| OLD | NEW |
| 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" |
| 16 #include "webrtc/base/logging.h" |
| 17 #include "webrtc/base/stringutils.h" |
| 15 #include "webrtc/common_types.h" | 18 #include "webrtc/common_types.h" |
| 16 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" | 19 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" |
| 17 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 20 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| 18 #include "webrtc/rtc_base/checks.h" | |
| 19 #include "webrtc/rtc_base/logging.h" | |
| 20 #include "webrtc/rtc_base/stringutils.h" | |
| 21 | 21 |
| 22 namespace webrtc { | 22 namespace webrtc { |
| 23 | 23 |
| 24 namespace { | 24 namespace { |
| 25 | 25 |
| 26 bool PayloadIsCompatible(const RtpUtility::Payload& payload, | 26 bool PayloadIsCompatible(const RtpUtility::Payload& payload, |
| 27 const CodecInst& audio_codec) { | 27 const CodecInst& audio_codec) { |
| 28 if (!payload.audio) | 28 if (!payload.audio) |
| 29 return false; | 29 return false; |
| 30 if (_stricmp(payload.name, audio_codec.plname) != 0) | 30 if (_stricmp(payload.name, audio_codec.plname) != 0) |
| (...skipping 391 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 422 const char* payload_name) const { | 422 const char* payload_name) const { |
| 423 rtc::CritScope cs(&crit_sect_); | 423 rtc::CritScope cs(&crit_sect_); |
| 424 for (const auto& it : payload_type_map_) { | 424 for (const auto& it : payload_type_map_) { |
| 425 if (_stricmp(it.second.name, payload_name) == 0) | 425 if (_stricmp(it.second.name, payload_name) == 0) |
| 426 return it.first; | 426 return it.first; |
| 427 } | 427 } |
| 428 return -1; | 428 return -1; |
| 429 } | 429 } |
| 430 | 430 |
| 431 } // namespace webrtc | 431 } // namespace webrtc |
| OLD | NEW |