mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-19 08:37:54 +01:00
c62f6c7121
This is needed in the general case, now that we aim to support codecs other than those built-in to WebRTC. BUG=webrtc:8159 Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706 Reviewed-on: https://webrtc-review.googlesource.com/5380 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20136}
57 lines
2 KiB
C++
57 lines
2 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 MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
|
|
#define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
|
|
|
|
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
|
|
#include "modules/rtp_rtcp/source/rtp_utility.h"
|
|
#include "rtc_base/onetimeevent.h"
|
|
#include "typedefs.h" // NOLINT(build/include)
|
|
|
|
namespace webrtc {
|
|
|
|
class RTPReceiverVideo : public RTPReceiverStrategy {
|
|
public:
|
|
explicit RTPReceiverVideo(RtpData* data_callback);
|
|
|
|
virtual ~RTPReceiverVideo();
|
|
|
|
int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
|
|
const PayloadUnion& specific_payload,
|
|
bool is_red,
|
|
const uint8_t* packet,
|
|
size_t packet_length,
|
|
int64_t timestamp) override;
|
|
|
|
TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }
|
|
|
|
RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
|
|
|
|
bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
|
|
|
|
int32_t OnNewPayloadTypeCreated(int payload_type,
|
|
const SdpAudioFormat& audio_format) override;
|
|
|
|
int32_t InvokeOnInitializeDecoder(
|
|
RtpFeedback* callback,
|
|
int8_t payload_type,
|
|
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
|
const PayloadUnion& specific_payload) const override;
|
|
|
|
void SetPacketOverHead(uint16_t packet_over_head);
|
|
|
|
private:
|
|
OneTimeEvent first_packet_received_;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
|