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Replaced the _audio parameter with a strategy.

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The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org
2012-12-18 15:40:53 +00:00
parent 59ad541e57
commit 07bf43c673
13 changed files with 581 additions and 242 deletions
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20
webrtc/modules/rtp_rtcp/source/rtp_receiver.h
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@ -19,8 +19,6 @@
#include "rtp_header_extension.h"
#include "rtp_rtcp.h"
#include "rtp_rtcp_defines.h"
#include "rtp_receiver_audio.h"
#include "rtp_receiver_video.h"
#include "rtcp_receiver_help.h"
#include "Bitrate.h"
@ -28,6 +26,11 @@ namespace webrtc {
class RtpRtcpFeedback;
class ModuleRtpRtcpImpl;
class Trace;
class RTPReceiverAudio;
class RTPReceiverVideo;
class RTPReceiverStrategy;
const WebRtc_Word32 kDefaultVideoFrequency = 90000;
class RTPReceiver : public Bitrate
{
@ -186,8 +189,7 @@ private:
WebRtc_Word32 CheckPayloadChanged(const WebRtcRTPHeader* rtpHeader,
const WebRtc_Word8 firstPayloadByte,
bool& isRED,
ModuleRTPUtility::AudioPayload& audioSpecific,
ModuleRTPUtility::VideoPayload& videoSpecific);
ModuleRTPUtility::PayloadUnion* payload);
void UpdateNACKBitRate(WebRtc_Word32 bytes, WebRtc_UWord32 now);
bool ProcessNACKBitRate(WebRtc_UWord32 now);
@ -195,9 +197,9 @@ private:
private:
RTPReceiverAudio* _rtpReceiverAudio;
RTPReceiverVideo* _rtpReceiverVideo;
RTPReceiverStrategy* _rtpMediaReceiver;
WebRtc_Word32 _id;
const bool _audio;
ModuleRtpRtcpImpl& _rtpRtcp;
CriticalSectionWrapper* _criticalSectionCbs;
@ -210,14 +212,11 @@ private:
WebRtc_Word8 _lastReceivedPayloadType;
WebRtc_Word8 _lastReceivedMediaPayloadType;
ModuleRTPUtility::AudioPayload _lastReceivedAudioSpecific;
ModuleRTPUtility::VideoPayload _lastReceivedVideoSpecific;
WebRtc_UWord32 _packetTimeOutMS;
WebRtc_Word8 _redPayloadType;
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
RtpHeaderExtensionMap _rtpHeaderExtensionMap;
ModuleRTPUtility::PayloadTypeMap _payloadTypeMap;
RtpHeaderExtensionMap _rtpHeaderExtensionMap;
// SSRCs
WebRtc_UWord32 _SSRC;
@ -265,6 +264,7 @@ private:
bool _RTX;
WebRtc_UWord32 _ssrcRTX;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_