Removed map_wrapper from rtp_sender
Review URL: https://webrtc-codereview.appspot.com/343014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1478 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
parent
3c9be1bc4d
commit
0074187436
@ -101,44 +101,26 @@ RTPSender::RTPSender(const WebRtc_Word32 id,
|
||||
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
|
||||
}
|
||||
|
||||
RTPSender::~RTPSender()
|
||||
{
|
||||
if(_remoteSSRC != 0)
|
||||
{
|
||||
_ssrcDB.ReturnSSRC(_remoteSSRC);
|
||||
}
|
||||
_ssrcDB.ReturnSSRC(_ssrc);
|
||||
RTPSender::~RTPSender() {
|
||||
if(_remoteSSRC != 0) {
|
||||
_ssrcDB.ReturnSSRC(_remoteSSRC);
|
||||
}
|
||||
_ssrcDB.ReturnSSRC(_ssrc);
|
||||
|
||||
SSRCDatabase::ReturnSSRCDatabase();
|
||||
SSRCDatabase::ReturnSSRCDatabase();
|
||||
delete _sendCritsect;
|
||||
delete _transportCritsect;
|
||||
while (!_payloadTypeMap.empty()) {
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
|
||||
_payloadTypeMap.begin();
|
||||
delete it->second;
|
||||
_payloadTypeMap.erase(it);
|
||||
}
|
||||
delete _packetHistory;
|
||||
delete _audio;
|
||||
delete _video;
|
||||
|
||||
delete _sendCritsect;
|
||||
delete _transportCritsect;
|
||||
|
||||
// empty map
|
||||
bool loop = true;
|
||||
do
|
||||
{
|
||||
MapItem* item = _payloadTypeMap.First();
|
||||
if(item)
|
||||
{
|
||||
// delete
|
||||
ModuleRTPUtility::Payload* payload= ((ModuleRTPUtility::Payload*)item->GetItem());
|
||||
delete payload;
|
||||
|
||||
// remove from map and delete Item
|
||||
_payloadTypeMap.Erase(item);
|
||||
} else
|
||||
{
|
||||
loop = false;
|
||||
}
|
||||
} while (loop);
|
||||
|
||||
delete _packetHistory;
|
||||
|
||||
delete _audio;
|
||||
delete _video;
|
||||
|
||||
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
|
||||
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
|
||||
}
|
||||
|
||||
WebRtc_Word32
|
||||
@ -171,20 +153,12 @@ RTPSender::Init(const WebRtc_UWord32 remoteSSRC)
|
||||
|
||||
_rtpHeaderExtensionMap.Erase();
|
||||
|
||||
bool loop = true;
|
||||
do
|
||||
{
|
||||
MapItem* item = _payloadTypeMap.First();
|
||||
if(item)
|
||||
{
|
||||
ModuleRTPUtility::Payload* payload= ((ModuleRTPUtility::Payload*)item->GetItem());
|
||||
delete payload;
|
||||
_payloadTypeMap.Erase(item);
|
||||
} else
|
||||
{
|
||||
loop = false;
|
||||
}
|
||||
} while (loop);
|
||||
while (!_payloadTypeMap.empty()) {
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
|
||||
_payloadTypeMap.begin();
|
||||
delete it->second;
|
||||
_payloadTypeMap.erase(it);
|
||||
}
|
||||
|
||||
memset(_CSRC, 0, sizeof(_CSRC));
|
||||
|
||||
@ -297,89 +271,80 @@ RTPSender::RtpHeaderExtensionTotalLength() const
|
||||
}
|
||||
|
||||
//can be called multiple times
|
||||
WebRtc_Word32
|
||||
RTPSender::RegisterPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
|
||||
const WebRtc_Word8 payloadNumber,
|
||||
const WebRtc_UWord32 frequency,
|
||||
const WebRtc_UWord8 channels,
|
||||
const WebRtc_UWord32 rate)
|
||||
{
|
||||
if (!payloadName)
|
||||
{
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
|
||||
return -1;
|
||||
}
|
||||
WebRtc_Word32 RTPSender::RegisterPayload(
|
||||
const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
|
||||
const WebRtc_Word8 payloadNumber,
|
||||
const WebRtc_UWord32 frequency,
|
||||
const WebRtc_UWord8 channels,
|
||||
const WebRtc_UWord32 rate) {
|
||||
if (!payloadName) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument",
|
||||
__FUNCTION__);
|
||||
return -1;
|
||||
}
|
||||
CriticalSectionScoped cs(_sendCritsect);
|
||||
|
||||
CriticalSectionScoped cs(_sendCritsect);
|
||||
if (payloadNumber == _keepAlivePayloadType) {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "invalid state",
|
||||
__FUNCTION__);
|
||||
return -1;
|
||||
}
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
|
||||
_payloadTypeMap.find(payloadNumber);
|
||||
|
||||
if(payloadNumber == _keepAlivePayloadType)
|
||||
{
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "invalid state", __FUNCTION__);
|
||||
return -1;
|
||||
}
|
||||
if (_payloadTypeMap.end() != it) {
|
||||
// we already use this payload type
|
||||
ModuleRTPUtility::Payload* payload = it->second;
|
||||
assert(payload);
|
||||
|
||||
MapItem* item = _payloadTypeMap.Find(payloadNumber);
|
||||
if( NULL != item)
|
||||
{
|
||||
// we already use this payload type
|
||||
|
||||
ModuleRTPUtility::Payload* payload = (ModuleRTPUtility::Payload*)item->GetItem();
|
||||
assert(payload);
|
||||
|
||||
// check if it's the same as we already have
|
||||
WebRtc_Word32 payloadNameLength = (WebRtc_Word32)strlen(payloadName);
|
||||
WebRtc_Word32 nameLength = (WebRtc_Word32)strlen(payload->name);
|
||||
if(payloadNameLength == nameLength && ModuleRTPUtility::StringCompare(payload->name, payloadName, nameLength))
|
||||
{
|
||||
if(_audioConfigured && payload->audio &&
|
||||
payload->typeSpecific.Audio.frequency == frequency &&
|
||||
(payload->typeSpecific.Audio.rate == rate || payload->typeSpecific.Audio.rate == 0 || rate == 0))
|
||||
{
|
||||
payload->typeSpecific.Audio.rate = rate; // Ensure that we update the rate if new or old is zero
|
||||
return 0;
|
||||
}
|
||||
if(!_audioConfigured && !payload->audio)
|
||||
{
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
return -1;
|
||||
}
|
||||
|
||||
WebRtc_Word32 retVal = -1;
|
||||
ModuleRTPUtility::Payload* payload = NULL;
|
||||
|
||||
if(_audioConfigured)
|
||||
{
|
||||
retVal = _audio->RegisterAudioPayload(payloadName, payloadNumber, frequency, channels, rate, payload);
|
||||
} else
|
||||
{
|
||||
retVal = _video->RegisterVideoPayload(payloadName, payloadNumber, rate, payload);
|
||||
}
|
||||
if(payload)
|
||||
{
|
||||
_payloadTypeMap.Insert(payloadNumber, payload);
|
||||
}
|
||||
return retVal;
|
||||
}
|
||||
|
||||
WebRtc_Word32
|
||||
RTPSender::DeRegisterSendPayload(const WebRtc_Word8 payloadType)
|
||||
{
|
||||
CriticalSectionScoped lock(_sendCritsect);
|
||||
|
||||
MapItem* item = _payloadTypeMap.Find(payloadType);
|
||||
if( NULL != item)
|
||||
{
|
||||
ModuleRTPUtility::Payload* payload = (ModuleRTPUtility::Payload*)item->GetItem();
|
||||
delete payload;
|
||||
|
||||
_payloadTypeMap.Erase(item);
|
||||
// check if it's the same as we already have
|
||||
WebRtc_Word32 payloadNameLength = (WebRtc_Word32)strlen(payloadName);
|
||||
WebRtc_Word32 nameLength = (WebRtc_Word32)strlen(payload->name);
|
||||
if (payloadNameLength == nameLength &&
|
||||
ModuleRTPUtility::StringCompare(payload->name, payloadName,
|
||||
nameLength)) {
|
||||
if (_audioConfigured && payload->audio &&
|
||||
payload->typeSpecific.Audio.frequency == frequency &&
|
||||
(payload->typeSpecific.Audio.rate == rate ||
|
||||
payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
|
||||
payload->typeSpecific.Audio.rate = rate;
|
||||
// Ensure that we update the rate if new or old is zero
|
||||
return 0;
|
||||
}
|
||||
if(!_audioConfigured && !payload->audio) {
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
return -1;
|
||||
}
|
||||
WebRtc_Word32 retVal = -1;
|
||||
ModuleRTPUtility::Payload* payload = NULL;
|
||||
if (_audioConfigured) {
|
||||
retVal = _audio->RegisterAudioPayload(payloadName, payloadNumber, frequency,
|
||||
channels, rate, payload);
|
||||
} else {
|
||||
retVal = _video->RegisterVideoPayload(payloadName, payloadNumber, rate,
|
||||
payload);
|
||||
}
|
||||
if(payload) {
|
||||
_payloadTypeMap[payloadNumber] = payload;
|
||||
}
|
||||
return retVal;
|
||||
}
|
||||
|
||||
WebRtc_Word32 RTPSender::DeRegisterSendPayload(const WebRtc_Word8 payloadType) {
|
||||
CriticalSectionScoped lock(_sendCritsect);
|
||||
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
|
||||
_payloadTypeMap.find(payloadType);
|
||||
|
||||
if (_payloadTypeMap.end() == it) return -1;
|
||||
|
||||
ModuleRTPUtility::Payload* payload = it->second;
|
||||
delete payload;
|
||||
_payloadTypeMap.erase(it);
|
||||
return 0;
|
||||
}
|
||||
|
||||
WebRtc_Word8 RTPSender::SendPayloadType() const
|
||||
{
|
||||
@ -423,23 +388,24 @@ RTPSender::RTPKeepaliveStatus(bool* enable,
|
||||
return 0;
|
||||
}
|
||||
|
||||
WebRtc_Word32
|
||||
RTPSender::EnableRTPKeepalive( const WebRtc_Word8 unknownPayloadType,
|
||||
const WebRtc_UWord16 deltaTransmitTimeMS)
|
||||
{
|
||||
CriticalSectionScoped cs(_sendCritsect);
|
||||
WebRtc_Word32 RTPSender::EnableRTPKeepalive(
|
||||
const WebRtc_Word8 unknownPayloadType,
|
||||
const WebRtc_UWord16 deltaTransmitTimeMS) {
|
||||
CriticalSectionScoped cs(_sendCritsect);
|
||||
|
||||
if( NULL != _payloadTypeMap.Find(unknownPayloadType))
|
||||
{
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
|
||||
return -1;
|
||||
}
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
|
||||
_payloadTypeMap.find(unknownPayloadType);
|
||||
|
||||
_keepAliveIsActive = true;
|
||||
_keepAlivePayloadType = unknownPayloadType;
|
||||
_keepAliveLastSent = _clock.GetTimeInMS();
|
||||
_keepAliveDeltaTimeSend = deltaTransmitTimeMS;
|
||||
return 0;
|
||||
if (it != _payloadTypeMap.end()) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument",
|
||||
__FUNCTION__);
|
||||
return -1;
|
||||
}
|
||||
_keepAliveIsActive = true;
|
||||
_keepAlivePayloadType = unknownPayloadType;
|
||||
_keepAliveLastSent = _clock.GetTimeInMS();
|
||||
_keepAliveDeltaTimeSend = deltaTransmitTimeMS;
|
||||
return 0;
|
||||
}
|
||||
|
||||
WebRtc_Word32
|
||||
@ -619,92 +585,75 @@ void RTPSender::RTXStatus(bool* enable,
|
||||
*SSRC = _ssrcRTX;
|
||||
}
|
||||
|
||||
WebRtc_Word32
|
||||
RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType,
|
||||
RtpVideoCodecTypes& videoType)
|
||||
{
|
||||
CriticalSectionScoped cs(_sendCritsect);
|
||||
WebRtc_Word32 RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType,
|
||||
RtpVideoCodecTypes& videoType) {
|
||||
CriticalSectionScoped cs(_sendCritsect);
|
||||
|
||||
if(payloadType < 0)
|
||||
{
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "\tinvalid payloadType (%d)", payloadType);
|
||||
return -1;
|
||||
if (payloadType < 0) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
|
||||
"\tinvalid payloadType (%d)", payloadType);
|
||||
return -1;
|
||||
}
|
||||
if (_audioConfigured) {
|
||||
WebRtc_Word8 redPlType = -1;
|
||||
if (_audio->RED(redPlType) == 0) {
|
||||
// We have configured RED.
|
||||
if(redPlType == payloadType) {
|
||||
// And it's a match...
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
|
||||
if(_audioConfigured)
|
||||
{
|
||||
WebRtc_Word8 redPlType = -1;
|
||||
if(_audio->RED(redPlType) == 0)
|
||||
{
|
||||
// we have configured RED
|
||||
if(redPlType == payloadType)
|
||||
{
|
||||
// and it's a match
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if(_payloadType != payloadType)
|
||||
{
|
||||
MapItem* item = _payloadTypeMap.Find(payloadType);
|
||||
if( NULL == item)
|
||||
{
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "\tpayloadType:%d not registered", payloadType);
|
||||
return -1;
|
||||
}
|
||||
_payloadType = payloadType;
|
||||
ModuleRTPUtility::Payload* payload = (ModuleRTPUtility::Payload*)item->GetItem();
|
||||
if(payload)
|
||||
{
|
||||
if(payload->audio)
|
||||
{
|
||||
if(_audioConfigured)
|
||||
{
|
||||
// Extract payload frequency
|
||||
int payloadFreqHz;
|
||||
if(ModuleRTPUtility::StringCompare(payload->name,"g722",4)&&
|
||||
(payload->name[4] == 0)) //Check that strings end there, g722.1...
|
||||
{
|
||||
// Special case for G.722, bug in spec
|
||||
payloadFreqHz=8000;
|
||||
}
|
||||
else
|
||||
{
|
||||
payloadFreqHz=payload->typeSpecific.Audio.frequency;
|
||||
}
|
||||
|
||||
//we don't do anything if it's CN
|
||||
if((_audio->AudioFrequency() != payloadFreqHz)&&
|
||||
(!ModuleRTPUtility::StringCompare(payload->name,"cn",2)))
|
||||
{
|
||||
_audio->SetAudioFrequency(payloadFreqHz);
|
||||
// We need to correct the timestamp again,
|
||||
// since this might happen after we've set it
|
||||
WebRtc_UWord32 RTPtime =
|
||||
ModuleRTPUtility::GetCurrentRTP(&_clock, payloadFreqHz);
|
||||
SetStartTimestamp(RTPtime);
|
||||
// will be ignored if it's already configured via API
|
||||
}
|
||||
}
|
||||
}else
|
||||
{
|
||||
if(!_audioConfigured)
|
||||
{
|
||||
_video->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
|
||||
videoType = payload->typeSpecific.Video.videoCodecType;
|
||||
_video->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
|
||||
}
|
||||
}
|
||||
}
|
||||
} else
|
||||
{
|
||||
if(!_audioConfigured)
|
||||
{
|
||||
videoType = _video->VideoCodecType();
|
||||
}
|
||||
}
|
||||
if (_payloadType == payloadType) {
|
||||
if (!_audioConfigured) {
|
||||
videoType = _video->VideoCodecType();
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
|
||||
_payloadTypeMap.find(payloadType);
|
||||
if (it == _payloadTypeMap.end()) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
|
||||
"\tpayloadType:%d not registered", payloadType);
|
||||
return -1;
|
||||
}
|
||||
_payloadType = payloadType;
|
||||
ModuleRTPUtility::Payload* payload = it->second;
|
||||
assert(payload);
|
||||
if (payload->audio) {
|
||||
if (_audioConfigured) {
|
||||
// Extract payload frequency
|
||||
int payloadFreqHz;
|
||||
if (ModuleRTPUtility::StringCompare(payload->name,"g722",4)&&
|
||||
(payload->name[4] == 0)) {
|
||||
//Check that strings end there, g722.1...
|
||||
// Special case for G.722, bug in spec
|
||||
payloadFreqHz=8000;
|
||||
} else {
|
||||
payloadFreqHz=payload->typeSpecific.Audio.frequency;
|
||||
}
|
||||
|
||||
//we don't do anything if it's CN
|
||||
if ((_audio->AudioFrequency() != payloadFreqHz)&&
|
||||
(!ModuleRTPUtility::StringCompare(payload->name,"cn",2))) {
|
||||
_audio->SetAudioFrequency(payloadFreqHz);
|
||||
// We need to correct the timestamp again,
|
||||
// since this might happen after we've set it
|
||||
WebRtc_UWord32 RTPtime =
|
||||
ModuleRTPUtility::GetCurrentRTP(&_clock, payloadFreqHz);
|
||||
SetStartTimestamp(RTPtime);
|
||||
// will be ignored if it's already configured via API
|
||||
}
|
||||
}
|
||||
} else {
|
||||
if(!_audioConfigured) {
|
||||
_video->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
|
||||
videoType = payload->typeSpecific.Video.videoCodecType;
|
||||
_video->SetMaxConfiguredBitrateVideo(
|
||||
payload->typeSpecific.Video.maxRate);
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
WebRtc_Word32
|
||||
|
||||
@ -11,20 +11,19 @@
|
||||
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
|
||||
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
|
||||
|
||||
#include <cassert>
|
||||
#include <cmath>
|
||||
#include <map>
|
||||
|
||||
#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
|
||||
#include "rtp_rtcp_defines.h"
|
||||
#include "common_types.h" // Encryption
|
||||
#include "ssrc_database.h"
|
||||
#include "list_wrapper.h"
|
||||
#include "map_wrapper.h"
|
||||
#include "Bitrate.h"
|
||||
#include "rtp_header_extension.h"
|
||||
#include "video_codec_information.h"
|
||||
#include "transmission_bucket.h"
|
||||
|
||||
#include <cassert>
|
||||
#include <cmath>
|
||||
|
||||
#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1
|
||||
|
||||
namespace webrtc {
|
||||
@ -331,7 +330,7 @@ private:
|
||||
WebRtc_UWord16 _packetOverHead;
|
||||
|
||||
WebRtc_Word8 _payloadType;
|
||||
MapWrapper _payloadTypeMap;
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
|
||||
|
||||
RtpHeaderExtensionMap _rtpHeaderExtensionMap;
|
||||
WebRtc_Word32 _transmissionTimeOffset;
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user