LibRTMP源代码分析10：处理消息(Handle Message)
转载自：http://nkwavelet.blog.163.com/blog/static/227756038201412473745809/
函数RTMP_ClientPacket()是libRTMP中最重要的核心函数之一，主要完成各种消息的处理。
/**
* @brief 根据接收到的消息类型的不同，作出不同的处理。
* 消息类型有协议控制消息、Flv数据、Flex消息等。
* 协议控制消息的定义可参考官方协议：RTMP消息格式(第5章)
*
* @return 如果含有多媒体(音视频或metadata)数据，则返回1，否则返回0.
*/
int RTMP_ClientPacket(RTMP *r, RTMPPacket *packet)
int RTMP_ClientPacket(RTMP *r, RTMPPacket *packet)
{
int bHasMediaPacket = 0; // 是否含有音视频或metadata数据
switch (packet->m_packetType)
{
case RTMP_PACKET_TYPE_CHUNK_SIZE:
/* chunk size，设置块大小 */
HandleChangeChunkSize(r, packet);
break;
case RTMP_PACKET_TYPE_BYTES_READ_REPORT:
/* bytes read report，致谢 */
RTMP_Log(RTMP_LOGDEBUG, "%s, received: bytes read report", __FUNCTION__);
break;
case RTMP_PACKET_TYPE_CONTROL:
/* ctrl，处理用户控制消息 */
HandleCtrl(r, packet);
break;
case RTMP_PACKET_TYPE_SERVER_BW:
/* server bw，致谢窗口大小 */
HandleServerBW(r, packet);
break;
case RTMP_PACKET_TYPE_CLIENT_BW:
/* client bw，设置对等端带宽 */
HandleClientBW(r, packet);
break;
case RTMP_PACKET_TYPE_AUDIO:
/* audio data，音频数据 */
/*RTMP_Log(RTMP_LOGDEBUG, "%s, received: audio %lu bytes", __FUNCTION__, packet.m_nBodySize); */
HandleAudio(r, packet);
bHasMediaPacket = 1;
if (!r->m_mediaChannel)
r->m_mediaChannel = packet->m_nChannel;
if (!r->m_pausing)
r->m_mediaStamp = packet->m_nTimeStamp;
break;
case RTMP_PACKET_TYPE_VIDEO:
/* video data，视频数据 */
/*RTMP_Log(RTMP_LOGDEBUG, "%s, received: video %lu bytes", __FUNCTION__, packet.m_nBodySize); */
HandleVideo(r, packet);
bHasMediaPacket = 1;
if (!r->m_mediaChannel)
r->m_mediaChannel = packet->m_nChannel;
if (!r->m_pausing)
r->m_mediaStamp = packet->m_nTimeStamp;
break;
case RTMP_PACKET_TYPE_FLEX_STREAM_SEND:
/* flex stream send，AMF3编码，忽略 */
RTMP_Log(RTMP_LOGDEBUG, "%s, flex stream send, size %u bytes, not supported, ignoring",
__FUNCTION__, packet->m_nBodySize);
__FUNCTION__, packet->m_nBodySize);
break;
case RTMP_PACKET_TYPE_FLEX_SHARED_OBJECT:
/* flex shared object，AMF3编码，忽略 */
RTMP_Log(RTMP_LOGDEBUG, "%s, flex shared object, size %u bytes, not supported, ignoring",
__FUNCTION__, packet->m_nBodySize);
__FUNCTION__, packet->m_nBodySize);
break;
case RTMP_PACKET_TYPE_FLEX_MESSAGE:
/* flex message，AMF3编码 */
{
RTMP_Log(RTMP_LOGDEBUG, "%s, flex message, size %u bytes, not fully supported",
__FUNCTION__, packet->m_nBodySize);
__FUNCTION__, packet->m_nBodySize);
/*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */
/* some DEBUG code */
#if 0
RTMP_LIB_AMFObject obj;
int nRes = obj.Decode(packet.m_body+1, packet.m_nBodySize-1);
if(nRes < 0)
{
RTMP_Log(RTMP_LOGERROR, "%s, error decoding AMF3 packet", __FUNCTION__);
/*return; */
}
obj.Dump();
#endif
if (HandleInvoke(r, packet->m_body + 1, packet->m_nBodySize - 1) == 1)
bHasMediaPacket = 2;
break;
}
case RTMP_PACKET_TYPE_INFO:
/* metadata (notify)，元数据，AMF0编码 */
RTMP_Log(RTMP_LOGDEBUG, "%s, received: notify %u bytes", __FUNCTION__, packet->m_nBodySize);
if (HandleMetadata(r, packet->m_body, packet->m_nBodySize))
bHasMediaPacket = 1;
break;
case RTMP_PACKET_TYPE_SHARED_OBJECT:
/* AMF0编码，忽略 */
RTMP_Log(RTMP_LOGDEBUG, "%s, shared object, not supported, ignoring", __FUNCTION__);
break;
case RTMP_PACKET_TYPE_INVOKE:
/* invoke，AMF0编码的命令消息，例如各种控制命令：播放、暂停、停止 */
RTMP_Log(RTMP_LOGDEBUG, "%s, received: invoke %u bytes",
__FUNCTION__, packet->m_nBodySize);
__FUNCTION__, packet->m_nBodySize);
/*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */
if (HandleInvoke(r, packet->m_body, packet->m_nBodySize) == 1)
bHasMediaPacket = 2;
break;
case RTMP_PACKET_TYPE_FLASH_VIDEO:
{
/* go through FLV packets and handle metadata packets */
unsigned int pos = 0;
uint32_t nTimeStamp = packet->m_nTimeStamp;
while (pos + 11 < packet->m_nBodySize)
{
// Flv Tag = Type(1字节) + Datasize(3字节) + Timestamp(3字节) + TimestampExtended(1字节)
// + 0x000000(3字节) + Data(Datasize字节) + PreviousTagSize(4字节)
// Flv Tag = TagHeader(11字节) + Data(Datasize字节) + PreviousTagSize(4字节)
/* size without header (11) and prevTagSize (4) */
uint32_t dataSize = AMF_DecodeInt24(packet->m_body + pos + 1);
if (pos + 11 + dataSize + 4 > packet->m_nBodySize)
{
RTMP_Log(RTMP_LOGWARNING, "Stream corrupt?!");
break;
}
if (packet->m_body[pos] == 0x12) // Metadata Tag
{
HandleMetadata(r, packet->m_body + pos + 11, dataSize);
}
else if (packet->m_body[pos] == 8 || packet->m_body[pos] == 9) // 音频或视频Tag
{
nTimeStamp = AMF_DecodeInt24(packet->m_body + pos + 4); // 时间戳
nTimeStamp |= (packet->m_body[pos + 7] << 24); // 扩展的时间戳
}
pos += (11 + dataSize + 4);
}
if (!r->m_pausing)
r->m_mediaStamp = nTimeStamp;
/* FLV tag(s) */
/*RTMP_Log(RTMP_LOGDEBUG, "%s, received: FLV tag(s) %lu bytes",
__FUNCTION__, packet.m_nBodySize); */
__FUNCTION__, packet.m_nBodySize); */
bHasMediaPacket = 1;
break;
}
default:
RTMP_Log(RTMP_LOGDEBUG, "%s, unknown packet type received: 0x%02x",
__FUNCTION__, packet->m_packetType);
__FUNCTION__, packet->m_packetType);
#ifdef _DEBUG
RTMP_LogHex(RTMP_LOGDEBUG, packet->m_body, packet->m_nBodySize);
#endif
}
return bHasMediaPacket;
}
/**
* @brief 改变块大小（消息类型ID为0x01）
*/
static void HandleChangeChunkSize(RTMP *r, const RTMPPacket *packet)
{
if (packet->m_nBodySize >= 4)
{
r->m_inChunkSize = AMF_DecodeInt32(packet->m_body);
RTMP_Log(RTMP_LOGDEBUG, "%s, received: chunk size change to %d", __FUNCTION__, r->m_inChunkSize);
}
}
上面红色标记的函数大部分已经在前面的章节中介绍过了，下面简单介绍一下剩下的几个函数：
/**
* @brief 处理用户控制(UserControl)消息（消息类型ID为0x04）
* 用户控制消息是服务器端发出的。 可参考rtmp协议：rtmp命令消息 -- 3.7节
*/
static void HandleCtrl(RTMP *r, const RTMPPacket *packet)
{
short nType = -1;
unsigned int tmp;
if (packet->m_body && packet->m_nBodySize >= 2)
nType = AMF_DecodeInt16(packet->m_body); // 事件类型(2B)
RTMP_Log(RTMP_LOGDEBUG, "%s, received ctrl. type: %d, len: %d", __FUNCTION__, nType, packet->m_nBodySize);
/*RTMP_LogHex(packet.m_body, packet.m_nBodySize); */
if (packet->m_nBodySize >= 6)
{
switch (nType)
{
case 0: // 流开始
tmp = AMF_DecodeInt32(packet->m_body + 2); // 流ID
RTMP_Log(RTMP_LOGDEBUG, "%s, Stream Begin %d", __FUNCTION__, tmp);
break;
case 1: // 流结束
tmp = AMF_DecodeInt32(packet->m_body + 2); // 流ID
RTMP_Log(RTMP_LOGDEBUG, "%s, Stream EOF %d", __FUNCTION__, tmp);
if (r->m_pausing == 1)
r->m_pausing = 2;
break;
case 2: // 流枯竭
tmp = AMF_DecodeInt32(packet->m_body + 2); // 流ID
RTMP_Log(RTMP_LOGDEBUG, "%s, Stream Dry %d", __FUNCTION__, tmp);
break;
case 4: // 是录制流
tmp = AMF_DecodeInt32(packet->m_body + 2);
RTMP_Log(RTMP_LOGDEBUG, "%s, Stream IsRecorded %d", __FUNCTION__, tmp);
break;
case 6: // Ping客户端 /* server ping. reply with pong. */
tmp = AMF_DecodeInt32(packet->m_body + 2);
RTMP_Log(RTMP_LOGDEBUG, "%s, Ping %d", __FUNCTION__, tmp);
RTMP_SendCtrl(r, 0x07, tmp, 0);
break;
/* FMS 3.5 servers send the following two controls to let the client
* know when the server has sent a complete buffer. I.e., when the
* server has sent an amount of data equal to m_nBufferMS in duration.
* The server meters its output so that data arrives at the client
* in realtime and no faster.
*
* The rtmpdump program tries to set m_nBufferMS as large as
* possible, to force the server to send data as fast as possible.
* In practice, the server appears to cap this at about 1 hour's
* worth of data. After the server has sent a complete buffer, and
* sends this BufferEmpty message, it will wait until the play
* duration of that buffer has passed before sending a new buffer.
* The BufferReady message will be sent when the new buffer starts.
* (There is no BufferReady message for the very first buffer;
* presumably the Stream Begin message is sufficient for that
* purpose.)
*
* If the network speed is much faster than the data bitrate, then
* there may be long delays between the end of one buffer and the
* start of the next.
*
* Since usually the network allows data to be sent at
* faster than realtime, and rtmpdump wants to download the data
* as fast as possible, we use this RTMP_LF_BUFX hack: when we
* get the BufferEmpty message, we send a Pause followed by an
* Unpause. This causes the server to send the next buffer immediately
* instead of waiting for the full duration to elapse. (That's
* also the purpose of the ToggleStream function, which rtmpdump
* calls if we get a read timeout.)
*
* Media player apps don't need this hack since they are just
* going to play the data in realtime anyway. It also doesn't work
* for live streams since they obviously can only be sent in
* realtime. And it's all moot if the network speed is actually
* slower than the media bitrate.
*/
case 31:
tmp = AMF_DecodeInt32(packet->m_body + 2);
RTMP_Log(RTMP_LOGDEBUG, "%s, Stream BufferEmpty %d", __FUNCTION__, tmp);
if (!(r->Link.lFlags & RTMP_LF_BUFX))
break;
if (!r->m_pausing)
{
r->m_pauseStamp = r->m_channelTimestamp[r->m_mediaChannel];
RTMP_SendPause(r, TRUE, r->m_pauseStamp);
r->m_pausing = 1;
}
else if (r->m_pausing == 2)
{
RTMP_SendPause(r, FALSE, r->m_pauseStamp);
r->m_pausing = 3;
}
break;
case 32:
tmp = AMF_DecodeInt32(packet->m_body + 2);
RTMP_Log(RTMP_LOGDEBUG, "%s, Stream BufferReady %d", __FUNCTION__, tmp);
break;
default:
tmp = AMF_DecodeInt32(packet->m_body + 2);
RTMP_Log(RTMP_LOGDEBUG, "%s, Stream xx %d", __FUNCTION__, tmp);
break;
}
}
if (nType == 0x1A)
{
RTMP_Log(RTMP_LOGDEBUG, "%s, SWFVerification ping received: ", __FUNCTION__);
if (packet->m_nBodySize > 2 && packet->m_body[2] > 0x01)
{
RTMP_Log(RTMP_LOGERROR, "%s: SWFVerification Type %d request not supported!
Patches welcome...", __FUNCTION__, packet->m_body[2]);
Patches welcome...", __FUNCTION__, packet->m_body[2]);
}
RTMP_Log(RTMP_LOGERROR, "%s: Ignoring SWFVerification request, no CRYPTO support!", __FUNCTION__);
}
}
/**
* @brief 致谢消息的窗口大小（服务端）（消息类型ID为0x05）
*/
static void HandleServerBW(RTMP *r, const RTMPPacket *packet)
{
r->m_nServerBW = AMF_DecodeInt32(packet->m_body);
RTMP_Log(RTMP_LOGDEBUG, "%s: server BW = %d", __FUNCTION__, r->m_nServerBW);
}
/**
* @brief 对等端带宽（客户端）
*/
static void HandleClientBW(RTMP *r, const RTMPPacket *packet)
{
r->m_nClientBW = AMF_DecodeInt32(packet->m_body);
if (packet->m_nBodySize > 4)
r->m_nClientBW2 = packet->m_body[4];
// 限制类型：硬（0），软（1），或者动态（2）
else
r->m_nClientBW2 = -1;
RTMP_Log(RTMP_LOGDEBUG, "%s: client BW = %d %d", __FUNCTION__, r->m_nClientBW, r->m_nClientBW2);
}
消息类型 ID 是 0x03 的消息功能是“致谢”，没有处理函数。
消息类型 ID 是 0x08 的消息用于传输音频数据，在这里不处理。
消息类型 ID 是 0x09 的消息用于传输音频数据，在这里不处理。
消息类型 ID 是 0x03 的消息功能是“致谢”，没有处理函数。
消息类型 ID 是 0x08 的消息用于传输音频数据，在这里不处理。
消息类型 ID 是 0x09 的消息用于传输音频数据，在这里不处理。
消息类型 ID 是 0x0F、0x10 和 0x11 的消息用于传输AMF3编码的命令。
消息类型 ID 是 0x12、0x13 和 0x14 的消息用于传输AMF0编码的命令。
注意：消息类型 ID 是 0x14 的消息很重要，用于传输AMF0编码的命令，
例如各种控制命令：播放、暂停、停止等。
/**
* @brief 处理服务器发来的AMF0编码的命令.主要的步骤有：
*
* 1) 调用AMF_Decode()解码AMF命令数据
* 2) 调用AMFProp_GetString()获取具体命令的字符串
* 3) 调用AVMATCH()比较字符串，不同的命令做不同的处理
*/
static int HandleInvoke(RTMP *r, const char *body, unsigned int nBodySize)
{
AMFObject obj;
AVal method;
double txn;
int ret = 0, nRes;
// 起始字节为包类型AMF_STRING(0x02), 表示接下来是命令字符串
if (body[0] != 0x02)
/* make sure it is a string method name we start with */
{
RTMP_Log(RTMP_LOGWARNING, "%s, Sanity failed. no string method in invoke packet", __FUNCTION__);
return 0;
}
// 解码一个AMF对象
nRes =
AMF_Decode(&obj, body, nBodySize, FALSE);
if (nRes < 0)
{
RTMP_Log(RTMP_LOGERROR, "%s, error decoding invoke packet", __FUNCTION__);
return 0;
}
AMF_Dump(&obj);
AMFProp_GetString(
AMF_GetProp(&obj, NULL, 0), &method);
// 获取AMF对象obj第0个属性的AVal值
txn =
AMFProp_GetNumber(
AMF_GetProp(&obj, NULL, 1));
// 获取AMF对象obj第1个属性的double值
RTMP_Log(RTMP_LOGDEBUG, "%s, server invoking <%s>", __FUNCTION__, method.av_val);
if (AVMATCH(&method, &av__result))
{
AVal methodInvoked = {0};
int i;
for (i=0; im_numCalls; i++)
{
if (r->m_methodCalls[i].num == (int)txn)
{
methodInvoked = r->m_methodCalls[i].name;
AV_erase(r->m_methodCalls, &r->m_numCalls, i, FALSE);
break;
}
}
if (!methodInvoked.av_val)
{
RTMP_Log(RTMP_LOGDEBUG, "%s, received result id %f without matching request", __FUNCTION__, txn);
goto leave;
}
RTMP_Log(RTMP_LOGDEBUG, "%s, received result for method call <%s>",
__FUNCTION__, methodInvoked.av_val);
__FUNCTION__, methodInvoked.av_val);
if (AVMATCH(&methodInvoked, &av_connect))
{
if (r->Link.token.av_len)
{
AMFObjectProperty p;
if (
RTMP_FindFirstMatchingProperty(&obj, &av_secureToken, &p))
{
DecodeTEA(&r->Link.token, &p.p_vu.p_aval);
SendSecureTokenResponse(r, &p.p_vu.p_aval);
}
}
if (r->Link.protocol & RTMP_FEATURE_WRITE)
{
SendReleaseStream(r);
SendFCPublish(r);
}
else
{
RTMP_SendServerBW(r);
RTMP_SendCtrl(r, 3, 0, 300);
}
RTMP_SendCreateStream(r);
if (!(r->Link.protocol & RTMP_FEATURE_WRITE))
{
/* Authenticate on Justin.tv legacy servers before sending FCSubscribe */
if (r->Link.usherToken.av_len)
SendUsherToken(r, &r->Link.usherToken);
/* Send the FCSubscribe if live stream or if subscribepath is set */
if (r->Link.subscribepath.av_len)
SendFCSubscribe(r, &r->Link.subscribepath);
else if (r->Link.lFlags & RTMP_LF_LIVE)
SendFCSubscribe(r, &r->Link.playpath);
}
}
// 建立网络流，通常发生在建立网络连接(NetConnection)之后，播放(Play)之前
else if (AVMATCH(&methodInvoked, &av_createStream))
{
r->m_stream_id = (int)
AMFProp_GetNumber(AMF_GetProp(&obj, NULL, 3));
if (r->Link.protocol & RTMP_FEATURE_WRITE)
{
SendPublish(r);
}
else
{
if (r->Link.lFlags & RTMP_LF_PLST)
SendPlaylist(r);
// 获取播放列表
SendPlay(r);
// 发送命令消息“play”，开始播放流媒体数据
RTMP_SendCtrl(r, 3, r->m_stream_id, r->m_nBufferMS);
}
}
else if (AVMATCH(&methodInvoked, &av_play) || AVMATCH(&methodInvoked, &av_publish))
{
r->m_bPlaying = TRUE;
}
free(methodInvoked.av_val);
}
else if (AVMATCH(&method, &av_onBWDone))
{
if (!r->m_nBWCheckCounter)
SendCheckBW(r);
}
else if (AVMATCH(&method, &av_onFCSubscribe))
{
/* SendOnFCSubscribe(); */
}
else if (AVMATCH(&method, &av_onFCUnsubscribe))
{
RTMP_Close(r);
ret = 1;
}
else if (AVMATCH(&method, &av_ping))
{
SendPong(r, txn);
}
else if (AVMATCH(&method, &av__onbwcheck))
{
SendCheckBWResult(r, txn);
}
else if (AVMATCH(&method, &av__onbwdone))
{
int i;
for (i = 0; i < r->m_numCalls; i++)
if (AVMATCH(&r->m_methodCalls[i].name, &av__checkbw))
{
AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);
break;
}
}
else if (AVMATCH(&method, &av__error))
{
RTMP_Log(RTMP_LOGERROR, "rtmp server sent error");
}
else if (AVMATCH(&method, &av_close))
{
RTMP_Log(RTMP_LOGERROR, "rtmp server requested close");
RTMP_Close(r);
}
else if (AVMATCH(&method, &av_onStatus))
{
AMFObject obj2;
AVal code, level;
AMFProp_GetObject(
AMF_GetProp(&obj, NULL, 3), &obj2);
AMFProp_GetString(
AMF_GetProp(&obj2, &av_code, -1), &code);
AMFProp_GetString(
AMF_GetProp(&obj2, &av_level, -1), &level);
RTMP_Log(RTMP_LOGDEBUG, "%s, onStatus: %s", __FUNCTION__, code.av_val);
if (AVMATCH(&code, &av_NetStream_Failed)
|| AVMATCH(&code, &av_NetStream_Play_Failed)
|| AVMATCH(&code, &av_NetStream_Play_StreamNotFound)
|| AVMATCH(&code, &av_NetConnection_Connect_InvalidApp))
{
r->m_stream_id = -1;
RTMP_Close(r);
RTMP_Log(RTMP_LOGERROR, "Closing connection: %s", code.av_val);
}
else if (AVMATCH(&code, &av_NetStream_Play_Start) || AVMATCH(&code, &av_NetStream_Play_PublishNotify))
{
int i;
r->m_bPlaying = TRUE;
for (i = 0; i < r->m_numCalls; i++)
{
if (AVMATCH(&r->m_methodCalls[i].name, &av_play))
{
AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);
break;
}
}
}
else if (AVMATCH(&code, &av_NetStream_Publish_Start))
{
int i;
r->m_bPlaying = TRUE;
for (i = 0; i < r->m_numCalls; i++)
{
if (AVMATCH(&r->m_methodCalls[i].name, &av_publish))
{
AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);
break;
}
}
}
/* Return 1 if this is a Play.Complete or Play.Stop */
else if (AVMATCH(&code, &av_NetStream_Play_Complete)
|| AVMATCH(&code, &av_NetStream_Play_Stop)
|| AVMATCH(&code, &av_NetStream_Play_UnpublishNotify))
{
RTMP_Close(r);
ret = 1;
}
else if (AVMATCH(&code, &av_NetStream_Seek_Notify))
{
r->m_read.flags &= ~RTMP_READ_SEEKING;
}
else if (AVMATCH(&code, &av_NetStream_Pause_Notify))
{
if (r->m_pausing == 1 || r->m_pausing == 2)
{
RTMP_SendPause(r, FALSE, r->m_pauseStamp);
r->m_pausing = 3;
}
}
}
else if (AVMATCH(&method, &av_playlist_ready))
{
int i;
for (i = 0; i < r->m_numCalls; i++)
{
if (AVMATCH(&r->m_methodCalls[i].name, &av_set_playlist))
{
AV_erase(r->m_methodCalls, &r->m_numCalls, i, TRUE);
break;
}
}
}
else
{
}
leave:
AMF_Reset(&obj);
return ret;
}
/**
* @brief 查找AMF对象obj中第一个与指定name匹配的属性
* @return 查找成功返回TRUE，否则返回FALSE
*/
int RTMP_FindFirstMatchingProperty(AMFObject *obj, const AVal *name, AMFObjectProperty * p);
/**
* @brief 检测指定name是否是AMF对象obj某个属性名字的前缀.
* Like above, but only check if name is a prefix of property.
* @return 查找成功返回TRUE，否则返回FALSE
*/
int RTMP_FindPrefixProperty(AMFObject *obj, const AVal *name, AMFObjectProperty * p);
/**
* @brief 以小端字节序解码一个32位无符号整数.
*/
static int DecodeInt32LE(const char *data);
/**
* @brief 以小端字节序编码一个32位整数.
*/
static int EncodeInt32LE(char *output, int nVal);
/**
* @brief 解密用TEA加密过的密钥
*
* @param key : 用TEA加密过的密钥
* @param text : 用于保存解密出来的结果
*/
static void DecodeTEA(AVal *key, AVal *text);
/**
* @brief 将第i个方法从队列vals中删除掉
*/
static void AV_erase(RTMP_METHOD *vals, int *num, int i, int freeit)
{
if (freeit)
free(vals[i].name.av_val);
// 队列中的方法数量减一
(*num)--;
// 第i个方法之后的方法往前移动一个位置
for (; i < *num; i++)
vals[i] = vals[i + 1];
// 最后一个方法(即第*num个方法)置空
vals[i].name.av_val = NULL;
vals[i].name.av_len = 0;
vals[i].num = 0;
}
/**
* @brief 将RTMP待调用的方法队列中的第i个方法删除掉
*/
void RTMP_DropRequest(RTMP *r, int i, int freeit)
{
AV_erase(r->m_methodCalls, &r->m_numCalls, i, freeit);
}
/**
* @brief 将新方法av添加到方法队列vals中
*/
static void AV_queue(RTMP_METHOD **vals, int *num, AVal *av, int txn)
{
char *tmp;
if (!(*num & 0x0f))
*vals = realloc(*vals, (*num + 16) * sizeof(RTMP_METHOD));
tmp = malloc(av->av_len + 1);
memcpy(tmp, av->av_val, av->av_len);
tmp[av->av_len] = '\0';
(*vals)[*num].num = txn;
(*vals)[*num].name.av_len = av->av_len;
(*vals)[(*num)++].name.av_val = tmp;
}
/**
* @brief 将方法队列vals中的所有方法清空
*/
static void AV_clear(RTMP_METHOD *vals, int num)
{
int i;
for (i = 0; i < num; i++)
free(vals[i].name.av_val);
free(vals);
}