blob: 7d9870b8f1b9086e01b6096faac65de8b1d1a74e [file] [log] [blame]
* Copyright 2015 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.
// This file contains interfaces for RtpReceivers
#include <string>
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/proxy.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "rtc_base/ref_count.h"
namespace webrtc {
enum class RtpSourceType {
class RtpSource {
RtpSource() = delete;
RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type);
RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
uint8_t audio_level);
RtpSource(const RtpSource&);
RtpSource& operator=(const RtpSource&);
int64_t timestamp_ms() const { return timestamp_ms_; }
void update_timestamp_ms(int64_t timestamp_ms) {
RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
timestamp_ms_ = timestamp_ms;
// The identifier of the source can be the CSRC or the SSRC.
uint32_t source_id() const { return source_id_; }
// The source can be either a contributing source or a synchronization source.
RtpSourceType source_type() const { return source_type_; }
absl::optional<uint8_t> audio_level() const { return audio_level_; }
void set_audio_level(const absl::optional<uint8_t>& level) {
audio_level_ = level;
bool operator==(const RtpSource& o) const {
return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
source_type_ == o.source_type() && audio_level_ == o.audio_level_;
int64_t timestamp_ms_;
uint32_t source_id_;
RtpSourceType source_type_;
absl::optional<uint8_t> audio_level_;
class RtpReceiverObserverInterface {
// Note: Currently if there are multiple RtpReceivers of the same media type,
// they will all call OnFirstPacketReceived at once.
// In the future, it's likely that an RtpReceiver will only call
// OnFirstPacketReceived when a packet is received specifically for its
// SSRC/mid.
virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
virtual ~RtpReceiverObserverInterface() {}
class RtpReceiverInterface : public rtc::RefCountInterface {
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
// The dtlsTransport attribute exposes the DTLS transport on which the
// media is received. It may be null.
// TODO( remove default implementation
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
// The list of streams that |track| is associated with. This is the same as
// the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
// TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
// TODO( Remove streams() in favor of
// stream_ids() as soon as downstream projects are no longer dependent on
// stream objects.
virtual std::vector<std::string> stream_ids() const;
virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
// Audio or video receiver?
virtual cricket::MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
// The WebRTC specification only defines RTCRtpParameters in terms of senders,
// but this API also applies them to receivers, similar to ORTC:
virtual RtpParameters GetParameters() const = 0;
// Currently, doesn't support changing any parameters, but may in the future.
virtual bool SetParameters(const RtpParameters& parameters) = 0;
// Does not take ownership of observer.
// Must call SetObserver(nullptr) before the observer is destroyed.
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
// TODO(zhihuang): Remove the default implementation once the subclasses
// implement this. Currently, the only relevant subclass is the
// content::FakeRtpReceiver in Chromium.
virtual std::vector<RtpSource> GetSources() const;
// Sets a user defined frame decryptor that will decrypt the entire frame
// before it is sent across the network. This will decrypt the entire frame
// using the user provided decryption mechanism regardless of whether SRTP is
// enabled or not.
virtual void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
// Returns a pointer to the frame decryptor set previously by the
// user. This can be used to update the state of the object.
virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const;
~RtpReceiverInterface() override = default;
// Define proxy for RtpReceiverInterface.
// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
// are called on is an implementation detail.
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
} // namespace webrtc