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/* Copyright 2018 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 is EXPERIMENTAL interface for media transport.
//
// The goal is to refactor WebRTC code so that audio and video frames
// are sent / received through the media transport interface. This will
// enable different media transport implementations, including QUIC-based
// media transport.
#ifndef API_MEDIA_TRANSPORT_INTERFACE_H_
#define API_MEDIA_TRANSPORT_INTERFACE_H_
#include <api/transport/network_control.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtcerror.h"
#include "api/video/encoded_image.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/networkroute.h"
namespace rtc {
class PacketTransportInternal;
class Thread;
} // namespace rtc
namespace webrtc {
// A collection of settings for creation of media transport.
struct MediaTransportSettings final {
MediaTransportSettings();
MediaTransportSettings(const MediaTransportSettings&);
MediaTransportSettings& operator=(const MediaTransportSettings&);
~MediaTransportSettings();
// Group calls are not currently supported, in 1:1 call one side must set
// is_caller = true and another is_caller = false.
bool is_caller;
// Must be set if a pre-shared key is used for the call.
// TODO(bugs.webrtc.org/9944): This should become zero buffer in the distant
// future.
absl::optional<std::string> pre_shared_key;
};
// Represents encoded audio frame in any encoding (type of encoding is opaque).
// To avoid copying of encoded data use move semantics when passing by value.
class MediaTransportEncodedAudioFrame final {
public:
enum class FrameType {
// Normal audio frame (equivalent to webrtc::kAudioFrameSpeech).
kSpeech,
// DTX frame (equivalent to webrtc::kAudioFrameCN).
// DTX frame (equivalent to webrtc::kAudioFrameCN).
kDiscontinuousTransmission,
// TODO(nisse): Mis-spelled version, update users, then delete.
kDiscountinuousTransmission = kDiscontinuousTransmission,
};
MediaTransportEncodedAudioFrame(
// Audio sampling rate, for example 48000.
int sampling_rate_hz,
// Starting sample index of the frame, i.e. how many audio samples were
// before this frame since the beginning of the call or beginning of time
// in one channel (the starting point should not matter for NetEq). In
// WebRTC it is used as a timestamp of the frame.
// TODO(sukhanov): Starting_sample_index is currently adjusted on the
// receiver side in RTP path. Non-RTP implementations should preserve it.
// For NetEq initial offset should not matter so we should consider fixing
// RTP path.
int starting_sample_index,
// Number of audio samples in audio frame in 1 channel.
int samples_per_channel,
// Sequence number of the frame in the order sent, it is currently
// required by NetEq, but we can fix NetEq, because starting_sample_index
// should be enough.
int sequence_number,
// If audio frame is a speech or discontinued transmission.
FrameType frame_type,
// Opaque payload type. In RTP codepath payload type is stored in RTP
// header. In other implementations it should be simply passed through the
// wire -- it's needed for decoder.
uint8_t payload_type,
// Vector with opaque encoded data.
std::vector<uint8_t> encoded_data);
~MediaTransportEncodedAudioFrame();
MediaTransportEncodedAudioFrame(const MediaTransportEncodedAudioFrame&);
MediaTransportEncodedAudioFrame& operator=(
const MediaTransportEncodedAudioFrame& other);
MediaTransportEncodedAudioFrame& operator=(
MediaTransportEncodedAudioFrame&& other);
MediaTransportEncodedAudioFrame(MediaTransportEncodedAudioFrame&&);
// Getters.
int sampling_rate_hz() const { return sampling_rate_hz_; }
int starting_sample_index() const { return starting_sample_index_; }
int samples_per_channel() const { return samples_per_channel_; }
int sequence_number() const { return sequence_number_; }
uint8_t payload_type() const { return payload_type_; }
FrameType frame_type() const { return frame_type_; }
rtc::ArrayView<const uint8_t> encoded_data() const { return encoded_data_; }
private:
int sampling_rate_hz_;
int starting_sample_index_;
int samples_per_channel_;
// TODO(sukhanov): Refactor NetEq so we don't need sequence number.
// Having sample_index and samples_per_channel should be enough.
int sequence_number_;
FrameType frame_type_;
// TODO(sukhanov): Consider enumerating allowed encodings and store enum
// instead of uint payload_type.
uint8_t payload_type_;
std::vector<uint8_t> encoded_data_;
};
// Callback to notify about network route changes.
class MediaTransportNetworkChangeCallback {
public:
virtual ~MediaTransportNetworkChangeCallback() = default;
// Called when the network route is changed, with the new network route.
virtual void OnNetworkRouteChanged(
const rtc::NetworkRoute& new_network_route) = 0;
};
// Interface for receiving encoded audio frames from MediaTransportInterface
// implementations.
class MediaTransportAudioSinkInterface {
public:
virtual ~MediaTransportAudioSinkInterface() = default;
// Called when new encoded audio frame is received.
virtual void OnData(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) = 0;
};
// Represents encoded video frame, along with the codec information.
class MediaTransportEncodedVideoFrame final {
public:
MediaTransportEncodedVideoFrame(int64_t frame_id,
std::vector<int64_t> referenced_frame_ids,
VideoCodecType codec_type,
const webrtc::EncodedImage& encoded_image);
~MediaTransportEncodedVideoFrame();
MediaTransportEncodedVideoFrame(const MediaTransportEncodedVideoFrame&);
MediaTransportEncodedVideoFrame& operator=(
const MediaTransportEncodedVideoFrame& other);
MediaTransportEncodedVideoFrame& operator=(
MediaTransportEncodedVideoFrame&& other);
MediaTransportEncodedVideoFrame(MediaTransportEncodedVideoFrame&&);
VideoCodecType codec_type() const { return codec_type_; }
const webrtc::EncodedImage& encoded_image() const { return encoded_image_; }
int64_t frame_id() const { return frame_id_; }
const std::vector<int64_t>& referenced_frame_ids() const {
return referenced_frame_ids_;
}
// Hack to workaround lack of ownership of the encoded_image_._buffer. If we
// don't already own the underlying data, make a copy.
void Retain();
private:
MediaTransportEncodedVideoFrame();
VideoCodecType codec_type_;
// The buffer is not owned by the encoded image. On the sender it means that
// it will need to make a copy using the Retain() method, if it wants to
// deliver it asynchronously.
webrtc::EncodedImage encoded_image_;
// If non-empty, this is the data for the encoded image.
std::vector<uint8_t> encoded_data_;
// Frame id uniquely identifies a frame in a stream. It needs to be unique in
// a given time window (i.e. technically unique identifier for the lifetime of
// the connection is not needed, but you need to guarantee that remote side
// got rid of the previous frame_id if you plan to reuse it).
//
// It is required by a remote jitter buffer, and is the same as
// EncodedFrame::id::picture_id.
//
// This data must be opaque to the media transport, and media transport should
// itself not make any assumptions about what it is and its uniqueness.
int64_t frame_id_;
// A single frame might depend on other frames. This is set of identifiers on
// which the current frame depends.
std::vector<int64_t> referenced_frame_ids_;
};
// Interface for receiving encoded video frames from MediaTransportInterface
// implementations.
class MediaTransportVideoSinkInterface {
public:
virtual ~MediaTransportVideoSinkInterface() = default;
// Called when new encoded video frame is received.
virtual void OnData(uint64_t channel_id,
MediaTransportEncodedVideoFrame frame) = 0;
// Called when the request for keyframe is received.
virtual void OnKeyFrameRequested(uint64_t channel_id) = 0;
};
// State of the media transport. Media transport begins in the pending state.
// It transitions to writable when it is ready to send media. It may transition
// back to pending if the connection is blocked. It may transition to closed at
// any time. Closed is terminal: a transport will never re-open once closed.
enum class MediaTransportState {
kPending,
kWritable,
kClosed,
};
// Callback invoked whenever the state of the media transport changes.
class MediaTransportStateCallback {
public:
virtual ~MediaTransportStateCallback() = default;
// Invoked whenever the state of the media transport changes.
virtual void OnStateChanged(MediaTransportState state) = 0;
};
// Supported types of application data messages.
enum class DataMessageType {
// Application data buffer with the binary bit unset.
kText,
// Application data buffer with the binary bit set.
kBinary,
// Transport-agnostic control messages, such as open or open-ack messages.
kControl,
};
// Parameters for sending data. The parameters may change from message to
// message, even within a single channel. For example, control messages may be
// sent reliably and in-order, even if the data channel is configured for
// unreliable delivery.
struct SendDataParams {
SendDataParams();
SendDataParams(const SendDataParams&);
DataMessageType type = DataMessageType::kText;
// Whether to deliver the message in order with respect to other ordered
// messages with the same channel_id.
bool ordered = false;
// If set, the maximum number of times this message may be
// retransmitted by the transport before it is dropped.
// Setting this value to zero disables retransmission.
// Must be non-negative. |max_rtx_count| and |max_rtx_ms| may not be set
// simultaneously.
absl::optional<int> max_rtx_count;
// If set, the maximum number of milliseconds for which the transport
// may retransmit this message before it is dropped.
// Setting this value to zero disables retransmission.
// Must be non-negative. |max_rtx_count| and |max_rtx_ms| may not be set
// simultaneously.
absl::optional<int> max_rtx_ms;
};
// Sink for callbacks related to a data channel.
class DataChannelSink {
public:
virtual ~DataChannelSink() = default;
// Callback issued when data is received by the transport.
virtual void OnDataReceived(int channel_id,
DataMessageType type,
const rtc::CopyOnWriteBuffer& buffer) = 0;
// Callback issued when a remote data channel begins the closing procedure.
// Messages sent after the closing procedure begins will not be transmitted.
virtual void OnChannelClosing(int channel_id) = 0;
// Callback issued when a (remote or local) data channel completes the closing
// procedure. Closing channels become closed after all pending data has been
// transmitted.
virtual void OnChannelClosed(int channel_id) = 0;
};
// Media transport interface for sending / receiving encoded audio/video frames
// and receiving bandwidth estimate update from congestion control.
class MediaTransportInterface {
public:
virtual ~MediaTransportInterface() = default;
// Start asynchronous send of audio frame. The status returned by this method
// only pertains to the synchronous operations (e.g.
// serialization/packetization), not to the asynchronous operation.
virtual RTCError SendAudioFrame(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) = 0;
// Start asynchronous send of video frame. The status returned by this method
// only pertains to the synchronous operations (e.g.
// serialization/packetization), not to the asynchronous operation.
virtual RTCError SendVideoFrame(
uint64_t channel_id,
const MediaTransportEncodedVideoFrame& frame) = 0;
// Requests a keyframe for the particular channel (stream). The caller should
// check that the keyframe is not present in a jitter buffer already (i.e.
// don't request a keyframe if there is one that you will get from the jitter
// buffer in a moment).
virtual RTCError RequestKeyFrame(uint64_t channel_id) = 0;
// Sets audio sink. Sink must be unset by calling SetReceiveAudioSink(nullptr)
// before the media transport is destroyed or before new sink is set.
virtual void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) = 0;
// Registers a video sink. Before destruction of media transport, you must
// pass a nullptr.
virtual void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) = 0;
// Adds a target bitrate observer. Before media transport is destructed
// the observer must be unregistered (by calling
// RemoveTargetTransferRateObserver).
// A newly registered observer will be called back with the latest recorded
// target rate, if available.
virtual void AddTargetTransferRateObserver(
webrtc::TargetTransferRateObserver* observer);
// Removes an existing |observer| from observers. If observer was never
// registered, an error is logged and method does nothing.
virtual void RemoveTargetTransferRateObserver(
webrtc::TargetTransferRateObserver* observer);
// Returns the last known target transfer rate as reported to the above
// observers.
virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate();
// Gets the audio packet overhead in bytes. Returned overhead does not include
// transport overhead (ipv4/6, turn channeldata, tcp/udp, etc.).
// If the transport is capable of fusing packets together, this overhead
// might not be a very accurate number.
virtual size_t GetAudioPacketOverhead() const;
// Sets an observer for network change events. If the network route is already
// established when the callback is set, |callback| will be called immediately
// with the current network route.
// Before media transport is destroyed, the callback must be unregistered by
// setting it to nullptr.
virtual void SetNetworkChangeCallback(
MediaTransportNetworkChangeCallback* callback);
// Sets a state observer callback. Before media transport is destroyed, the
// callback must be unregistered by setting it to nullptr.
// A newly registered callback will be called with the current state.
// Media transport does not invoke this callback concurrently.
virtual void SetMediaTransportStateCallback(
MediaTransportStateCallback* callback) = 0;
// Sends a data buffer to the remote endpoint using the given send parameters.
// |buffer| may not be larger than 256 KiB. Returns an error if the send
// fails.
virtual RTCError SendData(int channel_id,
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& buffer) = 0;
// Closes |channel_id| gracefully. Returns an error if |channel_id| is not
// open. Data sent after the closing procedure begins will not be
// transmitted. The channel becomes closed after pending data is transmitted.
virtual RTCError CloseChannel(int channel_id) = 0;
// Sets a sink for data messages and channel state callbacks. Before media
// transport is destroyed, the sink must be unregistered by setting it to
// nullptr.
virtual void SetDataSink(DataChannelSink* sink) = 0;
// TODO(sukhanov): RtcEventLogs.
};
// If media transport factory is set in peer connection factory, it will be
// used to create media transport for sending/receiving encoded frames and
// this transport will be used instead of default RTP/SRTP transport.
//
// Currently Media Transport negotiation is not supported in SDP.
// If application is using media transport, it must negotiate it before
// setting media transport factory in peer connection.
class MediaTransportFactory {
public:
virtual ~MediaTransportFactory() = default;
// Creates media transport.
// - Does not take ownership of packet_transport or network_thread.
// - Does not support group calls, in 1:1 call one side must set
// is_caller = true and another is_caller = false.
// TODO(bugs.webrtc.org/9938) This constructor will be removed and replaced
// with the one below.
virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
bool is_caller);
// Creates media transport.
// - Does not take ownership of packet_transport or network_thread.
// TODO(bugs.webrtc.org/9938): remove default implementation once all children
// override it.
virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
const MediaTransportSettings& settings);
};
} // namespace webrtc
#endif // API_MEDIA_TRANSPORT_INTERFACE_H_