1 /*
2 * Copyright (C) 2019 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 * http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17 #include <webrtc/OpusPacketizer.h>
18
19 #include "Utils.h"
20
21 #include <webrtc/RTPSocketHandler.h>
22
23 #include <https/SafeCallbackable.h>
24
25 using namespace android;
26
OpusPacketizer(std::shared_ptr<RunLoop> runLoop,std::shared_ptr<StreamingSource> audioSource)27 OpusPacketizer::OpusPacketizer(
28 std::shared_ptr<RunLoop> runLoop,
29 std::shared_ptr<StreamingSource> audioSource)
30 : Packetizer(runLoop, audioSource),
31 mFirstInTalkspurt(true) {
32 }
33
packetize(const std::shared_ptr<SBuffer> & accessUnit,int64_t timeUs)34 void OpusPacketizer::packetize(const std::shared_ptr<SBuffer> &accessUnit, int64_t timeUs) {
35 LOG(VERBOSE) << "Received Opus frame of size " << accessUnit->size();
36
37 static constexpr uint8_t PT = 98;
38 static constexpr uint32_t SSRC = 0x8badf00d;
39
40 // XXX Retransmission packets add 2 bytes (for the original seqNum), should
41 // probably reserve that amount in the original packets so we don't exceed
42 // the MTU on retransmission.
43 static const size_t kMaxSRTPPayloadSize =
44 RTPSocketHandler::kMaxUDPPayloadSize - SRTP_MAX_TRAILER_LEN;
45
46 const uint8_t *audioData = accessUnit->data();
47 size_t size = accessUnit->size();
48
49 uint32_t rtpTime = ((timeUs - mediaStartTime()) * 48) / 1000;
50
51 CHECK_LE(12 + size, kMaxSRTPPayloadSize);
52
53 std::vector<uint8_t> packet(12 + size);
54 uint8_t *data = packet.data();
55
56 packet[0] = 0x80;
57 packet[1] = PT;
58
59 if (mFirstInTalkspurt) {
60 packet[1] |= 0x80; // (M)ark
61 mFirstInTalkspurt = false;
62 }
63
64 SET_U16(&data[2], 0); // seqNum
65 SET_U32(&data[4], rtpTime);
66 SET_U32(&data[8], SSRC);
67
68 memcpy(&data[12], audioData, size);
69
70 queueRTPDatagram(&packet);
71 }
72
rtpNow() const73 uint32_t OpusPacketizer::rtpNow() const {
74 return (timeSinceStart() * 48) / 1000;
75 }
76