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 | 10 |
11 #include <memory> | 11 #include <memory> |
12 | 12 |
13 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 13 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
14 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 14 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
15 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" | 15 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" |
16 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 16 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
17 #include "webrtc/rtc_base/checks.h" | 17 #include "webrtc/rtc_base/checks.h" |
18 #include "webrtc/test/layer_filtering_transport.h" | 18 #include "webrtc/test/layer_filtering_transport.h" |
19 #include "webrtc/test/rtp_file_writer.h" | |
19 | 20 |
20 namespace webrtc { | 21 namespace webrtc { |
21 namespace test { | 22 namespace test { |
22 | 23 |
23 LayerFilteringTransport::LayerFilteringTransport( | 24 LayerFilteringTransport::LayerFilteringTransport( |
24 SingleThreadedTaskQueueForTesting* task_queue, | 25 SingleThreadedTaskQueueForTesting* task_queue, |
25 const FakeNetworkPipe::Config& config, | 26 const FakeNetworkPipe::Config& config, |
26 Call* send_call, | 27 Call* send_call, |
27 uint8_t vp8_video_payload_type, | 28 uint8_t vp8_video_payload_type, |
28 uint8_t vp9_video_payload_type, | 29 uint8_t vp9_video_payload_type, |
29 int selected_tl, | 30 int selected_tl, |
30 int selected_sl, | 31 int selected_sl, |
31 const std::map<uint8_t, MediaType>& payload_type_map) | 32 const std::map<uint8_t, MediaType>& payload_type_map, |
33 std::unique_ptr<test::RtpFileWriter> rtp_file_writer) | |
32 : DirectTransport(task_queue, config, send_call, payload_type_map), | 34 : DirectTransport(task_queue, config, send_call, payload_type_map), |
33 vp8_video_payload_type_(vp8_video_payload_type), | 35 vp8_video_payload_type_(vp8_video_payload_type), |
34 vp9_video_payload_type_(vp9_video_payload_type), | 36 vp9_video_payload_type_(vp9_video_payload_type), |
35 selected_tl_(selected_tl), | 37 selected_tl_(selected_tl), |
36 selected_sl_(selected_sl), | 38 selected_sl_(selected_sl), |
37 discarded_last_packet_(false) {} | 39 discarded_last_packet_(false), |
40 receiver_(nullptr), | |
41 start_ms_(rtc::TimeMillis()), | |
42 rtp_file_writer_(std::move(rtp_file_writer)) { | |
43 DirectTransport::SetReceiver(this); | |
44 } | |
38 | 45 |
39 bool LayerFilteringTransport::DiscardedLastPacket() const { | 46 bool LayerFilteringTransport::DiscardedLastPacket() const { |
40 return discarded_last_packet_; | 47 return discarded_last_packet_; |
41 } | 48 } |
42 | 49 |
43 bool LayerFilteringTransport::SendRtp(const uint8_t* packet, | 50 bool LayerFilteringTransport::SendRtp(const uint8_t* packet, |
44 size_t length, | 51 size_t length, |
45 const PacketOptions& options) { | 52 const PacketOptions& options) { |
53 | |
46 if (selected_tl_ == -1 && selected_sl_ == -1) { | 54 if (selected_tl_ == -1 && selected_sl_ == -1) { |
47 // Nothing to change, forward the packet immediately. | 55 // Nothing to change, forward the packet immediately. |
48 return test::DirectTransport::SendRtp(packet, length, options); | 56 return test::DirectTransport::SendRtp(packet, length, options); |
49 } | 57 } |
50 | 58 |
51 bool set_marker_bit = false; | 59 bool set_marker_bit = false; |
52 RtpUtility::RtpHeaderParser parser(packet, length); | 60 RtpUtility::RtpHeaderParser parser(packet, length); |
53 RTPHeader header; | 61 RTPHeader header; |
54 parser.Parse(&header); | 62 parser.Parse(&header); |
55 | |
56 RTC_DCHECK_LE(length, IP_PACKET_SIZE); | 63 RTC_DCHECK_LE(length, IP_PACKET_SIZE); |
57 uint8_t temp_buffer[IP_PACKET_SIZE]; | 64 uint8_t temp_buffer[IP_PACKET_SIZE]; |
58 memcpy(temp_buffer, packet, length); | 65 memcpy(temp_buffer, packet, length); |
59 | 66 |
60 if (header.payloadType == vp8_video_payload_type_ || | 67 if (header.payloadType == vp8_video_payload_type_ || |
61 header.payloadType == vp9_video_payload_type_) { | 68 header.payloadType == vp9_video_payload_type_) { |
62 const uint8_t* payload = packet + header.headerLength; | 69 const uint8_t* payload = packet + header.headerLength; |
63 RTC_DCHECK_GT(length, header.headerLength); | 70 RTC_DCHECK_GT(length, header.headerLength); |
64 const size_t payload_length = length - header.headerLength; | 71 const size_t payload_length = length - header.headerLength; |
65 RTC_DCHECK_GT(payload_length, header.paddingLength); | 72 RTC_DCHECK_GT(payload_length, header.paddingLength); |
(...skipping 32 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
98 | 105 |
99 // We are discarding some of the packets (specifically, whole layers), so | 106 // We are discarding some of the packets (specifically, whole layers), so |
100 // make sure the marker bit is set properly, and that sequence numbers are | 107 // make sure the marker bit is set properly, and that sequence numbers are |
101 // continuous. | 108 // continuous. |
102 if (set_marker_bit) | 109 if (set_marker_bit) |
103 temp_buffer[1] |= kRtpMarkerBitMask; | 110 temp_buffer[1] |= kRtpMarkerBitMask; |
104 | 111 |
105 return test::DirectTransport::SendRtp(temp_buffer, length, options); | 112 return test::DirectTransport::SendRtp(temp_buffer, length, options); |
106 } | 113 } |
107 | 114 |
115 void LayerFilteringTransport::SetReceiver(PacketReceiver* receiver) { | |
116 receiver_ = receiver; | |
117 } | |
118 | |
119 PacketReceiver::DeliveryStatus LayerFilteringTransport::DeliverPacket( | |
stefan-webrtc
2017/08/24 12:52:18
Maybe we should make this a feature of DirectTrans
| |
120 MediaType media_type, | |
121 const uint8_t* packet, | |
122 size_t length, | |
123 const PacketTime& packet_time) { | |
124 if (rtp_file_writer_.get()) { | |
125 RtpPacket rtp_packet; | |
126 RTC_DCHECK_LE(length, IP_PACKET_SIZE); | |
127 memcpy(rtp_packet.data, packet, length); | |
128 rtp_packet.length = length; | |
129 rtp_packet.original_length = length; | |
130 rtp_packet.time_ms = rtc::TimeMillis() - start_ms_; | |
131 rtp_file_writer_->WritePacket(&rtp_packet); | |
132 } | |
133 | |
134 if (!receiver_) | |
135 return PacketReceiver::DELIVERY_PACKET_ERROR; | |
136 return receiver_->DeliverPacket(media_type, packet, length, packet_time); | |
137 } | |
138 | |
108 } // namespace test | 139 } // namespace test |
109 } // namespace webrtc | 140 } // namespace webrtc |
OLD | NEW |