mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 14:50:39 +01:00
a136ed4085
Setting the transport cc flag was only possible post creation for audio receive streams, while video receive streams need to be recreated. This CL moves the setter for transport_cc() to where the getter is and adds boiler plate implementations for the video streams. For audio streams this splits "SetUseTransportCcAndNackHistory" into two methods, SetTransportCc and SetNackHistory. Bug: none Change-Id: Idbec8217aef10ee77907cebaecdc27b4b0fb18e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264443 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37038}
101 lines
3.7 KiB
C++
101 lines
3.7 KiB
C++
/*
|
|
* Copyright (c) 2021 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 CALL_RECEIVE_STREAM_H_
|
|
#define CALL_RECEIVE_STREAM_H_
|
|
|
|
#include <vector>
|
|
|
|
#include "api/crypto/frame_decryptor_interface.h"
|
|
#include "api/frame_transformer_interface.h"
|
|
#include "api/media_types.h"
|
|
#include "api/scoped_refptr.h"
|
|
#include "api/transport/rtp/rtp_source.h"
|
|
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Common base interface for MediaReceiveStreamInterface based classes and
|
|
// FlexfecReceiveStream.
|
|
class ReceiveStreamInterface {
|
|
public:
|
|
// Receive-stream specific RTP settings.
|
|
// TODO(tommi): This struct isn't needed at this level anymore. Move it closer
|
|
// to where it's used.
|
|
struct ReceiveStreamRtpConfig {
|
|
// Synchronization source (stream identifier) to be received.
|
|
// This member will not change mid-stream and can be assumed to be const
|
|
// post initialization.
|
|
uint32_t remote_ssrc = 0;
|
|
|
|
// Sender SSRC used for sending RTCP (such as receiver reports).
|
|
// This value may change mid-stream and must be done on the same thread
|
|
// that the value is read on (i.e. packet delivery).
|
|
uint32_t local_ssrc = 0;
|
|
|
|
// Enable feedback for send side bandwidth estimation.
|
|
// See
|
|
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
|
|
// for details.
|
|
// This value may change mid-stream and must be done on the same thread
|
|
// that the value is read on (i.e. packet delivery).
|
|
bool transport_cc = false;
|
|
|
|
// RTP header extensions used for the received stream.
|
|
// This value may change mid-stream and must be done on the same thread
|
|
// that the value is read on (i.e. packet delivery).
|
|
std::vector<RtpExtension> extensions;
|
|
};
|
|
|
|
// Set/change the rtp header extensions. Must be called on the packet
|
|
// delivery thread.
|
|
virtual void SetRtpExtensions(std::vector<RtpExtension> extensions) = 0;
|
|
virtual RtpHeaderExtensionMap GetRtpExtensionMap() const = 0;
|
|
|
|
// Returns a bool for whether feedback for send side bandwidth estimation is
|
|
// enabled. See
|
|
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
|
|
// for details.
|
|
// This value may change mid-stream and must be done on the same thread
|
|
// that the value is read on (i.e. packet delivery).
|
|
virtual bool transport_cc() const = 0;
|
|
|
|
virtual void SetTransportCc(bool transport_cc) = 0;
|
|
|
|
protected:
|
|
virtual ~ReceiveStreamInterface() {}
|
|
};
|
|
|
|
// Either an audio or video receive stream.
|
|
class MediaReceiveStreamInterface : public ReceiveStreamInterface {
|
|
public:
|
|
// Starts stream activity.
|
|
// When a stream is active, it can receive, process and deliver packets.
|
|
virtual void Start() = 0;
|
|
|
|
// Stops stream activity. Must be called to match with a previous call to
|
|
// `Start()`. When a stream has been stopped, it won't receive, decode,
|
|
// process or deliver packets to downstream objects such as callback pointers
|
|
// set in the config struct.
|
|
virtual void Stop() = 0;
|
|
|
|
virtual void SetDepacketizerToDecoderFrameTransformer(
|
|
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
|
|
frame_transformer) = 0;
|
|
|
|
virtual void SetFrameDecryptor(
|
|
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
|
|
|
|
virtual std::vector<RtpSource> GetSources() const = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // CALL_RECEIVE_STREAM_H_
|