(3) Rename files to snake_case: move the files
Mechanically generated with this command: tools_webrtc/do-rename.sh move all-renames.txt Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690 Reviewed-on: https://webrtc-review.googlesource.com/c/115481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26225}
This commit is contained in:
Steve Anton
@ -1,167 +0,0 @@
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/*
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* Copyright 2015 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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// This file contains interfaces for RtpReceivers
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// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
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#ifndef API_RTPRECEIVERINTERFACE_H_
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#define API_RTPRECEIVERINTERFACE_H_
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#include <string>
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#include <vector>
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#include "api/crypto/framedecryptorinterface.h"
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#include "api/mediastreaminterface.h"
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#include "api/mediatypes.h"
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#include "api/proxy.h"
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#include "api/rtpparameters.h"
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#include "rtc_base/refcount.h"
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#include "rtc_base/scoped_ref_ptr.h"
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namespace webrtc {
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enum class RtpSourceType {
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SSRC,
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CSRC,
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};
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class RtpSource {
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public:
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RtpSource() = delete;
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RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type);
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RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type,
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uint8_t audio_level);
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RtpSource(const RtpSource&);
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RtpSource& operator=(const RtpSource&);
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~RtpSource();
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int64_t timestamp_ms() const { return timestamp_ms_; }
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void update_timestamp_ms(int64_t timestamp_ms) {
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RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
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timestamp_ms_ = timestamp_ms;
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}
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// The identifier of the source can be the CSRC or the SSRC.
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uint32_t source_id() const { return source_id_; }
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// The source can be either a contributing source or a synchronization source.
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RtpSourceType source_type() const { return source_type_; }
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absl::optional<uint8_t> audio_level() const { return audio_level_; }
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void set_audio_level(const absl::optional<uint8_t>& level) {
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audio_level_ = level;
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}
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bool operator==(const RtpSource& o) const {
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return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
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source_type_ == o.source_type() && audio_level_ == o.audio_level_;
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}
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private:
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int64_t timestamp_ms_;
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uint32_t source_id_;
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RtpSourceType source_type_;
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absl::optional<uint8_t> audio_level_;
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};
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class RtpReceiverObserverInterface {
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public:
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// Note: Currently if there are multiple RtpReceivers of the same media type,
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// they will all call OnFirstPacketReceived at once.
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//
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// In the future, it's likely that an RtpReceiver will only call
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// OnFirstPacketReceived when a packet is received specifically for its
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// SSRC/mid.
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virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
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protected:
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virtual ~RtpReceiverObserverInterface() {}
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};
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class RtpReceiverInterface : public rtc::RefCountInterface {
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public:
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
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// The list of streams that |track| is associated with. This is the same as
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// the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
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// https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
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// TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
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// TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of
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// stream_ids() as soon as downstream projects are no longer dependent on
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// stream objects.
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virtual std::vector<std::string> stream_ids() const;
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virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
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// Audio or video receiver?
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virtual cricket::MediaType media_type() const = 0;
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// Not to be confused with "mid", this is a field we can temporarily use
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// to uniquely identify a receiver until we implement Unified Plan SDP.
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virtual std::string id() const = 0;
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// The WebRTC specification only defines RTCRtpParameters in terms of senders,
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// but this API also applies them to receivers, similar to ORTC:
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// http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
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virtual RtpParameters GetParameters() const = 0;
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// Currently, doesn't support changing any parameters, but may in the future.
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virtual bool SetParameters(const RtpParameters& parameters) = 0;
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// Does not take ownership of observer.
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// Must call SetObserver(nullptr) before the observer is destroyed.
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virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
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// TODO(zhihuang): Remove the default implementation once the subclasses
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// implement this. Currently, the only relevant subclass is the
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// content::FakeRtpReceiver in Chromium.
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virtual std::vector<RtpSource> GetSources() const;
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// Sets a user defined frame decryptor that will decrypt the entire frame
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// before it is sent across the network. This will decrypt the entire frame
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// using the user provided decryption mechanism regardless of whether SRTP is
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// enabled or not.
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virtual void SetFrameDecryptor(
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
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// Returns a pointer to the frame decryptor set previously by the
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// user. This can be used to update the state of the object.
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virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const;
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protected:
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~RtpReceiverInterface() override = default;
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};
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// Define proxy for RtpReceiverInterface.
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// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
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// are called on is an implementation detail.
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BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
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PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
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streams)
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PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
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PROXY_CONSTMETHOD0(std::string, id)
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PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
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PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
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PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
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PROXY_METHOD1(void,
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SetFrameDecryptor,
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rtc::scoped_refptr<FrameDecryptorInterface>);
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>,
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GetFrameDecryptor);
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END_PROXY_MAP()
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} // namespace webrtc
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#endif // API_RTPRECEIVERINTERFACE_H_
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