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}
58 lines
2.0 KiB
C++
58 lines
2.0 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/base/onetimeevent.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/modules/rtp_rtcp/source/rtp_utility.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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explicit RTPReceiverVideo(RtpData* data_callback);
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virtual ~RTPReceiverVideo();
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int32_t ParseRtpPacket(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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size_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() override { return NULL; }
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RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
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bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
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int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) override;
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int32_t InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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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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private:
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OneTimeEvent first_packet_received_;
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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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