dc80bae2a6
Clean some logs and add asserts in the way. BUG=3153 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
87 lines
2.8 KiB
C++
87 lines
2.8 KiB
C++
/*
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* Copyright (c) 2012 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_VIDEO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RTPReceiverVideo : public RTPReceiverStrategy {
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public:
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RTPReceiverVideo(RtpData* data_callback);
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virtual ~RTPReceiverVideo();
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virtual int32_t ParseRtpPacket(
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WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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bool is_red,
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const uint8_t* packet,
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uint16_t packet_length,
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int64_t timestamp,
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bool is_first_packet) OVERRIDE;
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TelephoneEventHandler* GetTelephoneEventHandler() {
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return NULL;
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}
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int GetPayloadTypeFrequency() const OVERRIDE;
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virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const
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OVERRIDE;
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virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE;
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virtual int32_t OnNewPayloadTypeCreated(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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int8_t payload_type,
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uint32_t frequency) OVERRIDE;
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virtual int32_t InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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int32_t id,
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int8_t payload_type,
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const PayloadUnion& specific_payload) const OVERRIDE;
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void SetPacketOverHead(uint16_t packet_over_head);
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protected:
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int32_t ReceiveGenericCodec(WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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uint16_t payload_data_length);
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int32_t ReceiveVp8Codec(WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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uint16_t payload_data_length);
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int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
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uint8_t* data_buffer) const;
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private:
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int32_t ParseVideoCodecSpecific(
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WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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uint16_t payload_data_length,
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RtpVideoCodecTypes video_type,
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int64_t now_ms,
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bool is_first_packet);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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