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882b32d00f
This reverts commite046787a5a. Reason for revert: Revised codec matching to fix issue. Changes also back out some changes that should not have been included (using PayloadTypePicker for codec list merging). Original change's description: > Revert "Use PayloadTypePicker for video PT assignment" > > This reverts commite5048949b0. > > Reason for revert: Broke internal tests. > > Original change's description: > > Use PayloadTypePicker for video PT assignment > > > > This includes changes that change the order of codecs. > > It is preparatory to doing late assignment of video PTs. > > > > Bug: webrtc:360058654 > > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400 > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#43489} > > Bug: webrtc:360058654 > Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43490} Bug: webrtc:360058654 Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43554}
63 lines
2.2 KiB
C++
63 lines
2.2 KiB
C++
/*
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* Copyright 2024 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 CALL_PAYLOAD_TYPE_H_
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#define CALL_PAYLOAD_TYPE_H_
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#include <cstdint>
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#include <string>
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#include "api/rtc_error.h"
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#include "media/base/codec.h"
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#include "rtc_base/strong_alias.h"
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namespace webrtc {
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class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
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public:
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// Non-explicit conversions from and to ints are to be deprecated and
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// removed once calling code is upgraded.
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PayloadType(uint8_t pt) { value_ = pt; } // NOLINT: explicit
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constexpr operator uint8_t() const& { return value_; } // NOLINT: Explicit
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static bool IsValid(PayloadType id, bool rtcp_mux) {
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// A payload type is a 7-bit value in the RTP header, so max = 127.
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// If RTCP multiplexing is used, the numbers from 64 to 95 are reserved
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// for RTCP packets.
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if (rtcp_mux && (id > 63 && id < 96)) {
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return false;
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}
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return id >= 0 && id <= 127;
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}
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template <typename Sink>
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friend void AbslStringify(Sink& sink, const PayloadType pt) {
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absl::Format(&sink, "%d", pt.value_);
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}
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};
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class PayloadTypeSuggester {
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public:
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virtual ~PayloadTypeSuggester() = default;
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// Suggest a payload type for a given codec on a given media section.
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// Media section is indicated by MID.
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// The function will either return a PT already in use on the connection
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// or a newly suggested one.
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virtual RTCErrorOr<PayloadType> SuggestPayloadType(const std::string& mid,
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cricket::Codec codec) = 0;
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// Register a payload type as mapped to a specific codec for this MID
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// at this time.
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virtual RTCError AddLocalMapping(const std::string& mid,
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PayloadType payload_type,
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const cricket::Codec& codec) = 0;
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};
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} // namespace webrtc
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#endif // CALL_PAYLOAD_TYPE_H_
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