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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" | |
20 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 19 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
21 | 20 |
22 namespace webrtc { | 21 namespace webrtc { |
23 | 22 |
24 namespace { | 23 namespace { |
25 | 24 |
26 bool PayloadIsCompatible(const RtpUtility::Payload& payload, | 25 bool PayloadIsCompatible(const RtpUtility::Payload& payload, |
27 const CodecInst& audio_codec) { | 26 const CodecInst& audio_codec) { |
28 if (!payload.audio) | 27 if (!payload.audio) |
29 return false; | 28 return false; |
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113 | 112 |
114 RTPPayloadRegistry::RTPPayloadRegistry() | 113 RTPPayloadRegistry::RTPPayloadRegistry() |
115 : incoming_payload_type_(-1), | 114 : incoming_payload_type_(-1), |
116 last_received_payload_type_(-1), | 115 last_received_payload_type_(-1), |
117 last_received_media_payload_type_(-1), | 116 last_received_media_payload_type_(-1), |
118 rtx_(false), | 117 rtx_(false), |
119 ssrc_rtx_(0) {} | 118 ssrc_rtx_(0) {} |
120 | 119 |
121 RTPPayloadRegistry::~RTPPayloadRegistry() = default; | 120 RTPPayloadRegistry::~RTPPayloadRegistry() = default; |
122 | 121 |
123 void RTPPayloadRegistry::SetAudioReceivePayloads( | |
124 std::map<int, SdpAudioFormat> codecs) { | |
125 rtc::CritScope cs(&crit_sect_); | |
126 | |
127 #if RTC_DCHECK_IS_ON | |
128 RTC_DCHECK(!used_for_video_); | |
129 used_for_audio_ = true; | |
130 #endif | |
131 | |
132 payload_type_map_.clear(); | |
133 for (const auto& kv : codecs) { | |
134 const int& rtp_payload_type = kv.first; | |
135 const SdpAudioFormat& audio_format = kv.second; | |
136 const CodecInst ci = SdpToCodecInst(rtp_payload_type, audio_format); | |
137 RTC_DCHECK(IsPayloadTypeValid(rtp_payload_type)); | |
138 payload_type_map_.insert( | |
139 std::make_pair(rtp_payload_type, CreatePayloadType(ci))); | |
140 } | |
141 | |
142 // Clear the value of last received payload type since it might mean | |
143 // something else now. | |
144 last_received_payload_type_ = -1; | |
145 last_received_media_payload_type_ = -1; | |
146 } | |
147 | |
148 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec, | 122 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec, |
149 bool* created_new_payload) { | 123 bool* created_new_payload) { |
150 rtc::CritScope cs(&crit_sect_); | 124 rtc::CritScope cs(&crit_sect_); |
151 | 125 |
152 #if RTC_DCHECK_IS_ON | 126 #if RTC_DCHECK_IS_ON |
153 RTC_DCHECK(!used_for_video_); | 127 RTC_DCHECK(!used_for_video_); |
154 used_for_audio_ = true; | 128 used_for_audio_ = true; |
155 #endif | 129 #endif |
156 | 130 |
157 *created_new_payload = false; | 131 *created_new_payload = false; |
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422 const char* payload_name) const { | 396 const char* payload_name) const { |
423 rtc::CritScope cs(&crit_sect_); | 397 rtc::CritScope cs(&crit_sect_); |
424 for (const auto& it : payload_type_map_) { | 398 for (const auto& it : payload_type_map_) { |
425 if (_stricmp(it.second.name, payload_name) == 0) | 399 if (_stricmp(it.second.name, payload_name) == 0) |
426 return it.first; | 400 return it.first; |
427 } | 401 } |
428 return -1; | 402 return -1; |
429 } | 403 } |
430 | 404 |
431 } // namespace webrtc | 405 } // namespace webrtc |
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