| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ |
| #define MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ |
| |
| #include <map> |
| #include <set> |
| |
| #include "api/audio_codecs/audio_format.h" |
| #include "api/optional.h" |
| #include "modules/rtp_rtcp/source/rtp_utility.h" |
| #include "rtc_base/criticalsection.h" |
| |
| namespace webrtc { |
| |
| class VideoCodec; |
| |
| class RTPPayloadRegistry { |
| public: |
| RTPPayloadRegistry(); |
| ~RTPPayloadRegistry(); |
| |
| // TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class |
| // and simplify the code. http://crbug/webrtc/6743. |
| |
| // Replace all audio receive payload types with the given map. |
| void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs); |
| |
| int32_t RegisterReceivePayload(int payload_type, |
| const SdpAudioFormat& audio_format, |
| bool* created_new_payload_type); |
| int32_t RegisterReceivePayload(const VideoCodec& video_codec); |
| |
| int32_t DeRegisterReceivePayload(int8_t payload_type); |
| |
| int32_t ReceivePayloadType(const SdpAudioFormat& audio_format, |
| int8_t* payload_type) const; |
| int32_t ReceivePayloadType(const VideoCodec& video_codec, |
| int8_t* payload_type) const; |
| |
| int GetPayloadTypeFrequency(uint8_t payload_type) const; |
| |
| rtc::Optional<RtpUtility::Payload> PayloadTypeToPayload( |
| uint8_t payload_type) const; |
| |
| void ResetLastReceivedPayloadTypes() { |
| rtc::CritScope cs(&crit_sect_); |
| last_received_payload_type_ = -1; |
| } |
| |
| int8_t last_received_payload_type() const { |
| rtc::CritScope cs(&crit_sect_); |
| return last_received_payload_type_; |
| } |
| void set_last_received_payload_type(int8_t last_received_payload_type) { |
| rtc::CritScope cs(&crit_sect_); |
| last_received_payload_type_ = last_received_payload_type; |
| } |
| |
| private: |
| // Prunes the payload type map of the specific payload type, if it exists. |
| void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( |
| const SdpAudioFormat& audio_format); |
| |
| rtc::CriticalSection crit_sect_; |
| std::map<int, RtpUtility::Payload> payload_type_map_; |
| int8_t last_received_payload_type_; |
| |
| // As a first step in splitting this class up in separate cases for audio and |
| // video, DCHECK that no instance is used for both audio and video. |
| #if RTC_DCHECK_IS_ON |
| bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false; |
| bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false; |
| #endif |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ |