webrtc_m130/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
danilchap 799a9d017a Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
Reason for revert:
breaks downstream code

Original issue's description:
> Remove unnecessary interface TelephoneEventHandler.
>
> BUG=webrtc:2795
>
> Committed: https://crrev.com/2beb42983ca24e1326a9a7f2c06b3ad740eea2c3
> Cr-Commit-Position: refs/heads/master@{#14346}

TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2362673002
Cr-Commit-Position: refs/heads/master@{#14348}
2016-09-22 10:36:34 +00:00

93 lines
3.3 KiB
C++

/*
* Copyright (c) 2013 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_SOURCE_RTP_RECEIVER_IMPL_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
#include <memory>
#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RtpReceiverImpl : public RtpReceiver {
public:
// Callbacks passed in here may not be NULL (use Null Object callbacks if you
// want callbacks to do nothing). This class takes ownership of the media
// receiver but nothing else.
RtpReceiverImpl(Clock* clock,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry,
RTPReceiverStrategy* rtp_media_receiver);
virtual ~RtpReceiverImpl();
int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
const size_t channels,
const uint32_t rate) override;
int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
bool IncomingRtpPacket(const RTPHeader& rtp_header,
const uint8_t* payload,
size_t payload_length,
PayloadUnion payload_specific,
bool in_order) override;
// Returns the last received timestamp.
bool Timestamp(uint32_t* timestamp) const override;
bool LastReceivedTimeMs(int64_t* receive_time_ms) const override;
uint32_t SSRC() const override;
int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
TelephoneEventHandler* GetTelephoneEventHandler() override;
private:
bool HaveReceivedFrame() const;
void CheckSSRCChanged(const RTPHeader& rtp_header);
void CheckCSRC(const WebRtcRTPHeader& rtp_header);
int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
const int8_t first_payload_byte,
bool* is_red,
PayloadUnion* payload);
Clock* clock_;
RTPPayloadRegistry* rtp_payload_registry_;
std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
RtpFeedback* cb_rtp_feedback_;
rtc::CriticalSection critical_section_rtp_receiver_;
int64_t last_receive_time_;
size_t last_received_payload_length_;
// SSRCs.
uint32_t ssrc_;
uint8_t num_csrcs_;
uint32_t current_remote_csrc_[kRtpCsrcSize];
uint32_t last_received_timestamp_;
int64_t last_received_frame_time_ms_;
uint16_t last_received_sequence_number_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_