generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Changed RTP reveivers to use stl map and list.

Browse Source 
Review URL: https://webrtc-codereview.appspot.com/349010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1475 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2012-01-19 15:53:59 +00:00
parent 38f4816737
commit af6f15c1bf
5 changed files with 551 additions and 638 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1122
src/modules/rtp_rtcp/source/rtp_receiver.cc
View File

File diff suppressed because it is too large Load Diff

5
src/modules/rtp_rtcp/source/rtp_receiver.h
View File

@ -11,6 +11,8 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
#include <map>
#include "typedefs.h" #include "typedefs.h"
#include "rtp_utility.h" #include "rtp_utility.h"
@ -208,8 +210,7 @@ private:
WebRtc_Word8 _redPayloadType; WebRtc_Word8 _redPayloadType;
// std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
MapWrapper _payloadTypeMap;
RtpHeaderExtensionMap _rtpHeaderExtensionMap; RtpHeaderExtensionMap _rtpHeaderExtensionMap;
// SSRCs // SSRCs

49
src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
View File

@ -24,7 +24,6 @@ RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id):
_telephoneEventForwardToDecoder(false), _telephoneEventForwardToDecoder(false),
_telephoneEventDetectEndOfTone(false), _telephoneEventDetectEndOfTone(false),
_telephoneEventPayloadType(-1), _telephoneEventPayloadType(-1),
_telephoneEventReported(),
_cngNBPayloadType(-1), _cngNBPayloadType(-1),
_cngWBPayloadType(-1), _cngWBPayloadType(-1),
_cngSWBPayloadType(-1), _cngSWBPayloadType(-1),
@ -41,26 +40,22 @@ RTPReceiverAudio::~RTPReceiverAudio()
delete _criticalSectionFeedback; delete _criticalSectionFeedback;
} }
WebRtc_Word32 WebRtc_Word32 RTPReceiverAudio::Init() {
RTPReceiverAudio::Init() _lastReceivedFrequency = 8000;
{ _telephoneEvent = false;
_lastReceivedFrequency = 8000; _telephoneEventForwardToDecoder = false;
_telephoneEvent = false; _telephoneEventDetectEndOfTone = false;
_telephoneEventForwardToDecoder = false; _telephoneEventPayloadType = -1;
_telephoneEventDetectEndOfTone = false;
_telephoneEventPayloadType = -1;
while(_telephoneEventReported.Size() > 0) _telephoneEventReported.clear();
{
_telephoneEventReported.Erase(_telephoneEventReported.First()); _cngNBPayloadType = -1;
} _cngWBPayloadType = -1;
_cngNBPayloadType = -1; _cngSWBPayloadType = -1;
_cngWBPayloadType = -1; _cngPayloadType = -1;
_cngSWBPayloadType = -1; _G722PayloadType = -1;
_cngPayloadType = -1; _lastReceivedG722 = false;
_G722PayloadType = -1; return 0;
_lastReceivedG722 = false;
return 0;
} }
void void
@ -342,14 +337,17 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
{ {
bool end = (payloadData[(4*n)+1] & 0x80)? true:false; bool end = (payloadData[(4*n)+1] & 0x80)? true:false;
if(_telephoneEventReported.Find(payloadData[4*n]) != NULL) std::set<WebRtc_UWord8>::iterator event =
_telephoneEventReported.find(payloadData[4*n]);
if(event != _telephoneEventReported.end())
{ {
// we have already seen this event // we have already seen this event
if(end) if(end)
{ {
removedEvents[numberOfRemovedEvents]= payloadData[4*n]; removedEvents[numberOfRemovedEvents]= payloadData[4*n];
numberOfRemovedEvents++; numberOfRemovedEvents++;
_telephoneEventReported.Erase(payloadData[4*n]); _telephoneEventReported.erase(payloadData[4*n]);
} }
}else }else
{ {
@ -360,7 +358,7 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
{ {
newEvents[numberOfNewEvents] = payloadData[4*n]; newEvents[numberOfNewEvents] = payloadData[4*n];
numberOfNewEvents++; numberOfNewEvents++;
_telephoneEventReported.Insert(payloadData[4*n],NULL); _telephoneEventReported.insert(payloadData[4*n]);
} }
} }
} }
@ -411,8 +409,9 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
// don't forward event to decoder // don't forward event to decoder
return 0; return 0;
} }
MapItem* first = _telephoneEventReported.First(); std::set<WebRtc_UWord8>::iterator first =
if(first && first->GetId() > 15) _telephoneEventReported.begin();
if(first != _telephoneEventReported.end() && *first > 15)
{ {
// don't forward non DTMF events // don't forward non DTMF events
return 0; return 0;

7
src/modules/rtp_rtcp/source/rtp_receiver_audio.h
View File

@ -11,11 +11,12 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#include <set>
#include "rtp_rtcp_defines.h" #include "rtp_rtcp_defines.h"
#include "rtp_utility.h" #include "rtp_utility.h"
#include "typedefs.h" #include "typedefs.h"
#include "map_wrapper.h"
namespace webrtc { namespace webrtc {
class CriticalSectionWrapper; class CriticalSectionWrapper;
@ -77,8 +78,8 @@ private:
bool _telephoneEvent; bool _telephoneEvent;
bool _telephoneEventForwardToDecoder; bool _telephoneEventForwardToDecoder;
bool _telephoneEventDetectEndOfTone; bool _telephoneEventDetectEndOfTone;
WebRtc_Word8 _telephoneEventPayloadType; WebRtc_Word8 _telephoneEventPayloadType;
MapWrapper _telephoneEventReported; std::set<WebRtc_UWord8> _telephoneEventReported;
WebRtc_Word8 _cngNBPayloadType; WebRtc_Word8 _cngNBPayloadType;
WebRtc_Word8 _cngWBPayloadType; WebRtc_Word8 _cngWBPayloadType;

6
src/modules/rtp_rtcp/source/rtp_receiver_video.h
View File

@ -15,8 +15,6 @@
#include "rtp_utility.h" #include "rtp_utility.h"
#include "typedefs.h" #include "typedefs.h"
#include "map_wrapper.h"
#include "list_wrapper.h"
#include "overuse_detector.h" #include "overuse_detector.h"
#include "remote_rate_control.h" #include "remote_rate_control.h"
@ -25,12 +23,12 @@
namespace webrtc { namespace webrtc {
class ReceiverFEC; class ReceiverFEC;
class ModuleRtpRtcpImpl; class ModuleRtpRtcpImpl;
class CriticalSectionWrapper;
class RTPReceiverVideo class RTPReceiverVideo
{ {
public: public:
RTPReceiverVideo(const WebRtc_Word32 id, RTPReceiverVideo(const WebRtc_Word32 id, ModuleRtpRtcpImpl* owner);
ModuleRtpRtcpImpl* owner);
virtual ~RTPReceiverVideo(); virtual ~RTPReceiverVideo();