T31开发笔记：librtmp拉流测试

若该文为原创文章，转载请注明原文出处。

T31使用librtmp拉流并保存成FLV文件或H264和AAC文件。

librtmp编译在前面有教程，自行编译。

实现的目的是想要获取获取rtmp的AAC流并播放，实时双向对讲功能。

一、硬件和开发环境
1、硬件：T31X+SC5235 

2、开发环境： ubuntu16.04-64bit

3、编译器：mips-gcc540-glibc222-32bit-r3.3.0.tar.gz

注：板子和和WIFI模块是某淘上淘的，使用的是RTL8188，使用的是USB接口，uboot和内核是自己裁剪移植的，内核默认自带WIFI驱动，所以不用移植可以直接使用。
 

二、使用librtmp拉流流程

1. 初使化RTMP上下文
2. 设置拉流地址
3. 连接服务器
4. 连接流地址
5. 循环拉流，提取媒体数据，保存为文件或者交给解码模块
6. 拉流完毕，释放资源

三、代码解析

编译代码需要用到的库，本人在T31上使用的是静态库，可以自行改成动态库，以减少编译文件的大小 。

编译所需的库有：librtmp.a、libssl.a、libcrypto.a、libz.a

下面这个例子演示了使用librtmp库从服务器拉流到本地保存为flv文件或是h264和aac文件，代码如下：

/*!*******************************************************************************  Copyright ? 2017-2018 yifeng. All Rights Reserved.** \file      main.c* \author    yifeng* \version   1.0* \date      2022年3月3日* \brief     rtmp测试代码**----------------------------------------------------------------------------* \attention********************************************************************************//*****************************************************************************change history: 1.date  : 2022年3月3日author: yifengchange: create file*****************************************************************************/#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <fcntl.h>
#include <limits.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <arpa/inet.h>
#include <sys/select.h>
#include <sys/time.h>
#include <time.h>
#include <unistd.h>
#include <signal.h>
#include <sys/ioctl.h>
#include <net/if.h>
#include <netinet/in.h>
#include <pthread.h>
#include <stdbool.h>/* 环形队列头文件 */
#include "xiecc_rtmp.h"
#include "rtmp.h"typedef unsigned long   ULONG;
typedef unsigned int    UINT;
typedef unsigned char   BYTE;
typedef char            _TCHAR;uint8_t nalu_header[4] = { 0x00, 0x00, 0x00, 0x01 };/*!* \fn     main* \brief  主函数*          * \param  [in] int argc       #* \param  [in] char *argv[]   #* * \retval int*/
int main(int argc, char *argv[])
{ uint16_t object_type = 0;uint16_t sample_frequency_index = 0;uint8_t  channels = 0;uint8_t  frame_length_flag = 0;uint8_t  depend_on_core_coder = 0;uint8_t  extension_flag = 0;// 初使化RTMP上下文RTMP* pRTMP = RTMP_Alloc();RTMP_Init(pRTMP);// 设置拉流地址RTMP_SetupURL(pRTMP, (char*)"rtmp://192.168.0.109/live/stream");// 连接服务器pRTMP->Link.timeout = 10;pRTMP->Link.lFlags |= RTMP_LF_LIVE;bool b = RTMP_Connect(pRTMP, NULL);if (!b){printf("connect failed! \n");return -1;}// 连接流地址b = RTMP_ConnectStream(pRTMP, 0);if (!b){printf("connect stream failed! \n");return -1;}bool bSaveFlv = false;  // 保存成FLV格式FILE *pFile = fopen("testrtmp.flv", "wb");FILE *h264_file_ptr = fopen("testrtmp.h264", "wb");FILE *aac_file_ptr = fopen("testrtmp.aac", "wb");while (RTMP_IsConnected(pRTMP)){if (bSaveFlv == true){char sBuf[4096] = {0};int bytes = RTMP_Read(pRTMP, sBuf, sizeof(sBuf));printf("RTMP_Read() ret:[%d] \n", bytes);if (bytes <= 0)break;fwrite(sBuf, 1, bytes, pFile);}else{RTMPPacket packet;RTMPPacket_Reset(&packet);packet.m_body = NULL;packet.m_chunk = NULL;b = RTMP_ReadPacket(pRTMP, &packet);if (b < 0)break;if (0 == b )continue;if (RTMPPacket_IsReady(&packet)){RTMP_ClientPacket(pRTMP, &packet);if (packet.m_packetType == RTMP_PACKET_TYPE_VIDEO){bool keyframe = 0x17 == packet.m_body[0] ? true : false;bool sequence = 0x00 == packet.m_body[1];printf("keyframe=%s, sequence=%s\n", keyframe ? "true" : "false", sequence ? "true" : "false");// SPS/PPS sequenceif (sequence){uint32_t offset = 10;uint32_t sps_num = packet.m_body[offset++] & 0x1f;for (int i = 0; i < sps_num; i++) {uint8_t ch0 = packet.m_body[offset];uint8_t ch1 = packet.m_body[offset + 1];uint32_t sps_len = ((ch0 << 8) | ch1);offset += 2;// Write sps dataprintf("Write sps data len: %d\n", sps_len);fwrite(nalu_header, sizeof(uint8_t), 4, h264_file_ptr);fwrite(packet.m_body + offset, sizeof(uint8_t), sps_len, h264_file_ptr);offset += sps_len;}uint32_t pps_num = packet.m_body[offset++] & 0x1f;for (int i = 0; i < pps_num; i++) {uint8_t ch0 = packet.m_body[offset];uint8_t ch1 = packet.m_body[offset + 1];uint32_t pps_len = ((ch0 << 8) | ch1);offset += 2;// Write pps dataprintf("Write pps data len: %d\n", pps_len);fwrite(nalu_header, sizeof(uint8_t), 4, h264_file_ptr);fwrite(packet.m_body + offset, sizeof(uint8_t), pps_len, h264_file_ptr);offset += pps_len;}}// Nalu frameselse{uint32_t offset = 5;uint8_t ch0 = packet.m_body[offset];uint8_t ch1 = packet.m_body[offset + 1];uint8_t ch2 = packet.m_body[offset + 2];uint8_t ch3 = packet.m_body[offset + 3];uint32_t data_len = ((ch0 << 24) | (ch1 << 16) | (ch2 << 8) | ch3);offset += 4;// Write nalu data(already started with '0x00,0x00,0x00,0x01')printf("Write nalu data len: %d\n", data_len);fwrite(nalu_header, sizeof(uint8_t), 4, h264_file_ptr);fwrite(packet.m_body + offset, sizeof(uint8_t), data_len, h264_file_ptr);offset += data_len;}}else if (packet.m_packetType == RTMP_PACKET_TYPE_AUDIO){bool sequence = 0x00 == packet.m_body[1];printf("sequence=%s\n", sequence ? "true" : "false");// AAC sequenceif (sequence) {uint8_t format = (packet.m_body[0] & 0xf0) >> 4;uint8_t samplerate = (packet.m_body[0] & 0x0c) >> 2;uint8_t sampledepth = (packet.m_body[0] & 0x02) >> 1;uint8_t type = packet.m_body[0] & 0x01;// sequence = packet.m_body[1];// AAC(AudioSpecificConfig)if (format == 10){uint8_t ch0 = packet.m_body[2];uint8_t ch1 = packet.m_body[3];uint16_t config = ((ch0 << 8) | ch1);object_type = (config & 0xF800) >> 11;sample_frequency_index = (config & 0x0780) >> 7;channels = (config & 0x78) >> 3;frame_length_flag = (config & 0x04) >> 2;depend_on_core_coder = (config & 0x02) >> 1;extension_flag = config & 0x01;}// Speex(Fix data here, so no need to parse...)else if (format == 11) {// 16 KHz, mono, 16bit/sampletype = 0;sampledepth = 1;samplerate = 4;}}// Audio frameselse {// ADTS(7 bytes) + AAC datauint32_t data_len = packet.m_nBodySize - 2 + 7;uint8_t adts[7];adts[0] = 0xff;adts[1] = 0xf1;adts[2] = ((object_type - 1) << 6) | (sample_frequency_index << 2) | (channels >> 2);adts[3] = ((channels & 3) << 6) + (data_len >> 11);adts[4] = (data_len & 0x7FF) >> 3;adts[5] = ((data_len & 7) << 5) + 0x1F;adts[6] = 0xfc;// Write audio framesprintf("Write audio frames len: %d\n", packet.m_nBodySize - 2);fwrite(adts, sizeof(uint8_t), 7, aac_file_ptr);fwrite(packet.m_body + 2, sizeof(uint8_t), packet.m_nBodySize - 2, aac_file_ptr);}}else if (packet.m_packetType == RTMP_PACKET_TYPE_INFO) {// TODO:// ...printf("RTMP_PACKET_TYPE_INFO1\n");}else {// TODO:// ...printf("RTMP_PACKET_TYPE_INFO2\n");}}RTMPPacket_Free(&packet);}}fclose(pFile);RTMP_Close(pRTMP);RTMP_Free(pRTMP);return 0;
}

定义了bSaveFlv标记，是否保存成flv文件，true保存成flv文件，否则保存成aac和h264，

rtmp的Url需要根据自己的服务器修改。

四、测试结果

执行后，在当前目录下生成264和aac文件

把文件复制到pc端用vlc播放

 使用librtmp拉流网友説会有问题，目前测试是正常，有遇到的网友麻烦告知一下。谢谢。

如有侵权，或需要完整代码，请及时联系博主。