665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
82 lines
2.7 KiB
C++
82 lines
2.7 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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#define MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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#include <map>
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#include <set>
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#include "absl/types/optional.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/video_codecs/video_codec.h"
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#include "modules/rtp_rtcp/source/rtp_utility.h"
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#include "rtc_base/criticalsection.h"
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namespace webrtc {
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class RTPPayloadRegistry {
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public:
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RTPPayloadRegistry();
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~RTPPayloadRegistry();
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// TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
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// and simplify the code. http://crbug/webrtc/6743.
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// Replace all audio receive payload types with the given map.
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void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
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int32_t RegisterReceivePayload(int payload_type,
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const SdpAudioFormat& audio_format,
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bool* created_new_payload_type);
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int32_t RegisterReceivePayload(const VideoCodec& video_codec);
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int32_t DeRegisterReceivePayload(int8_t payload_type);
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int GetPayloadTypeFrequency(uint8_t payload_type) const;
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absl::optional<RtpUtility::Payload> PayloadTypeToPayload(
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uint8_t payload_type) const;
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void ResetLastReceivedPayloadTypes() {
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rtc::CritScope cs(&crit_sect_);
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last_received_payload_type_ = -1;
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}
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int8_t last_received_payload_type() const {
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rtc::CritScope cs(&crit_sect_);
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return last_received_payload_type_;
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}
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void set_last_received_payload_type(int8_t last_received_payload_type) {
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rtc::CritScope cs(&crit_sect_);
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last_received_payload_type_ = last_received_payload_type;
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}
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private:
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// Prunes the payload type map of the specific payload type, if it exists.
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void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
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const SdpAudioFormat& audio_format);
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rtc::CriticalSection crit_sect_;
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std::map<int, RtpUtility::Payload> payload_type_map_;
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int8_t last_received_payload_type_;
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// As a first step in splitting this class up in separate cases for audio and
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// video, DCHECK that no instance is used for both audio and video.
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#if RTC_DCHECK_IS_ON
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bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false;
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bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false;
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#endif
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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