Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(471)

Side by Side Diff: webrtc/test/layer_filtering_transport.cc

Issue 1937693002: Replace scoped_ptr with unique_ptr everywhere (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@unique5
Patch Set: Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc ('k') | webrtc/test/rtp_file_reader.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 10
11 #include <memory>
12
11 #include "webrtc/base/checks.h" 13 #include "webrtc/base/checks.h"
12 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 14 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
13 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 15 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
14 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" 16 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
15 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 17 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
16 #include "webrtc/test/layer_filtering_transport.h" 18 #include "webrtc/test/layer_filtering_transport.h"
17 19
18 namespace webrtc { 20 namespace webrtc {
19 namespace test { 21 namespace test {
20 22
(...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after
55 57
56 if (header.payloadType == vp8_video_payload_type_ || 58 if (header.payloadType == vp8_video_payload_type_ ||
57 header.payloadType == vp9_video_payload_type_) { 59 header.payloadType == vp9_video_payload_type_) {
58 const uint8_t* payload = packet + header.headerLength; 60 const uint8_t* payload = packet + header.headerLength;
59 RTC_DCHECK_GT(length, header.headerLength); 61 RTC_DCHECK_GT(length, header.headerLength);
60 const size_t payload_length = length - header.headerLength; 62 const size_t payload_length = length - header.headerLength;
61 RTC_DCHECK_GT(payload_length, header.paddingLength); 63 RTC_DCHECK_GT(payload_length, header.paddingLength);
62 const size_t payload_data_length = payload_length - header.paddingLength; 64 const size_t payload_data_length = payload_length - header.paddingLength;
63 65
64 const bool is_vp8 = header.payloadType == vp8_video_payload_type_; 66 const bool is_vp8 = header.payloadType == vp8_video_payload_type_;
65 rtc::scoped_ptr<RtpDepacketizer> depacketizer( 67 std::unique_ptr<RtpDepacketizer> depacketizer(
66 RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9)); 68 RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
67 RtpDepacketizer::ParsedPayload parsed_payload; 69 RtpDepacketizer::ParsedPayload parsed_payload;
68 if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) { 70 if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) {
69 const int temporal_idx = static_cast<int>( 71 const int temporal_idx = static_cast<int>(
70 is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx 72 is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
71 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx); 73 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx);
72 const int spatial_idx = static_cast<int>( 74 const int spatial_idx = static_cast<int>(
73 is_vp8 ? kNoSpatialIdx 75 is_vp8 ? kNoSpatialIdx
74 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx); 76 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx);
75 if (selected_sl_ >= 0 && spatial_idx == selected_sl_ && 77 if (selected_sl_ >= 0 && spatial_idx == selected_sl_ &&
(...skipping 20 matching lines...) Expand all
96 // make sure the marker bit is set properly, and that sequence numbers are 98 // make sure the marker bit is set properly, and that sequence numbers are
97 // continuous. 99 // continuous.
98 if (set_marker_bit) 100 if (set_marker_bit)
99 temp_buffer[1] |= kRtpMarkerBitMask; 101 temp_buffer[1] |= kRtpMarkerBitMask;
100 102
101 return test::DirectTransport::SendRtp(temp_buffer, length, options); 103 return test::DirectTransport::SendRtp(temp_buffer, length, options);
102 } 104 }
103 105
104 } // namespace test 106 } // namespace test
105 } // namespace webrtc 107 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc ('k') | webrtc/test/rtp_file_reader.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698