils.h" |
34 #include "webrtc/test/video_renderer.h" | 35 #include "webrtc/test/video_renderer.h" |
| 36 #include "webrtc/video/video_capture_input.h" |
35 #include "webrtc/video/video_quality_test.h" | 37 #include "webrtc/video/video_quality_test.h" |
36 | 38 |
37 namespace webrtc { | 39 namespace webrtc { |
38 | 40 |
39 static const int kSendStatsPollingIntervalMs = 1000; | 41 static const int kSendStatsPollingIntervalMs = 1000; |
40 static const int kPayloadTypeVP8 = 123; | 42 static const int kPayloadTypeVP8 = 123; |
41 static const int kPayloadTypeVP9 = 124; | 43 static const int kPayloadTypeVP9 = 124; |
42 | 44 |
43 class VideoAnalyzer : public PacketReceiver, | 45 class VideoAnalyzer : public PacketReceiver, |
44 public Transport, | 46 public Transport, |
(...skipping 81 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
126 return receiver_->DeliverPacket(media_type, packet, length, packet_time); | 128 return receiver_->DeliverPacket(media_type, packet, length, packet_time); |
127 } | 129 } |
128 | 130 |
129 // EncodingTimeObserver. | 131 // EncodingTimeObserver. |
130 void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) override { | 132 void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) override { |
131 rtc::CritScope crit(&comparison_lock_); | 133 rtc::CritScope crit(&comparison_lock_); |
132 samples_encode_time_ms_[ntp_time_ms] = encode_time_ms; | 134 samples_encode_time_ms_[ntp_time_ms] = encode_time_ms; |
133 } | 135 } |
134 | 136 |
135 void IncomingCapturedFrame(const VideoFrame& video_frame) override { | 137 void IncomingCapturedFrame(const VideoFrame& video_frame) override { |
136 VideoFrame copy = video_frame; | 138 VideoFrame copy = static_cast<internal::VideoCaptureInput*>(input_) |
137 copy.set_timestamp(copy.ntp_time_ms() * 90); | 139 ->UpdateTimestamps(video_frame); |
138 | 140 |
139 { | 141 { |
140 rtc::CritScope lock(&crit_); | 142 rtc::CritScope lock(&crit_); |
141 if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0) | 143 if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0) |
142 first_send_frame_ = copy; | 144 first_send_frame_ = copy; |
143 | 145 |
144 frames_.push_back(copy); | 146 frames_.push_back(copy); |
145 } | 147 } |
146 | 148 |
147 input_->IncomingCapturedFrame(video_frame); | 149 input_->IncomingCapturedFrame(copy); |
148 } | 150 } |
149 | 151 |
150 bool SendRtp(const uint8_t* packet, | 152 bool SendRtp(const uint8_t* packet, |
151 size_t length, | 153 size_t length, |
152 const PacketOptions& options) override { | 154 const PacketOptions& options) override { |
153 RtpUtility::RtpHeaderParser parser(packet, length); | 155 RtpUtility::RtpHeaderParser parser(packet, length); |
154 RTPHeader header; | 156 RTPHeader header; |
155 parser.Parse(header); | 157 parser.Parse(header); |
156 | 158 |
157 int64_t current_time = | 159 int64_t current_time = |
(...skipping 903 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1061 send_stream_->Stop(); | 1063 send_stream_->Stop(); |
1062 receive_stream->Stop(); | 1064 receive_stream->Stop(); |
1063 | 1065 |
1064 call->DestroyVideoReceiveStream(receive_stream); | 1066 call->DestroyVideoReceiveStream(receive_stream); |
1065 call->DestroyVideoSendStream(send_stream_); | 1067 call->DestroyVideoSendStream(send_stream_); |
1066 | 1068 |
1067 transport.StopSending(); | 1069 transport.StopSending(); |
1068 } | 1070 } |
1069 | 1071 |
1070 } // namespace webrtc | 1072 } // namespace webrtc |
OLD | NEW |