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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 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 |
122 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec, | 148 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec, |
123 bool* created_new_payload) { | 149 bool* created_new_payload) { |
124 rtc::CritScope cs(&crit_sect_); | 150 rtc::CritScope cs(&crit_sect_); |
125 | 151 |
126 #if RTC_DCHECK_IS_ON | 152 #if RTC_DCHECK_IS_ON |
127 RTC_DCHECK(!used_for_video_); | 153 RTC_DCHECK(!used_for_video_); |
128 used_for_audio_ = true; | 154 used_for_audio_ = true; |
129 #endif | 155 #endif |
130 | 156 |
131 *created_new_payload = false; | 157 *created_new_payload = false; |
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396 const char* payload_name) const { | 422 const char* payload_name) const { |
397 rtc::CritScope cs(&crit_sect_); | 423 rtc::CritScope cs(&crit_sect_); |
398 for (const auto& it : payload_type_map_) { | 424 for (const auto& it : payload_type_map_) { |
399 if (_stricmp(it.second.name, payload_name) == 0) | 425 if (_stricmp(it.second.name, payload_name) == 0) |
400 return it.first; | 426 return it.first; |
401 } | 427 } |
402 return -1; | 428 return -1; |
403 } | 429 } |
404 | 430 |
405 } // namespace webrtc | 431 } // namespace webrtc |
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