Move MediaTransportInterface out of the libjingle_peerconnection_api target
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index a4cd2bc..fc7121c 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -114,7 +114,6 @@
"crypto_params.h",
"data_channel_interface.cc",
"data_channel_interface.h",
- "data_channel_transport_interface.cc",
"data_channel_transport_interface.h",
"datagram_transport_interface.h",
"dtls_transport_interface.cc",
@@ -130,9 +129,7 @@
"media_stream_interface.h",
"media_stream_proxy.h",
"media_stream_track_proxy.h",
- "media_transport_config.cc",
"media_transport_config.h",
- "media_transport_interface.cc",
"media_transport_interface.h",
"notifier.h",
"packet_socket_factory.h",
@@ -175,8 +172,10 @@
"rtc_event_log",
"task_queue",
"transport:bitrate_settings",
+ "transport:datagram_transport_interface",
"transport:network_control",
"transport/media:audio_interfaces",
+ "transport/media:media_transport_interface",
"transport/media:video_interfaces",
"transport/rtp:rtp_source",
"units:data_rate",
@@ -256,6 +255,7 @@
"../test:test_common",
"../test:video_test_common",
"transport:network_control",
+ "transport/media:media_transport_interface",
"video_codecs:video_codecs_api",
]
}
@@ -350,6 +350,7 @@
"rtc_event_log",
"task_queue",
"transport:network_control",
+ "transport/media:media_transport_interface",
"units:time_delta",
"video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/memory",
@@ -873,8 +874,9 @@
]
deps = [
- ":libjingle_peerconnection_api",
"../rtc_base:checks",
+ "transport:datagram_transport_interface",
+ "transport/media:media_transport_interface",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
]
@@ -889,9 +891,10 @@
]
deps = [
- ":libjingle_peerconnection_api",
"../rtc_base",
"../rtc_base:checks",
+ "transport:datagram_transport_interface",
+ "transport/media:media_transport_interface",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
]
diff --git a/api/congestion_control_interface.h b/api/congestion_control_interface.h
index 2e822db..3666022 100644
--- a/api/congestion_control_interface.h
+++ b/api/congestion_control_interface.h
@@ -7,61 +7,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// This is EXPERIMENTAL interface for media and datagram transports.
-
#ifndef API_CONGESTION_CONTROL_INTERFACE_H_
#define API_CONGESTION_CONTROL_INTERFACE_H_
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "api/media_transport_interface.h"
-#include "api/units/data_rate.h"
-
-namespace webrtc {
-
-// Defines congestion control feedback interface for media and datagram
-// transports.
-class CongestionControlInterface {
- public:
- virtual ~CongestionControlInterface() = default;
-
- // Updates allocation limits.
- virtual void SetAllocatedBitrateLimits(
- const MediaTransportAllocatedBitrateLimits& limits) = 0;
-
- // Sets starting rate.
- virtual void SetTargetBitrateLimits(
- const MediaTransportTargetRateConstraints& target_rate_constraints) = 0;
-
- // Intended for receive side. AddRttObserver registers an observer to be
- // called for each RTT measurement, typically once per ACK. Before media
- // transport is destructed the observer must be unregistered.
- //
- // TODO(sukhanov): Looks like AddRttObserver and RemoveRttObserver were
- // never implemented for media transport, so keeping noop implementation.
- virtual void AddRttObserver(MediaTransportRttObserver* observer) {}
- virtual void RemoveRttObserver(MediaTransportRttObserver* observer) {}
-
- // 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(
- TargetTransferRateObserver* observer) = 0;
-
- // Removes an existing |observer| from observers. If observer was never
- // registered, an error is logged and method does nothing.
- virtual void RemoveTargetTransferRateObserver(
- TargetTransferRateObserver* observer) = 0;
-
- // Returns the last known target transfer rate as reported to the above
- // observers.
- virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate() = 0;
-};
-
-} // namespace webrtc
+// TODO(bugs.webrtc.org/8733): Delete once users are updated for the new
+// location.
+#include "api/transport/congestion_control_interface.h"
#endif // API_CONGESTION_CONTROL_INTERFACE_H_
diff --git a/api/data_channel_transport_interface.h b/api/data_channel_transport_interface.h
index a6825f6..dcb693c 100644
--- a/api/data_channel_transport_interface.h
+++ b/api/data_channel_transport_interface.h
@@ -7,119 +7,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// This is an experimental interface and is subject to change without notice.
-
#ifndef API_DATA_CHANNEL_TRANSPORT_INTERFACE_H_
#define API_DATA_CHANNEL_TRANSPORT_INTERFACE_H_
-#include "absl/types/optional.h"
-#include "api/rtc_error.h"
-#include "rtc_base/copy_on_write_buffer.h"
-
-namespace webrtc {
-
-// 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;
-
- // Callback issued when the data channel becomes ready to send.
- // This callback will be issued immediately when the data channel sink is
- // registered if the transport is ready at that time. This callback may be
- // invoked again following send errors (eg. due to the transport being
- // temporarily blocked or unavailable).
- // TODO(mellem): Make pure virtual when downstream sinks override this.
- virtual void OnReadyToSend();
-};
-
-// Transport for data channels.
-class DataChannelTransportInterface {
- public:
- virtual ~DataChannelTransportInterface() = default;
-
- // Opens a data |channel_id| for sending. May return an error if the
- // specified |channel_id| is unusable. Must be called before |SendData|.
- virtual RTCError OpenChannel(int channel_id);
-
- // 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);
-
- // 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);
-
- // 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);
-
- // Returns whether this data channel transport is ready to send.
- // Note: the default implementation always returns false (as it assumes no one
- // has implemented the interface). This default implementation is temporary.
- // TODO(mellem): Change this to pure virtual.
- virtual bool IsReadyToSend() const;
-};
-
-} // namespace webrtc
+// TODO(bugs.webrtc.org/8733): Delete once users are updated for the new
+// location.
+#include "api/transport/data_channel_transport_interface.h"
#endif // API_DATA_CHANNEL_TRANSPORT_INTERFACE_H_
diff --git a/api/datagram_transport_interface.h b/api/datagram_transport_interface.h
index 38d6dd5..f36f5b3 100644
--- a/api/datagram_transport_interface.h
+++ b/api/datagram_transport_interface.h
@@ -7,143 +7,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// This is EXPERIMENTAL interface for media and datagram transports.
-
#ifndef API_DATAGRAM_TRANSPORT_INTERFACE_H_
#define API_DATAGRAM_TRANSPORT_INTERFACE_H_
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "absl/types/optional.h"
-#include "api/array_view.h"
-#include "api/congestion_control_interface.h"
-#include "api/data_channel_transport_interface.h"
-#include "api/media_transport_interface.h"
-#include "api/rtc_error.h"
-#include "api/units/data_rate.h"
-#include "api/units/timestamp.h"
-
-namespace rtc {
-class PacketTransportInternal;
-} // namespace rtc
-
-namespace webrtc {
-
-typedef int64_t DatagramId;
-
-struct DatagramAck {
- // |datagram_id| is same as passed in
- // DatagramTransportInterface::SendDatagram.
- DatagramId datagram_id;
-
- // The timestamp at which the remote peer received the identified datagram,
- // according to that peer's clock.
- Timestamp receive_timestamp = Timestamp::MinusInfinity();
-};
-
-// All sink methods are called on network thread.
-class DatagramSinkInterface {
- public:
- virtual ~DatagramSinkInterface() {}
-
- // Called when new packet is received.
- virtual void OnDatagramReceived(rtc::ArrayView<const uint8_t> data) = 0;
-
- // Called when datagram is actually sent (datragram can be delayed due
- // to congestion control or fusing). |datagram_id| is same as passed in
- // DatagramTransportInterface::SendDatagram.
- virtual void OnDatagramSent(DatagramId datagram_id) = 0;
-
- // Called when datagram is ACKed.
- // TODO(sukhanov): Make pure virtual.
- virtual void OnDatagramAcked(const DatagramAck& datagram_ack) {}
-
- // Called when a datagram is lost.
- virtual void OnDatagramLost(DatagramId datagram_id) {}
-};
-
-// Datagram transport allows to send and receive unreliable packets (datagrams)
-// and receive feedback from congestion control (via
-// CongestionControlInterface). The idea is to send RTP packets as datagrams and
-// have underlying implementation of datagram transport to use QUIC datagram
-// protocol.
-class DatagramTransportInterface : public DataChannelTransportInterface {
- public:
- virtual ~DatagramTransportInterface() = default;
-
- // Connect the datagram transport to the ICE transport.
- // The implementation must be able to ignore incoming packets that don't
- // belong to it.
- virtual void Connect(rtc::PacketTransportInternal* packet_transport) = 0;
-
- // Returns congestion control feedback interface or nullptr if datagram
- // transport does not implement congestion control.
- //
- // Note that right now datagram transport is used without congestion control,
- // but we plan to use it in the future.
- virtual CongestionControlInterface* congestion_control() = 0;
-
- // Sets a state observer callback. Before datagram transport is destroyed, the
- // callback must be unregistered by setting it to nullptr.
- // A newly registered callback will be called with the current state.
- // Datagram transport does not invoke this callback concurrently.
- virtual void SetTransportStateCallback(
- MediaTransportStateCallback* callback) = 0;
-
- // Start asynchronous send of datagram. The status returned by this method
- // only pertains to the synchronous operations (e.g. serialization /
- // packetization), not to the asynchronous operation.
- //
- // Datagrams larger than GetLargestDatagramSize() will fail and return error.
- //
- // Datagrams are sent in FIFO order.
- //
- // |datagram_id| is only used in ACK/LOST notifications in
- // DatagramSinkInterface and does not need to be unique.
- virtual RTCError SendDatagram(rtc::ArrayView<const uint8_t> data,
- DatagramId datagram_id) = 0;
-
- // Returns maximum size of datagram message, does not change.
- // TODO(sukhanov): Because value may be undefined before connection setup
- // is complete, consider returning error when called before connection is
- // established. Currently returns hardcoded const, because integration
- // prototype may call before connection is established.
- virtual size_t GetLargestDatagramSize() const = 0;
-
- // Sets packet sink. Sink must be unset by calling
- // SetDataTransportSink(nullptr) before the data transport is destroyed or
- // before new sink is set.
- virtual void SetDatagramSink(DatagramSinkInterface* sink) = 0;
-
- // Retrieves callers config (i.e. media transport offer) that should be passed
- // to the callee, before the call is connected. Such config is opaque to SDP
- // (sdp just passes it through). The config is a binary blob, so SDP may
- // choose to use base64 to serialize it (or any other approach that guarantees
- // that the binary blob goes through). This should only be called for the
- // caller's perspective.
- //
- // TODO(mellem): Delete.
- virtual absl::optional<std::string> GetTransportParametersOffer() const {
- return absl::nullopt;
- }
-
- // Retrieves transport parameters for this datagram transport. May be called
- // on either client- or server-perspective transports.
- //
- // For servers, the parameters represent what kind of connections and data the
- // server is prepared to accept. This is generally a superset of acceptable
- // parameters.
- //
- // For clients, the parameters echo the server configuration used to create
- // the client, possibly removing any fields or parameters which the client
- // does not understand.
- //
- // TODO(mellem): Make pure virtual.
- virtual std::string GetTransportParameters() const { return ""; }
-};
-
-} // namespace webrtc
+// TODO(bugs.webrtc.org/8733): Delete once users are updated for the new
+// location.
+#include "api/transport/datagram_transport_interface.h"
#endif // API_DATAGRAM_TRANSPORT_INTERFACE_H_
diff --git a/api/media_transport_config.h b/api/media_transport_config.h
index 7c5104b..c74f38d 100644
--- a/api/media_transport_config.h
+++ b/api/media_transport_config.h
@@ -9,39 +9,8 @@
#ifndef API_MEDIA_TRANSPORT_CONFIG_H_
#define API_MEDIA_TRANSPORT_CONFIG_H_
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "absl/types/optional.h"
-
-namespace webrtc {
-
-class MediaTransportInterface;
-
-// Media transport config is made available to both transport and audio / video
-// layers, but access to individual interfaces should not be open without
-// necessity.
-struct MediaTransportConfig {
- // Default constructor for no-media transport scenarios.
- MediaTransportConfig() = default;
-
- // Constructor for media transport scenarios.
- // Note that |media_transport| may not be nullptr.
- explicit MediaTransportConfig(MediaTransportInterface* media_transport);
-
- // Constructor for datagram transport scenarios.
- explicit MediaTransportConfig(size_t rtp_max_packet_size);
-
- std::string DebugString() const;
-
- // If provided, all media is sent through media_transport.
- MediaTransportInterface* media_transport = nullptr;
-
- // If provided, limits RTP packet size (excludes ICE, IP or network overhead).
- absl::optional<size_t> rtp_max_packet_size;
-};
-
-} // namespace webrtc
+// TODO(bugs.webrtc.org/8733): Delete once users are updated for the new
+// location.
+#include "api/transport/media/media_transport_config.h"
#endif // API_MEDIA_TRANSPORT_CONFIG_H_
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index 609ae2c..867871b 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -7,322 +7,11 @@
* 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 <memory>
-#include <string>
-#include <utility>
+// TODO(bugs.webrtc.org/8733): Delete once users are updated for the new
+// location.
+#include "api/transport/media/media_transport_interface.h"
-#include "absl/types/optional.h"
-#include "api/array_view.h"
-#include "api/data_channel_transport_interface.h"
-#include "api/rtc_error.h"
-#include "api/transport/media/audio_transport.h"
-#include "api/transport/media/video_transport.h"
-#include "api/transport/network_control.h"
-#include "api/units/data_rate.h"
-#include "common_types.h" // NOLINT(build/include)
-#include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/network_route.h"
-
-namespace rtc {
-class PacketTransportInternal;
-class Thread;
-} // namespace rtc
-
-namespace webrtc {
-
-class DatagramTransportInterface;
-class RtcEventLog;
-
-class AudioPacketReceivedObserver {
- public:
- virtual ~AudioPacketReceivedObserver() = default;
-
- // Invoked for the first received audio packet on a given channel id.
- // It will be invoked once for each channel id.
- virtual void OnFirstAudioPacketReceived(int64_t channel_id) = 0;
-};
-
-// Used to configure stream allocations.
-struct MediaTransportAllocatedBitrateLimits {
- DataRate min_pacing_rate = DataRate::Zero();
- DataRate max_padding_bitrate = DataRate::Zero();
- DataRate max_total_allocated_bitrate = DataRate::Zero();
-};
-
-// Used to configure target bitrate constraints.
-// If the value is provided, the constraint is updated.
-// If the value is omitted, the value is left unchanged.
-struct MediaTransportTargetRateConstraints {
- absl::optional<DataRate> min_bitrate;
- absl::optional<DataRate> max_bitrate;
- absl::optional<DataRate> starting_bitrate;
-};
-
-// 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;
-
- // If present, this is a config passed from the caller to the answerer in the
- // offer. Each media transport knows how to understand its own parameters.
- absl::optional<std::string> remote_transport_parameters;
-
- // If present, provides the event log that media transport should use.
- // Media transport does not own it. The lifetime of |event_log| will exceed
- // the lifetime of the instance of MediaTransportInterface instance.
- RtcEventLog* event_log = nullptr;
-};
-
-// 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;
-};
-
-// 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;
-};
-
-// Callback for RTT measurements on the receive side.
-// TODO(nisse): Related interfaces: CallStatsObserver and RtcpRttStats. It's
-// somewhat unclear what type of measurement is needed. It's used to configure
-// NACK generation and playout buffer. Either raw measurement values or recent
-// maximum would make sense for this use. Need consolidation of RTT signalling.
-class MediaTransportRttObserver {
- public:
- virtual ~MediaTransportRttObserver() = default;
-
- // Invoked when a new RTT measurement is available, typically once per ACK.
- virtual void OnRttUpdated(int64_t rtt_ms) = 0;
-};
-
-// Media transport interface for sending / receiving encoded audio/video frames
-// and receiving bandwidth estimate update from congestion control.
-class MediaTransportInterface : public DataChannelTransportInterface {
- public:
- MediaTransportInterface();
- virtual ~MediaTransportInterface();
-
- // Retrieves callers config (i.e. media transport offer) that should be passed
- // to the callee, before the call is connected. Such config is opaque to SDP
- // (sdp just passes it through). The config is a binary blob, so SDP may
- // choose to use base64 to serialize it (or any other approach that guarantees
- // that the binary blob goes through). This should only be called for the
- // caller's perspective.
- //
- // This may return an unset optional, which means that the given media
- // transport is not supported / disabled and shouldn't be reported in SDP.
- //
- // It may also return an empty string, in which case the media transport is
- // supported, but without any extra settings.
- // TODO(psla): Make abstract.
- virtual absl::optional<std::string> GetTransportParametersOffer() const;
-
- // Connect the media transport to the ICE transport.
- // The implementation must be able to ignore incoming packets that don't
- // belong to it.
- // TODO(psla): Make abstract.
- virtual void Connect(rtc::PacketTransportInternal* packet_transport);
-
- // 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;
-
- // Used by video sender to be notified on key frame requests.
- virtual void SetKeyFrameRequestCallback(
- MediaTransportKeyFrameRequestCallback* callback);
-
- // 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(
- TargetTransferRateObserver* observer);
-
- // Removes an existing |observer| from observers. If observer was never
- // registered, an error is logged and method does nothing.
- virtual void RemoveTargetTransferRateObserver(
- TargetTransferRateObserver* observer);
-
- // Sets audio packets observer, which gets informed about incoming audio
- // packets. Before destruction, the observer must be unregistered by setting
- // nullptr.
- //
- // This method may be temporary, when the multiplexer is implemented (or
- // multiplexer may use it to demultiplex channel ids).
- virtual void SetFirstAudioPacketReceivedObserver(
- AudioPacketReceivedObserver* observer);
-
- // Intended for receive side. AddRttObserver registers an observer to be
- // called for each RTT measurement, typically once per ACK. Before media
- // transport is destructed the observer must be unregistered.
- virtual void AddRttObserver(MediaTransportRttObserver* observer);
- virtual void RemoveRttObserver(MediaTransportRttObserver* 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.
- // TODO(nisse): Deprecated.
- virtual size_t GetAudioPacketOverhead() const;
-
- // Corresponding observers for audio and video overhead. Before destruction,
- // the observers must be unregistered by setting nullptr.
-
- // TODO(nisse): Should move to per-stream objects, since packetization
- // overhead can vary per stream, e.g., depending on negotiated extensions. In
- // addition, we should move towards reporting total overhead including all
- // layers. Currently, overhead of the lower layers is reported elsewhere,
- // e.g., on route change between IPv4 and IPv6.
- virtual void SetAudioOverheadObserver(OverheadObserver* observer) {}
-
- // Registers an observer for network change events. If the network route is
- // already established when the callback is added, |callback| will be called
- // immediately with the current network route. Before media transport is
- // destroyed, the callback must be removed.
- virtual void AddNetworkChangeCallback(
- MediaTransportNetworkChangeCallback* callback);
- virtual void RemoveNetworkChangeCallback(
- 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;
-
- // Updates allocation limits.
- // TODO(psla): Make abstract when downstream implementation implement it.
- virtual void SetAllocatedBitrateLimits(
- const MediaTransportAllocatedBitrateLimits& limits);
-
- // Sets starting rate.
- // TODO(psla): Make abstract when downstream implementation implement it.
- virtual void SetTargetBitrateLimits(
- const MediaTransportTargetRateConstraints& target_rate_constraints) {}
-
- // 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.
- virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
- CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
- rtc::Thread* network_thread,
- const MediaTransportSettings& settings);
-
- // Creates a new Media Transport in a disconnected state. If the media
- // transport for the caller is created, one can then call
- // MediaTransportInterface::GetTransportParametersOffer on that new instance.
- // TODO(psla): Make abstract.
- virtual RTCErrorOr<std::unique_ptr<webrtc::MediaTransportInterface>>
- CreateMediaTransport(rtc::Thread* network_thread,
- const MediaTransportSettings& settings);
-
- // Creates a new Datagram Transport in a disconnected state. If the datagram
- // transport for the caller is created, one can then call
- // DatagramTransportInterface::GetTransportParametersOffer on that new
- // instance.
- //
- // TODO(sukhanov): Consider separating media and datagram transport factories.
- // TODO(sukhanov): Move factory to a separate .h file.
- virtual RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
- CreateDatagramTransport(rtc::Thread* network_thread,
- const MediaTransportSettings& settings);
-
- // Gets a transport name which is supported by the implementation.
- // Different factories should return different transport names, and at runtime
- // it will be checked that different names were used.
- // For example, "rtp" or "generic" may be returned by two different
- // implementations.
- // The value returned by this method must never change in the lifetime of the
- // factory.
- // TODO(psla): Make abstract.
- virtual std::string GetTransportName() const;
-};
-
-} // namespace webrtc
#endif // API_MEDIA_TRANSPORT_INTERFACE_H_
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index afa771f..835e9ae 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -84,7 +84,6 @@
#include "api/fec_controller.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
-#include "api/media_transport_interface.h"
#include "api/network_state_predictor.h"
#include "api/packet_socket_factory.h"
#include "api/rtc_error.h"
@@ -98,6 +97,7 @@
#include "api/stats_types.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/bitrate_settings.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/transport/network_control.h"
#include "api/turn_customizer.h"
#include "media/base/media_config.h"
diff --git a/api/test/fake_datagram_transport.h b/api/test/fake_datagram_transport.h
index 9a1ddef..8cb399c 100644
--- a/api/test/fake_datagram_transport.h
+++ b/api/test/fake_datagram_transport.h
@@ -14,7 +14,8 @@
#include <cstddef>
#include <string>
-#include "api/datagram_transport_interface.h"
+#include "api/transport/datagram_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
namespace webrtc {
diff --git a/api/test/fake_media_transport.h b/api/test/fake_media_transport.h
index 025965b..3bd4eba 100644
--- a/api/test/fake_media_transport.h
+++ b/api/test/fake_media_transport.h
@@ -18,8 +18,8 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
-#include "api/media_transport_interface.h"
#include "api/test/fake_datagram_transport.h"
+#include "api/transport/media/media_transport_interface.h"
namespace webrtc {
diff --git a/api/test/loopback_media_transport.h b/api/test/loopback_media_transport.h
index cc66d62..e00cc23 100644
--- a/api/test/loopback_media_transport.h
+++ b/api/test/loopback_media_transport.h
@@ -17,8 +17,8 @@
#include <vector>
#include "absl/memory/memory.h"
-#include "api/datagram_transport_interface.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/datagram_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "rtc_base/async_invoker.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread.h"
diff --git a/api/test/peerconnection_quality_test_fixture.h b/api/test/peerconnection_quality_test_fixture.h
index fa63ca0..3a654dd 100644
--- a/api/test/peerconnection_quality_test_fixture.h
+++ b/api/test/peerconnection_quality_test_fixture.h
@@ -21,7 +21,6 @@
#include "api/call/call_factory_interface.h"
#include "api/fec_controller.h"
#include "api/function_view.h"
-#include "api/media_transport_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
#include "api/task_queue/task_queue_factory.h"
@@ -29,6 +28,7 @@
#include "api/test/simulated_network.h"
#include "api/test/stats_observer_interface.h"
#include "api/test/video_quality_analyzer_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/transport/network_control.h"
#include "api/units/time_delta.h"
#include "api/video_codecs/video_decoder_factory.h"
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 1b48555..b07021b 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -69,6 +69,25 @@
]
}
+rtc_source_set("datagram_transport_interface") {
+ visibility = [ "*" ]
+ sources = [
+ "congestion_control_interface.h",
+ "data_channel_transport_interface.cc",
+ "data_channel_transport_interface.h",
+ "datagram_transport_interface.h",
+ ]
+ deps = [
+ ":network_control",
+ "..:array_view",
+ "..:rtc_error",
+ "../../rtc_base:rtc_base_approved",
+ "../units:data_rate",
+ "../units:timestamp",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
+}
+
rtc_static_library("goog_cc") {
visibility = [ "*" ]
sources = [
diff --git a/api/transport/congestion_control_interface.h b/api/transport/congestion_control_interface.h
new file mode 100644
index 0000000..40552cb
--- /dev/null
+++ b/api/transport/congestion_control_interface.h
@@ -0,0 +1,75 @@
+/* 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 and datagram transports.
+
+#ifndef API_TRANSPORT_CONGESTION_CONTROL_INTERFACE_H_
+#define API_TRANSPORT_CONGESTION_CONTROL_INTERFACE_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "api/transport/network_control.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+
+// TODO(nisse): Defined together with MediaTransportInterface. But we should use
+// types that aren't tied to media, so that MediaTransportInterface can depend
+// on CongestionControlInterface, but not the other way around.
+// api/transport/network_control.h may be a reasonable place.
+class MediaTransportRttObserver;
+struct MediaTransportAllocatedBitrateLimits;
+struct MediaTransportTargetRateConstraints;
+
+// Defines congestion control feedback interface for media and datagram
+// transports.
+class CongestionControlInterface {
+ public:
+ virtual ~CongestionControlInterface() = default;
+
+ // Updates allocation limits.
+ virtual void SetAllocatedBitrateLimits(
+ const MediaTransportAllocatedBitrateLimits& limits) = 0;
+
+ // Sets starting rate.
+ virtual void SetTargetBitrateLimits(
+ const MediaTransportTargetRateConstraints& target_rate_constraints) = 0;
+
+ // Intended for receive side. AddRttObserver registers an observer to be
+ // called for each RTT measurement, typically once per ACK. Before media
+ // transport is destructed the observer must be unregistered.
+ //
+ // TODO(sukhanov): Looks like AddRttObserver and RemoveRttObserver were
+ // never implemented for media transport, so keeping noop implementation.
+ virtual void AddRttObserver(MediaTransportRttObserver* observer) {}
+ virtual void RemoveRttObserver(MediaTransportRttObserver* observer) {}
+
+ // 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(
+ TargetTransferRateObserver* observer) = 0;
+
+ // Removes an existing |observer| from observers. If observer was never
+ // registered, an error is logged and method does nothing.
+ virtual void RemoveTargetTransferRateObserver(
+ TargetTransferRateObserver* observer) = 0;
+
+ // Returns the last known target transfer rate as reported to the above
+ // observers.
+ virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate() = 0;
+};
+
+} // namespace webrtc
+
+#endif // API_TRANSPORT_CONGESTION_CONTROL_INTERFACE_H_
diff --git a/api/data_channel_transport_interface.cc b/api/transport/data_channel_transport_interface.cc
similarity index 95%
rename from api/data_channel_transport_interface.cc
rename to api/transport/data_channel_transport_interface.cc
index d9947e2..122e282 100644
--- a/api/data_channel_transport_interface.cc
+++ b/api/transport/data_channel_transport_interface.cc
@@ -7,7 +7,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "api/data_channel_transport_interface.h"
+#include "api/transport/data_channel_transport_interface.h"
namespace webrtc {
diff --git a/api/transport/data_channel_transport_interface.h b/api/transport/data_channel_transport_interface.h
new file mode 100644
index 0000000..9b29323
--- /dev/null
+++ b/api/transport/data_channel_transport_interface.h
@@ -0,0 +1,125 @@
+/* Copyright 2019 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 an experimental interface and is subject to change without notice.
+
+#ifndef API_TRANSPORT_DATA_CHANNEL_TRANSPORT_INTERFACE_H_
+#define API_TRANSPORT_DATA_CHANNEL_TRANSPORT_INTERFACE_H_
+
+#include "absl/types/optional.h"
+#include "api/rtc_error.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+// 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;
+
+ // Callback issued when the data channel becomes ready to send.
+ // This callback will be issued immediately when the data channel sink is
+ // registered if the transport is ready at that time. This callback may be
+ // invoked again following send errors (eg. due to the transport being
+ // temporarily blocked or unavailable).
+ // TODO(mellem): Make pure virtual when downstream sinks override this.
+ virtual void OnReadyToSend();
+};
+
+// Transport for data channels.
+class DataChannelTransportInterface {
+ public:
+ virtual ~DataChannelTransportInterface() = default;
+
+ // Opens a data |channel_id| for sending. May return an error if the
+ // specified |channel_id| is unusable. Must be called before |SendData|.
+ virtual RTCError OpenChannel(int channel_id);
+
+ // 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);
+
+ // 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);
+
+ // 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);
+
+ // Returns whether this data channel transport is ready to send.
+ // Note: the default implementation always returns false (as it assumes no one
+ // has implemented the interface). This default implementation is temporary.
+ // TODO(mellem): Change this to pure virtual.
+ virtual bool IsReadyToSend() const;
+};
+
+} // namespace webrtc
+
+#endif // API_TRANSPORT_DATA_CHANNEL_TRANSPORT_INTERFACE_H_
diff --git a/api/transport/datagram_transport_interface.h b/api/transport/datagram_transport_interface.h
new file mode 100644
index 0000000..9820c75
--- /dev/null
+++ b/api/transport/datagram_transport_interface.h
@@ -0,0 +1,150 @@
+/* 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 and datagram transports.
+
+#ifndef API_TRANSPORT_DATAGRAM_TRANSPORT_INTERFACE_H_
+#define API_TRANSPORT_DATAGRAM_TRANSPORT_INTERFACE_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/rtc_error.h"
+#include "api/transport/congestion_control_interface.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+
+namespace rtc {
+class PacketTransportInternal;
+} // namespace rtc
+
+namespace webrtc {
+
+class MediaTransportStateCallback;
+
+typedef int64_t DatagramId;
+
+struct DatagramAck {
+ // |datagram_id| is same as passed in
+ // DatagramTransportInterface::SendDatagram.
+ DatagramId datagram_id;
+
+ // The timestamp at which the remote peer received the identified datagram,
+ // according to that peer's clock.
+ Timestamp receive_timestamp = Timestamp::MinusInfinity();
+};
+
+// All sink methods are called on network thread.
+class DatagramSinkInterface {
+ public:
+ virtual ~DatagramSinkInterface() {}
+
+ // Called when new packet is received.
+ virtual void OnDatagramReceived(rtc::ArrayView<const uint8_t> data) = 0;
+
+ // Called when datagram is actually sent (datragram can be delayed due
+ // to congestion control or fusing). |datagram_id| is same as passed in
+ // DatagramTransportInterface::SendDatagram.
+ virtual void OnDatagramSent(DatagramId datagram_id) = 0;
+
+ // Called when datagram is ACKed.
+ // TODO(sukhanov): Make pure virtual.
+ virtual void OnDatagramAcked(const DatagramAck& datagram_ack) {}
+
+ // Called when a datagram is lost.
+ virtual void OnDatagramLost(DatagramId datagram_id) {}
+};
+
+// Datagram transport allows to send and receive unreliable packets (datagrams)
+// and receive feedback from congestion control (via
+// CongestionControlInterface). The idea is to send RTP packets as datagrams and
+// have underlying implementation of datagram transport to use QUIC datagram
+// protocol.
+class DatagramTransportInterface : public DataChannelTransportInterface {
+ public:
+ virtual ~DatagramTransportInterface() = default;
+
+ // Connect the datagram transport to the ICE transport.
+ // The implementation must be able to ignore incoming packets that don't
+ // belong to it.
+ virtual void Connect(rtc::PacketTransportInternal* packet_transport) = 0;
+
+ // Returns congestion control feedback interface or nullptr if datagram
+ // transport does not implement congestion control.
+ //
+ // Note that right now datagram transport is used without congestion control,
+ // but we plan to use it in the future.
+ virtual CongestionControlInterface* congestion_control() = 0;
+
+ // Sets a state observer callback. Before datagram transport is destroyed, the
+ // callback must be unregistered by setting it to nullptr.
+ // A newly registered callback will be called with the current state.
+ // Datagram transport does not invoke this callback concurrently.
+ virtual void SetTransportStateCallback(
+ MediaTransportStateCallback* callback) = 0;
+
+ // Start asynchronous send of datagram. The status returned by this method
+ // only pertains to the synchronous operations (e.g. serialization /
+ // packetization), not to the asynchronous operation.
+ //
+ // Datagrams larger than GetLargestDatagramSize() will fail and return error.
+ //
+ // Datagrams are sent in FIFO order.
+ //
+ // |datagram_id| is only used in ACK/LOST notifications in
+ // DatagramSinkInterface and does not need to be unique.
+ virtual RTCError SendDatagram(rtc::ArrayView<const uint8_t> data,
+ DatagramId datagram_id) = 0;
+
+ // Returns maximum size of datagram message, does not change.
+ // TODO(sukhanov): Because value may be undefined before connection setup
+ // is complete, consider returning error when called before connection is
+ // established. Currently returns hardcoded const, because integration
+ // prototype may call before connection is established.
+ virtual size_t GetLargestDatagramSize() const = 0;
+
+ // Sets packet sink. Sink must be unset by calling
+ // SetDataTransportSink(nullptr) before the data transport is destroyed or
+ // before new sink is set.
+ virtual void SetDatagramSink(DatagramSinkInterface* sink) = 0;
+
+ // Retrieves callers config (i.e. media transport offer) that should be passed
+ // to the callee, before the call is connected. Such config is opaque to SDP
+ // (sdp just passes it through). The config is a binary blob, so SDP may
+ // choose to use base64 to serialize it (or any other approach that guarantees
+ // that the binary blob goes through). This should only be called for the
+ // caller's perspective.
+ //
+ // TODO(mellem): Delete.
+ virtual absl::optional<std::string> GetTransportParametersOffer() const {
+ return absl::nullopt;
+ }
+
+ // Retrieves transport parameters for this datagram transport. May be called
+ // on either client- or server-perspective transports.
+ //
+ // For servers, the parameters represent what kind of connections and data the
+ // server is prepared to accept. This is generally a superset of acceptable
+ // parameters.
+ //
+ // For clients, the parameters echo the server configuration used to create
+ // the client, possibly removing any fields or parameters which the client
+ // does not understand.
+ //
+ // TODO(mellem): Make pure virtual.
+ virtual std::string GetTransportParameters() const { return ""; }
+};
+
+} // namespace webrtc
+
+#endif // API_TRANSPORT_DATAGRAM_TRANSPORT_INTERFACE_H_
diff --git a/api/transport/media/BUILD.gn b/api/transport/media/BUILD.gn
index f338021..fe8e4e4 100644
--- a/api/transport/media/BUILD.gn
+++ b/api/transport/media/BUILD.gn
@@ -8,6 +8,31 @@
import("../../../webrtc.gni")
+rtc_source_set("media_transport_interface") {
+ visibility = [ "*" ]
+ sources = [
+ "media_transport_config.cc",
+ "media_transport_config.h",
+ "media_transport_interface.cc",
+ "media_transport_interface.h",
+ ]
+ deps = [
+ ":audio_interfaces",
+ ":video_interfaces",
+ "..:datagram_transport_interface",
+ "..:network_control",
+ "../..:array_view",
+ "../..:rtc_error",
+ "../../..:webrtc_common",
+ "../../../rtc_base",
+ "../../../rtc_base:checks",
+ "../../../rtc_base:rtc_base_approved",
+ "../../../rtc_base:stringutils",
+ "../../units:data_rate",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
+}
+
rtc_source_set("audio_interfaces") {
visibility = [ "*" ]
sources = [
diff --git a/api/media_transport_config.cc b/api/transport/media/media_transport_config.cc
similarity index 95%
rename from api/media_transport_config.cc
rename to api/transport/media/media_transport_config.cc
index 08a8756..cea3f16 100644
--- a/api/media_transport_config.cc
+++ b/api/transport/media/media_transport_config.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "api/media_transport_config.h"
+#include "api/transport/media/media_transport_config.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
diff --git a/api/transport/media/media_transport_config.h b/api/transport/media/media_transport_config.h
new file mode 100644
index 0000000..6a12630
--- /dev/null
+++ b/api/transport/media/media_transport_config.h
@@ -0,0 +1,46 @@
+/* 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.
+ */
+#ifndef API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
+#define API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/transport/media/media_transport_interface.h"
+
+namespace webrtc {
+
+// Media transport config is made available to both transport and audio / video
+// layers, but access to individual interfaces should not be open without
+// necessity.
+struct MediaTransportConfig {
+ // Default constructor for no-media transport scenarios.
+ MediaTransportConfig() = default;
+
+ // Constructor for media transport scenarios.
+ // Note that |media_transport| may not be nullptr.
+ explicit MediaTransportConfig(MediaTransportInterface* media_transport);
+
+ // Constructor for datagram transport scenarios.
+ explicit MediaTransportConfig(size_t rtp_max_packet_size);
+
+ std::string DebugString() const;
+
+ // If provided, all media is sent through media_transport.
+ MediaTransportInterface* media_transport = nullptr;
+
+ // If provided, limits RTP packet size (excludes ICE, IP or network overhead).
+ absl::optional<size_t> rtp_max_packet_size;
+};
+
+} // namespace webrtc
+
+#endif // API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
diff --git a/api/media_transport_interface.cc b/api/transport/media/media_transport_interface.cc
similarity index 96%
rename from api/media_transport_interface.cc
rename to api/transport/media/media_transport_interface.cc
index 69f993e..323ddca 100644
--- a/api/media_transport_interface.cc
+++ b/api/transport/media/media_transport_interface.cc
@@ -15,12 +15,12 @@
// enable different media transport implementations, including QUIC-based
// media transport.
-#include "api/media_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include <cstdint>
#include <utility>
-#include "api/datagram_transport_interface.h"
+#include "api/transport/datagram_transport_interface.h"
namespace webrtc {
diff --git a/api/transport/media/media_transport_interface.h b/api/transport/media/media_transport_interface.h
new file mode 100644
index 0000000..04a8e50
--- /dev/null
+++ b/api/transport/media/media_transport_interface.h
@@ -0,0 +1,328 @@
+/* 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_TRANSPORT_MEDIA_MEDIA_TRANSPORT_INTERFACE_H_
+#define API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_INTERFACE_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/rtc_error.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "api/transport/media/audio_transport.h"
+#include "api/transport/media/video_transport.h"
+#include "api/transport/network_control.h"
+#include "api/units/data_rate.h"
+#include "common_types.h" // NOLINT(build/include)
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/network_route.h"
+
+namespace rtc {
+class PacketTransportInternal;
+class Thread;
+} // namespace rtc
+
+namespace webrtc {
+
+class DatagramTransportInterface;
+class RtcEventLog;
+
+class AudioPacketReceivedObserver {
+ public:
+ virtual ~AudioPacketReceivedObserver() = default;
+
+ // Invoked for the first received audio packet on a given channel id.
+ // It will be invoked once for each channel id.
+ virtual void OnFirstAudioPacketReceived(int64_t channel_id) = 0;
+};
+
+// Used to configure stream allocations.
+struct MediaTransportAllocatedBitrateLimits {
+ DataRate min_pacing_rate = DataRate::Zero();
+ DataRate max_padding_bitrate = DataRate::Zero();
+ DataRate max_total_allocated_bitrate = DataRate::Zero();
+};
+
+// Used to configure target bitrate constraints.
+// If the value is provided, the constraint is updated.
+// If the value is omitted, the value is left unchanged.
+struct MediaTransportTargetRateConstraints {
+ absl::optional<DataRate> min_bitrate;
+ absl::optional<DataRate> max_bitrate;
+ absl::optional<DataRate> starting_bitrate;
+};
+
+// 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;
+
+ // If present, this is a config passed from the caller to the answerer in the
+ // offer. Each media transport knows how to understand its own parameters.
+ absl::optional<std::string> remote_transport_parameters;
+
+ // If present, provides the event log that media transport should use.
+ // Media transport does not own it. The lifetime of |event_log| will exceed
+ // the lifetime of the instance of MediaTransportInterface instance.
+ RtcEventLog* event_log = nullptr;
+};
+
+// 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;
+};
+
+// 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;
+};
+
+// Callback for RTT measurements on the receive side.
+// TODO(nisse): Related interfaces: CallStatsObserver and RtcpRttStats. It's
+// somewhat unclear what type of measurement is needed. It's used to configure
+// NACK generation and playout buffer. Either raw measurement values or recent
+// maximum would make sense for this use. Need consolidation of RTT signalling.
+class MediaTransportRttObserver {
+ public:
+ virtual ~MediaTransportRttObserver() = default;
+
+ // Invoked when a new RTT measurement is available, typically once per ACK.
+ virtual void OnRttUpdated(int64_t rtt_ms) = 0;
+};
+
+// Media transport interface for sending / receiving encoded audio/video frames
+// and receiving bandwidth estimate update from congestion control.
+class MediaTransportInterface : public DataChannelTransportInterface {
+ public:
+ MediaTransportInterface();
+ virtual ~MediaTransportInterface();
+
+ // Retrieves callers config (i.e. media transport offer) that should be passed
+ // to the callee, before the call is connected. Such config is opaque to SDP
+ // (sdp just passes it through). The config is a binary blob, so SDP may
+ // choose to use base64 to serialize it (or any other approach that guarantees
+ // that the binary blob goes through). This should only be called for the
+ // caller's perspective.
+ //
+ // This may return an unset optional, which means that the given media
+ // transport is not supported / disabled and shouldn't be reported in SDP.
+ //
+ // It may also return an empty string, in which case the media transport is
+ // supported, but without any extra settings.
+ // TODO(psla): Make abstract.
+ virtual absl::optional<std::string> GetTransportParametersOffer() const;
+
+ // Connect the media transport to the ICE transport.
+ // The implementation must be able to ignore incoming packets that don't
+ // belong to it.
+ // TODO(psla): Make abstract.
+ virtual void Connect(rtc::PacketTransportInternal* packet_transport);
+
+ // 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;
+
+ // Used by video sender to be notified on key frame requests.
+ virtual void SetKeyFrameRequestCallback(
+ MediaTransportKeyFrameRequestCallback* callback);
+
+ // 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(
+ TargetTransferRateObserver* observer);
+
+ // Removes an existing |observer| from observers. If observer was never
+ // registered, an error is logged and method does nothing.
+ virtual void RemoveTargetTransferRateObserver(
+ TargetTransferRateObserver* observer);
+
+ // Sets audio packets observer, which gets informed about incoming audio
+ // packets. Before destruction, the observer must be unregistered by setting
+ // nullptr.
+ //
+ // This method may be temporary, when the multiplexer is implemented (or
+ // multiplexer may use it to demultiplex channel ids).
+ virtual void SetFirstAudioPacketReceivedObserver(
+ AudioPacketReceivedObserver* observer);
+
+ // Intended for receive side. AddRttObserver registers an observer to be
+ // called for each RTT measurement, typically once per ACK. Before media
+ // transport is destructed the observer must be unregistered.
+ virtual void AddRttObserver(MediaTransportRttObserver* observer);
+ virtual void RemoveRttObserver(MediaTransportRttObserver* 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.
+ // TODO(nisse): Deprecated.
+ virtual size_t GetAudioPacketOverhead() const;
+
+ // Corresponding observers for audio and video overhead. Before destruction,
+ // the observers must be unregistered by setting nullptr.
+
+ // TODO(nisse): Should move to per-stream objects, since packetization
+ // overhead can vary per stream, e.g., depending on negotiated extensions. In
+ // addition, we should move towards reporting total overhead including all
+ // layers. Currently, overhead of the lower layers is reported elsewhere,
+ // e.g., on route change between IPv4 and IPv6.
+ virtual void SetAudioOverheadObserver(OverheadObserver* observer) {}
+
+ // Registers an observer for network change events. If the network route is
+ // already established when the callback is added, |callback| will be called
+ // immediately with the current network route. Before media transport is
+ // destroyed, the callback must be removed.
+ virtual void AddNetworkChangeCallback(
+ MediaTransportNetworkChangeCallback* callback);
+ virtual void RemoveNetworkChangeCallback(
+ 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;
+
+ // Updates allocation limits.
+ // TODO(psla): Make abstract when downstream implementation implement it.
+ virtual void SetAllocatedBitrateLimits(
+ const MediaTransportAllocatedBitrateLimits& limits);
+
+ // Sets starting rate.
+ // TODO(psla): Make abstract when downstream implementation implement it.
+ virtual void SetTargetBitrateLimits(
+ const MediaTransportTargetRateConstraints& target_rate_constraints) {}
+
+ // 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.
+ virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
+ CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
+ rtc::Thread* network_thread,
+ const MediaTransportSettings& settings);
+
+ // Creates a new Media Transport in a disconnected state. If the media
+ // transport for the caller is created, one can then call
+ // MediaTransportInterface::GetTransportParametersOffer on that new instance.
+ // TODO(psla): Make abstract.
+ virtual RTCErrorOr<std::unique_ptr<webrtc::MediaTransportInterface>>
+ CreateMediaTransport(rtc::Thread* network_thread,
+ const MediaTransportSettings& settings);
+
+ // Creates a new Datagram Transport in a disconnected state. If the datagram
+ // transport for the caller is created, one can then call
+ // DatagramTransportInterface::GetTransportParametersOffer on that new
+ // instance.
+ //
+ // TODO(sukhanov): Consider separating media and datagram transport factories.
+ // TODO(sukhanov): Move factory to a separate .h file.
+ virtual RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
+ CreateDatagramTransport(rtc::Thread* network_thread,
+ const MediaTransportSettings& settings);
+
+ // Gets a transport name which is supported by the implementation.
+ // Different factories should return different transport names, and at runtime
+ // it will be checked that different names were used.
+ // For example, "rtp" or "generic" may be returned by two different
+ // implementations.
+ // The value returned by this method must never change in the lifetime of the
+ // factory.
+ // TODO(psla): Make abstract.
+ virtual std::string GetTransportName() const;
+};
+
+} // namespace webrtc
+#endif // API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_INTERFACE_H_
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index abf4c67..dba7b58 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -54,6 +54,7 @@
"../api/audio_codecs:audio_codecs_api",
"../api/rtc_event_log",
"../api/task_queue",
+ "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../call:bitrate_allocator",
"../call:call_interfaces",
@@ -139,6 +140,7 @@
"../api/audio_codecs/opus:audio_encoder_opus",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
+ "../api/transport/media:media_transport_interface",
"../api/units:time_delta",
"../call:mock_bitrate_allocator",
"../call:mock_call_interfaces",
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 479216a..cd137d9 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -21,8 +21,8 @@
#include "api/call/transport.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/function_view.h"
-#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/media/media_transport_config.h"
#include "audio/audio_state.h"
#include "audio/channel_send.h"
#include "audio/conversion.h"
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index a7151bc..7527ef2 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -22,8 +22,8 @@
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
-#include "api/media_transport_config.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/media/media_transport_config.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/transport/rtp/rtp_source.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 575f71f..6f94610 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -19,9 +19,9 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/crypto/crypto_options.h"
#include "api/function_view.h"
-#include "api/media_transport_config.h"
-#include "api/media_transport_interface.h"
#include "api/task_queue/task_queue_factory.h"
+#include "api/transport/media/media_transport_config.h"
+#include "api/transport/media/media_transport_interface.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
diff --git a/audio/test/media_transport_test.cc b/audio/test/media_transport_test.cc
index cc360df..aacee1e 100644
--- a/audio/test/media_transport_test.cc
+++ b/audio/test/media_transport_test.cc
@@ -13,11 +13,11 @@
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
-#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/loopback_media_transport.h"
#include "api/test/mock_audio_mixer.h"
+#include "api/transport/media/media_transport_config.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "call/rtp_transport_controller_send.h"
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 912abc2..f35c1f0 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -33,7 +33,7 @@
"../api:fec_controller_api",
"../api:rtc_error",
- # For api/media_transport_config.h
+ # For api/crypto/crypto_options.h
"../api:libjingle_peerconnection_api",
"../api:network_state_predictor_api",
"../api:rtp_headers",
@@ -44,6 +44,7 @@
"../api/audio_codecs:audio_codecs_api",
"../api/task_queue",
"../api/transport:network_control",
+ "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../modules/audio_device",
"../modules/audio_processing",
@@ -286,6 +287,7 @@
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:transport_api",
+ "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 0b764a1..935aaed 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -21,9 +21,9 @@
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_decryptor_interface.h"
-#include "api/media_transport_config.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
+#include "api/transport/media/media_transport_config.h"
#include "api/transport/rtp/rtp_source.h"
#include "call/rtp_config.h"
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 1f3d1d0..fb711c4 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -23,10 +23,10 @@
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_encryptor_interface.h"
-#include "api/media_transport_config.h"
-#include "api/media_transport_interface.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
+#include "api/transport/media/media_transport_config.h"
+#include "api/transport/media/media_transport_interface.h"
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 3869c81..b1d45ac 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -20,10 +20,10 @@
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_decryptor_interface.h"
-#include "api/media_transport_config.h"
-#include "api/media_transport_interface.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
+#include "api/transport/media/media_transport_config.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 2c31de0..478d73c 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -20,8 +20,8 @@
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
-#include "api/media_transport_interface.h"
#include "api/rtp_parameters.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 24cf303..be5b2b3 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -79,6 +79,7 @@
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/audio_codecs:audio_codecs_api",
+ "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator_factory",
@@ -266,6 +267,8 @@
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/task_queue",
+ "../api/transport:datagram_transport_interface",
+ "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator_factory",
@@ -530,6 +533,7 @@
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
+ "../api/transport/media:media_transport_interface",
"../api/units:time_delta",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocation",
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 9cc7876..da4f0d2 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -22,9 +22,9 @@
#include "api/audio_options.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
-#include "api/media_transport_config.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
+#include "api/transport/media/media_transport_config.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/video_content_type.h"
#include "api/video/video_sink_interface.h"
diff --git a/media/base/rtp_data_engine_unittest.cc b/media/base/rtp_data_engine_unittest.cc
index cd11eb5..79fb2b2 100644
--- a/media/base/rtp_data_engine_unittest.cc
+++ b/media/base/rtp_data_engine_unittest.cc
@@ -15,7 +15,7 @@
#include <memory>
#include <string>
-#include "api/media_transport_config.h"
+#include "api/transport/media/media_transport_config.h"
#include "media/base/fake_network_interface.h"
#include "media/base/media_constants.h"
#include "media/base/rtp_utils.h"
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 4fbb2c8..f31d69f 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -19,7 +19,7 @@
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
-#include "api/datagram_transport_interface.h"
+#include "api/transport/datagram_transport_interface.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_codec_type.h"
#include "api/video_codecs/sdp_video_format.h"
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index ba1f671..50dd8d8 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -19,7 +19,6 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
-#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/task_queue/default_task_queue_factory.h"
@@ -28,6 +27,7 @@
#include "api/test/mock_video_bitrate_allocator_factory.h"
#include "api/test/mock_video_decoder_factory.h"
#include "api/test/mock_video_encoder_factory.h"
+#include "api/transport/media/media_transport_config.h"
#include "api/units/time_delta.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/i420_buffer.h"
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 5ac32c4..1125780b 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -21,7 +21,7 @@
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/call/audio_sink.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "media/base/audio_source.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 909cdaf..2ecf89e 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -17,11 +17,11 @@
#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/media/media_transport_config.h"
#include "call/call.h"
#include "media/base/fake_media_engine.h"
#include "media/base/fake_network_interface.h"
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 7e1d53f..e75188d 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -88,6 +88,8 @@
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/rtc_event_log",
+ "../api/transport:datagram_transport_interface",
+ "../api/transport/media:media_transport_interface",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
@@ -227,6 +229,8 @@
"../api:scoped_refptr",
"../api/rtc_event_log",
"../api/task_queue",
+ "../api/transport:datagram_transport_interface",
+ "../api/transport/media:media_transport_interface",
"../api/units:data_rate",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_frame",
@@ -314,6 +318,7 @@
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
+ "../api/transport/media:media_transport_interface",
"../api/video:builtin_video_bitrate_allocator_factory",
"../call:rtp_interfaces",
"../call:rtp_receiver",
@@ -547,6 +552,7 @@
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue:default_task_queue_factory",
+ "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
"../api/video:builtin_video_bitrate_allocator_factory",
diff --git a/pc/channel.cc b/pc/channel.cc
index 95be5b6..5966951 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -16,7 +16,7 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "api/call/audio_sink.h"
-#include "api/media_transport_config.h"
+#include "api/transport/media/media_transport_config.h"
#include "media/base/media_constants.h"
#include "media/base/rtp_utils.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
diff --git a/pc/channel.h b/pc/channel.h
index 5222d98..6774f7e 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -20,8 +20,8 @@
#include "api/call/audio_sink.h"
#include "api/jsep.h"
-#include "api/media_transport_config.h"
#include "api/rtp_receiver_interface.h"
+#include "api/transport/media/media_transport_config.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "call/rtp_packet_sink_interface.h"
diff --git a/pc/channel_manager.h b/pc/channel_manager.h
index cae812f..661ab4b 100644
--- a/pc/channel_manager.h
+++ b/pc/channel_manager.h
@@ -19,7 +19,7 @@
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
-#include "api/media_transport_config.h"
+#include "api/transport/media/media_transport_config.h"
#include "call/call.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
diff --git a/pc/channel_manager_unittest.cc b/pc/channel_manager_unittest.cc
index c721614..e88b09c 100644
--- a/pc/channel_manager_unittest.cc
+++ b/pc/channel_manager_unittest.cc
@@ -13,9 +13,9 @@
#include <memory>
#include "absl/memory/memory.h"
-#include "api/media_transport_config.h"
#include "api/rtc_error.h"
#include "api/test/fake_media_transport.h"
+#include "api/transport/media/media_transport_config.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "media/base/fake_media_engine.h"
#include "media/base/test_utils.h"
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index db0e8a8..5b388ea 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -17,8 +17,8 @@
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "api/audio_options.h"
-#include "api/media_transport_config.h"
#include "api/rtp_parameters.h"
+#include "api/transport/media/media_transport_config.h"
#include "media/base/codec.h"
#include "media/base/fake_media_engine.h"
#include "media/base/fake_rtp.h"
diff --git a/pc/datagram_rtp_transport.h b/pc/datagram_rtp_transport.h
index 1dfa37b..8aadf97 100644
--- a/pc/datagram_rtp_transport.h
+++ b/pc/datagram_rtp_transport.h
@@ -17,7 +17,7 @@
#include <vector>
#include "api/crypto/crypto_options.h"
-#include "api/datagram_transport_interface.h"
+#include "api/transport/datagram_transport_interface.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "p2p/base/ice_transport_internal.h"
diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index 1a0e7b4..7bd0b07 100644
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h
@@ -18,9 +18,9 @@
#include "absl/types/optional.h"
#include "api/candidate.h"
-#include "api/datagram_transport_interface.h"
#include "api/jsep.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/datagram_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/transport_info.h"
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index cfb971a..10250ce 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -15,8 +15,8 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
-#include "api/datagram_transport_interface.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/datagram_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/no_op_dtls_transport.h"
#include "p2p/base/port.h"
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index de75db9..bad1315 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -19,10 +19,10 @@
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
-#include "api/media_transport_config.h"
-#include "api/media_transport_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/media/media_transport_config.h"
+#include "api/transport/media/media_transport_interface.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/p2p_transport_channel.h"
diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc
index 887f12b..35ebb87 100644
--- a/pc/jsep_transport_controller_unittest.cc
+++ b/pc/jsep_transport_controller_unittest.cc
@@ -14,9 +14,9 @@
#include <memory>
#include "absl/memory/memory.h"
-#include "api/media_transport_interface.h"
#include "api/test/fake_media_transport.h"
#include "api/test/loopback_media_transport.h"
+#include "api/transport/media/media_transport_interface.h"
#include "p2p/base/fake_dtls_transport.h"
#include "p2p/base/fake_ice_transport.h"
#include "p2p/base/no_op_dtls_transport.h"
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index b6da82a..3da8658 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -18,9 +18,9 @@
#include <utility>
#include <vector>
-#include "api/data_channel_transport_interface.h"
-#include "api/media_transport_interface.h"
#include "api/peer_connection_interface.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/turn_customizer.h"
#include "pc/ice_server_parsing.h"
#include "pc/jsep_transport_controller.h"
diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc
index 787e5ba..609a718 100644
--- a/pc/peer_connection_data_channel_unittest.cc
+++ b/pc/peer_connection_data_channel_unittest.cc
@@ -17,13 +17,13 @@
#include "absl/types/optional.h"
#include "api/call/call_factory_interface.h"
#include "api/jsep.h"
-#include "api/media_transport_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/peer_connection_proxy.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/fake_media_transport.h"
+#include "api/transport/media/media_transport_interface.h"
#include "media/base/codec.h"
#include "media/base/fake_media_engine.h"
#include "media/base/media_constants.h"
diff --git a/pc/peer_connection_factory.cc b/pc/peer_connection_factory.cc
index 1052b3b..16fb928 100644
--- a/pc/peer_connection_factory.cc
+++ b/pc/peer_connection_factory.cc
@@ -18,11 +18,11 @@
#include "api/fec_controller.h"
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
-#include "api/media_transport_interface.h"
#include "api/network_state_predictor.h"
#include "api/peer_connection_factory_proxy.h"
#include "api/peer_connection_proxy.h"
#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/turn_customizer.h"
#include "api/units/data_rate.h"
#include "api/video_track_source_proxy.h"
diff --git a/pc/peer_connection_factory.h b/pc/peer_connection_factory.h
index 9160730..648a3af 100644
--- a/pc/peer_connection_factory.h
+++ b/pc/peer_connection_factory.h
@@ -16,9 +16,9 @@
#include <string>
#include "api/media_stream_interface.h"
-#include "api/media_transport_interface.h"
#include "api/peer_connection_interface.h"
#include "api/scoped_refptr.h"
+#include "api/transport/media/media_transport_interface.h"
#include "media/sctp/sctp_transport_internal.h"
#include "pc/channel_manager.h"
#include "rtc_base/rtc_certificate_generator.h"
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index 1158229..125c6cb 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -929,6 +929,7 @@
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue:default_task_queue_factory",
+ "../api/transport/media:media_transport_interface",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",
@@ -1203,6 +1204,7 @@
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
+ "../api/transport/media:media_transport_interface",
"../api/video_codecs:video_codecs_api",
"../media:rtc_media_base",
"../modules:module_api",
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnection.mm b/sdk/objc/api/peerconnection/RTCPeerConnection.mm
index f3e91c4..097eeb4 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnection.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnection.mm
@@ -28,8 +28,8 @@
#include <memory>
#include "api/jsep_ice_candidate.h"
-#include "api/media_transport_interface.h"
#include "api/rtc_event_log_output_file.h"
+#include "api/transport/media/media_transport_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
index bb695b5..d63c08e 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
@@ -52,7 +52,7 @@
// TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
// API layer.
#include "absl/memory/memory.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "media/engine/webrtc_media_engine.h" // nogncheck
@implementation RTCPeerConnectionFactory {
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
index 0adaa30..af3d259 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
@@ -13,7 +13,7 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "modules/audio_device/include/audio_device.h"
diff --git a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
index 5f889a6..40b3aa0 100644
--- a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
+++ b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
@@ -22,7 +22,7 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/media_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "modules/audio_device/include/audio_device.h"
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index a718589..497fbe2 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -57,6 +57,7 @@
"../../../api/rtc_event_log",
"../../../api/task_queue",
"../../../api/transport:network_control",
+ "../../../api/transport/media:media_transport_interface",
"../../../api/video_codecs:video_codecs_api",
"../../../rtc_base",
"//third_party/abseil-cpp/absl/memory",
diff --git a/test/pc/e2e/peer_connection_quality_test_params.h b/test/pc/e2e/peer_connection_quality_test_params.h
index ea011f8..cb6add8 100644
--- a/test/pc/e2e/peer_connection_quality_test_params.h
+++ b/test/pc/e2e/peer_connection_quality_test_params.h
@@ -18,10 +18,10 @@
#include "api/async_resolver_factory.h"
#include "api/call/call_factory_interface.h"
#include "api/fec_controller.h"
-#include "api/media_transport_interface.h"
#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/transport/network_control.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 8edb069..af34436 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -60,6 +60,7 @@
"../api:transport_api",
"../api/rtc_event_log",
"../api/task_queue",
+ "../api/transport/media:media_transport_interface",
"../api/video:encoded_image",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator",
@@ -266,6 +267,7 @@
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
+ "../api/transport/media:media_transport_interface",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocator_factory",
"../api/video:video_frame",
diff --git a/video/encoder_rtcp_feedback.h b/video/encoder_rtcp_feedback.h
index 8f10442..21624db 100644
--- a/video/encoder_rtcp_feedback.h
+++ b/video/encoder_rtcp_feedback.h
@@ -12,7 +12,7 @@
#include <vector>
-#include "api/media_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "api/video/video_stream_encoder_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 663452a..1511254 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -19,9 +19,9 @@
#include "absl/memory/memory.h"
#include "api/fec_controller_override.h"
-#include "api/media_transport_config.h"
#include "api/rtc_event_log_output_file.h"
#include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/media/media_transport_config.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "call/fake_network_pipe.h"
diff --git a/video/video_receive_stream.h b/video/video_receive_stream.h
index 87a40e9..0d0c66a 100644
--- a/video/video_receive_stream.h
+++ b/video/video_receive_stream.h
@@ -14,8 +14,8 @@
#include <memory>
#include <vector>
-#include "api/media_transport_interface.h"
#include "api/task_queue/task_queue_factory.h"
+#include "api/transport/media/media_transport_interface.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "call/video_receive_stream.h"