56124bd158
The purpose with this CL is to be able to send video codec specific information down to RTPPayloadRegistry. We already do this for audio with explicit arguments for e.g. number of channels. Instead of extracting the arguments from webrtc::CodecInst (audio) and webrtc::VideoCodec, this CL sends the types unmodified all the way down to RTPPayloadRegistry. This CL does not contain any functional changes, and is just a preparation for future CL:s. In the dependent CL https://codereview.webrtc.org/2524923002/, RTPPayloadStrategy is removed. RTPPayloadStrategy previously handled audio/video specific aspects of payload handling. After this CL, we will know if we get audio or video codecs without any dependency injection, since we have different functions with different signatures for audio vs video. BUG=webrtc:6743 TBR=mflodman Review-Url: https://codereview.webrtc.org/2523843002 Cr-Commit-Position: refs/heads/master@{#15231}
95 lines
3.4 KiB
C++
95 lines
3.4 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#include <memory>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RtpReceiverImpl : public RtpReceiver {
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public:
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// Callbacks passed in here may not be NULL (use Null Object callbacks if you
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// want callbacks to do nothing). This class takes ownership of the media
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// receiver but nothing else.
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RtpReceiverImpl(Clock* clock,
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RtpFeedback* incoming_messages_callback,
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RTPPayloadRegistry* rtp_payload_registry,
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RTPReceiverStrategy* rtp_media_receiver);
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virtual ~RtpReceiverImpl();
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int32_t RegisterReceivePayload(const CodecInst& audio_codec) override;
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// TODO(magjed): Remove once external code is updated.
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int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type,
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const uint32_t frequency,
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const size_t channels,
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const uint32_t rate) override;
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int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
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bool IncomingRtpPacket(const RTPHeader& rtp_header,
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const uint8_t* payload,
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size_t payload_length,
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PayloadUnion payload_specific,
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bool in_order) override;
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// Returns the last received timestamp.
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bool Timestamp(uint32_t* timestamp) const override;
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bool LastReceivedTimeMs(int64_t* receive_time_ms) const override;
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uint32_t SSRC() const override;
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int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
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int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
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TelephoneEventHandler* GetTelephoneEventHandler() override;
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private:
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bool HaveReceivedFrame() const;
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void CheckSSRCChanged(const RTPHeader& rtp_header);
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void CheckCSRC(const WebRtcRTPHeader& rtp_header);
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int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
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const int8_t first_payload_byte,
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bool* is_red,
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PayloadUnion* payload);
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Clock* clock_;
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RTPPayloadRegistry* rtp_payload_registry_;
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std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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RtpFeedback* cb_rtp_feedback_;
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rtc::CriticalSection critical_section_rtp_receiver_;
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int64_t last_receive_time_;
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size_t last_received_payload_length_;
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// SSRCs.
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uint32_t ssrc_;
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uint8_t num_csrcs_;
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uint32_t current_remote_csrc_[kRtpCsrcSize];
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uint32_t last_received_timestamp_;
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int64_t last_received_frame_time_ms_;
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uint16_t last_received_sequence_number_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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