| /* |
| * Copyright 2025 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 PC_CODEC_VENDOR_H_ |
| #define PC_CODEC_VENDOR_H_ |
| |
| #include <utility> |
| #include <vector> |
| |
| #include "api/field_trials_view.h" |
| #include "api/rtc_error.h" |
| #include "api/rtp_transceiver_direction.h" |
| #include "call/payload_type.h" |
| #include "media/base/codec.h" |
| #include "media/base/codec_list.h" |
| #include "media/base/media_engine.h" |
| #include "pc/media_options.h" |
| #include "pc/session_description.h" |
| #include "pc/typed_codec_vendor.h" |
| |
| namespace webrtc { |
| |
| // This class contains the functions required to compute the list of codecs |
| // for SDP offer/answer. It is exposed to MediaSessionDescriptionFactory |
| // for the construction of offers and answers. |
| |
| // TODO: bugs.webrtc.org/360058654 - complete the architectural changes |
| // The list of things to be done: |
| // - Make as much as possible private. |
| // - Split object usage into four objects: sender/receiver/audio/video. |
| // - Remove audio/video from the call names, merge code where possible. |
| // - Make the class instances owned by transceivers, so that codec |
| // lists can differ per transceiver. |
| // For cleanliness: |
| // - Thread guard |
| class CodecVendor { |
| public: |
| CodecVendor(MediaEngineInterface* media_engine, |
| bool rtx_enabled, |
| const FieldTrialsView& trials); |
| |
| public: |
| RTCErrorOr<std::vector<Codec>> GetNegotiatedCodecsForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| PayloadTypeSuggester& pt_suggester); |
| |
| RTCErrorOr<Codecs> GetNegotiatedCodecsForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| RtpTransceiverDirection offer_rtd, |
| RtpTransceiverDirection answer_rtd, |
| const ContentInfo* current_content, |
| std::vector<Codec> codecs_from_offer, |
| PayloadTypeSuggester& pt_suggester); |
| |
| // Function exposed for issues.webrtc.org/412904801 |
| // Modify the video codecs to return on subsequent GetNegotiated* calls. |
| // The input is a vector of pairs of codecs. |
| // For each pair, the first element is the codec to be replaced, |
| // and the second element is the codec to replace it with. |
| void ModifyVideoCodecs(std::vector<std::pair<Codec, Codec>> changes); |
| // Functions exposed for testing |
| void set_audio_codecs(const CodecList& send_codecs, |
| const CodecList& recv_codecs); |
| void set_audio_codecs(const std::vector<Codec>& send_codecs, |
| const std::vector<Codec>& recv_codecs) { |
| set_audio_codecs(CodecList::CreateFromTrustedData(send_codecs), |
| CodecList::CreateFromTrustedData(recv_codecs)); |
| } |
| void set_video_codecs(const CodecList& send_codecs, |
| const CodecList& recv_codecs); |
| void set_video_codecs(const std::vector<Codec>& send_codecs, |
| const std::vector<Codec>& recv_codecs) { |
| set_video_codecs(CodecList::CreateFromTrustedData(send_codecs), |
| CodecList::CreateFromTrustedData(recv_codecs)); |
| } |
| CodecList audio_sendrecv_codecs() const; |
| const CodecList& audio_send_codecs() const; |
| const CodecList& audio_recv_codecs() const; |
| CodecList video_sendrecv_codecs() const; |
| const CodecList& video_send_codecs() const; |
| const CodecList& video_recv_codecs() const; |
| |
| private: |
| CodecList GetAudioCodecsForOffer( |
| const RtpTransceiverDirection& direction) const; |
| CodecList GetAudioCodecsForAnswer( |
| const RtpTransceiverDirection& offer, |
| const RtpTransceiverDirection& answer) const; |
| CodecList GetVideoCodecsForOffer( |
| const RtpTransceiverDirection& direction) const; |
| CodecList GetVideoCodecsForAnswer( |
| const RtpTransceiverDirection& offer, |
| const RtpTransceiverDirection& answer) const; |
| |
| TypedCodecVendor audio_send_codecs_; |
| TypedCodecVendor audio_recv_codecs_; |
| |
| TypedCodecVendor video_send_codecs_; |
| TypedCodecVendor video_recv_codecs_; |
| }; |
| |
| // A class to assist in looking up data for a codec mapping. |
| // Pure virtual to allow implementations that depend on things that |
| // codec_vendor.h should not depend on. |
| // Pointers returned are not stable, and should not be stored. |
| class CodecLookupHelper { |
| public: |
| virtual ~CodecLookupHelper() = default; |
| virtual PayloadTypeSuggester* PayloadTypeSuggester() = 0; |
| // Look up the codec vendor to use, depending on context. |
| // This call may get additional arguments in the future, to aid |
| // in selection of the correct context. |
| virtual CodecVendor* GetCodecVendor() = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| // Re-export symbols from the webrtc namespace for backwards compatibility. |
| // TODO(bugs.webrtc.org/4222596): Remove once all references are updated. |
| #ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES |
| namespace cricket { |
| using ::webrtc::CodecLookupHelper; |
| using ::webrtc::CodecVendor; |
| } // namespace cricket |
| #endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES |
| |
| #endif // PC_CODEC_VENDOR_H_ |