7bb8f02274
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
102 lines
3.7 KiB
C++
102 lines
3.7 KiB
C++
/*
|
|
* Copyright (c) 2012 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 WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
|
|
#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
|
|
|
|
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RTPPayloadRegistry;
|
|
|
|
class TelephoneEventHandler {
|
|
public:
|
|
virtual ~TelephoneEventHandler() {}
|
|
|
|
// The following three methods implement the TelephoneEventHandler interface.
|
|
// Forward DTMFs to decoder for playout.
|
|
virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
|
|
|
|
// Is forwarding of outband telephone events turned on/off?
|
|
virtual bool TelephoneEventForwardToDecoder() const = 0;
|
|
|
|
// Is TelephoneEvent configured with payload type payload_type
|
|
virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
|
|
};
|
|
|
|
class RtpReceiver {
|
|
public:
|
|
// Creates a video-enabled RTP receiver.
|
|
static RtpReceiver* CreateVideoReceiver(
|
|
int id, Clock* clock,
|
|
RtpData* incoming_payload_callback,
|
|
RtpFeedback* incoming_messages_callback,
|
|
RTPPayloadRegistry* rtp_payload_registry);
|
|
|
|
// Creates an audio-enabled RTP receiver.
|
|
static RtpReceiver* CreateAudioReceiver(
|
|
int id, Clock* clock,
|
|
RtpAudioFeedback* incoming_audio_feedback,
|
|
RtpData* incoming_payload_callback,
|
|
RtpFeedback* incoming_messages_callback,
|
|
RTPPayloadRegistry* rtp_payload_registry);
|
|
|
|
virtual ~RtpReceiver() {}
|
|
|
|
// Returns a TelephoneEventHandler if available.
|
|
virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
|
|
|
|
// Registers a receive payload in the payload registry and notifies the media
|
|
// receiver strategy.
|
|
virtual int32_t RegisterReceivePayload(
|
|
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
|
const int8_t payload_type,
|
|
const uint32_t frequency,
|
|
const uint8_t channels,
|
|
const uint32_t rate) = 0;
|
|
|
|
// De-registers |payload_type| from the payload registry.
|
|
virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
|
|
|
|
// Parses the media specific parts of an RTP packet and updates the receiver
|
|
// state. This for instance means that any changes in SSRC and payload type is
|
|
// detected and acted upon.
|
|
virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
|
|
const uint8_t* payload,
|
|
int payload_length,
|
|
PayloadUnion payload_specific,
|
|
bool in_order) = 0;
|
|
|
|
// Returns the currently configured NACK method.
|
|
virtual NACKMethod NACK() const = 0;
|
|
|
|
// Turn negative acknowledgement (NACK) requests on/off.
|
|
virtual void SetNACKStatus(const NACKMethod method) = 0;
|
|
|
|
// Returns the last received timestamp.
|
|
virtual uint32_t Timestamp() const = 0;
|
|
// Returns the time in milliseconds when the last timestamp was received.
|
|
virtual int32_t LastReceivedTimeMs() const = 0;
|
|
|
|
// Returns the remote SSRC of the currently received RTP stream.
|
|
virtual uint32_t SSRC() const = 0;
|
|
|
|
// Returns the current remote CSRCs.
|
|
virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
|
|
|
|
// Returns the current energy of the RTP stream received.
|
|
virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
|