blob: 6d38cf7e4d6287e847e0aab31153aceda2081881 [file] [log] [blame]
* 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.
#include <list>
#include <memory>
#include <unordered_map>
#include <vector>
#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/rtc_base/criticalsection.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RtpReceiverImpl : public RtpReceiver {
// 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 CodecInst& audio_codec) override;
int32_t RegisterReceivePayload(const VideoCodec& video_codec) 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;
std::vector<RtpSource> GetSources() const override;
const std::vector<RtpSource>& ssrc_sources_for_testing() const {
return ssrc_sources_;
const std::list<RtpSource>& csrc_sources_for_testing() const {
return csrc_sources_;
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);
void UpdateSources(const rtc::Optional<uint8_t>& ssrc_audio_level);
void RemoveOutdatedSources(int64_t now_ms);
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_;
std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
// The RtpSource objects are sorted chronologically.
std::list<RtpSource> csrc_sources_;
std::vector<RtpSource> ssrc_sources_;
} // namespace webrtc