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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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/modules/rtp_rtcp/include/rtp_payload_registry.h" | 11 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 | 14 |
15 #include "webrtc/base/checks.h" | 15 #include "webrtc/base/checks.h" |
16 #include "webrtc/base/logging.h" | 16 #include "webrtc/base/logging.h" |
17 #include "webrtc/base/stringutils.h" | 17 #include "webrtc/base/stringutils.h" |
18 #include "webrtc/common_types.h" | 18 #include "webrtc/common_types.h" |
19 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" | |
19 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 20 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
20 | 21 |
21 namespace webrtc { | 22 namespace webrtc { |
22 | 23 |
23 namespace { | 24 namespace { |
24 | 25 |
25 bool PayloadIsCompatible(const RtpUtility::Payload& payload, | 26 bool PayloadIsCompatible(const RtpUtility::Payload& payload, |
26 const CodecInst& audio_codec) { | 27 const CodecInst& audio_codec) { |
27 if (!payload.audio) | 28 if (!payload.audio) |
28 return false; | 29 return false; |
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112 | 113 |
113 RTPPayloadRegistry::RTPPayloadRegistry() | 114 RTPPayloadRegistry::RTPPayloadRegistry() |
114 : incoming_payload_type_(-1), | 115 : incoming_payload_type_(-1), |
115 last_received_payload_type_(-1), | 116 last_received_payload_type_(-1), |
116 last_received_media_payload_type_(-1), | 117 last_received_media_payload_type_(-1), |
117 rtx_(false), | 118 rtx_(false), |
118 ssrc_rtx_(0) {} | 119 ssrc_rtx_(0) {} |
119 | 120 |
120 RTPPayloadRegistry::~RTPPayloadRegistry() = default; | 121 RTPPayloadRegistry::~RTPPayloadRegistry() = default; |
121 | 122 |
123 bool RTPPayloadRegistry::SetAudioReceivePayloads( | |
the sun
2017/03/14 20:07:01
Like with DecoderDatabase, we should be able to tr
kwiberg-webrtc
2017/03/16 10:24:12
Done.
| |
124 std::map<int, SdpAudioFormat> codecs) { | |
125 for (const auto& kv : codecs) { | |
126 if (!IsPayloadTypeValid(kv.first)) { | |
127 return false; | |
128 } | |
129 } | |
130 | |
131 // Make sure we have no collisions. | |
132 auto existing_it = payload_type_map_.cbegin(); | |
133 auto new_it = codecs.cbegin(); | |
134 while (existing_it != payload_type_map_.cend() && new_it != codecs.cend()) { | |
135 if (existing_it->second.audio) { | |
136 ++existing_it; | |
137 } else { | |
138 if (existing_it->first == new_it->first) { | |
139 return false; // Trying to overwrite a video PT with audio. | |
the sun
2017/03/14 20:07:01
This should never happen, since we have an RTPPayl
kwiberg-webrtc
2017/03/16 10:24:12
Acknowledged. I'll try adding DCHECKs that verify
| |
140 } | |
141 if (existing_it->first < new_it->first) { | |
142 ++existing_it; | |
143 } else { | |
144 ++new_it; | |
145 } | |
146 } | |
147 } | |
148 | |
149 // Remove existing audio payload types. | |
the sun
2017/03/14 20:07:01
This shouldn't be allowed.
kwiberg-webrtc
2017/03/16 10:24:12
In principle, yes. See other comments...
| |
150 for (auto it = payload_type_map_.begin(); it != payload_type_map_.end();) { | |
151 if (it->second.audio) { | |
152 it = payload_type_map_.erase(it); | |
153 } else { | |
154 ++it; | |
155 } | |
156 } | |
157 | |
158 // Insert the new payload types. | |
159 for (const auto& kv : codecs) { | |
160 const int& rtp_payload_type = kv.first; | |
161 const SdpAudioFormat& audio_format = kv.second; | |
the sun
2017/03/14 20:07:01
I suggest you do this conversion in voe::Channel,
kwiberg-webrtc
2017/03/16 10:24:12
Well, yes... but I'm trying to reduce the use of C
| |
162 const CodecInst ci = SdpToCodecInst(rtp_payload_type, audio_format); | |
163 payload_type_map_[rtp_payload_type] = CreatePayloadType(ci); | |
164 } | |
165 | |
166 // Clear the value of last received payload type since it might mean | |
167 // something else now. | |
168 last_received_payload_type_ = -1; | |
169 last_received_media_payload_type_ = -1; | |
170 return true; | |
the sun
2017/03/14 20:07:01
If you address the above I don't think we even nee
kwiberg-webrtc
2017/03/16 10:24:12
Acknowledged.
| |
171 } | |
172 | |
122 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec, | 173 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec, |
123 bool* created_new_payload) { | 174 bool* created_new_payload) { |
124 *created_new_payload = false; | 175 *created_new_payload = false; |
125 if (!IsPayloadTypeValid(audio_codec.pltype)) | 176 if (!IsPayloadTypeValid(audio_codec.pltype)) |
126 return -1; | 177 return -1; |
127 | 178 |
128 rtc::CritScope cs(&crit_sect_); | 179 rtc::CritScope cs(&crit_sect_); |
129 | 180 |
130 auto it = payload_type_map_.find(audio_codec.pltype); | 181 auto it = payload_type_map_.find(audio_codec.pltype); |
131 if (it != payload_type_map_.end()) { | 182 if (it != payload_type_map_.end()) { |
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386 const char* payload_name) const { | 437 const char* payload_name) const { |
387 rtc::CritScope cs(&crit_sect_); | 438 rtc::CritScope cs(&crit_sect_); |
388 for (const auto& it : payload_type_map_) { | 439 for (const auto& it : payload_type_map_) { |
389 if (_stricmp(it.second.name, payload_name) == 0) | 440 if (_stricmp(it.second.name, payload_name) == 0) |
390 return it.first; | 441 return it.first; |
391 } | 442 } |
392 return -1; | 443 return -1; |
393 } | 444 } |
394 | 445 |
395 } // namespace webrtc | 446 } // namespace webrtc |
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