rtmp推流h264+aac
一. 概述
本文主要讲述如何使用开源的rtmp库来搭建一个rtmp client，推送h264和aac流到rtmp server。笔者基于两套开源的项目进行了测试：rtmpdump以及srslibrtmp（这个是国人在rtmpdump基础上优化改进扩展的rtmp项目）。
srslibrtmp支持多平台上linux/mac/windows以及arm/mips交叉平台但是对交叉编译的支持不是很好（其对交叉编译工具链的引用不灵活，只支持系统默认的sysroot路径，假如你的交叉编译toolchain在某一sdk中，那么你就得需要修改makefile了）。笔者基于某mips平台ipcamera的sdk进行开发测试，实测srs server以及rtmp client运行正常，且资源占用较理想，唯一的缺点是srslibrtmp所提供的接口对于aac的支持不是很好（8000 sample_rate不支持，个人认为是bug，或者是我使用的不当，也懒得去细看代码了，后面直接通过rtmpdump的接口开发的rtmp client运行正常）。
rtmpdump是官方的rtmp lib，但是这个项目缺乏sample/接口说明（据说rtmp server的架构模式也不是很好，笔者只做rtmp client的学习研究，rtmp server部分不做探讨），用它来开发rtmp client最大的优点是数据的封装可以由自己清楚把握（其实rtmp跟flv都是Adobe公司的，rtmp的数据格式也遵从了flv格式，这个稍后讲解），这与api怎么调用可以我是参考的这两篇博文：rtmp 推送h264 + aac 的数据 使用librtmp库发布直播流 ，不把握的地方可以翻看一下源码。rtmpdump的编译倒是还比较灵活这个可以参考项目的README。
rtmp server可以通过nginx搭（参考 手把手教你搭建Nginx-rtmp流媒体服务器+使用ffmpeg推流）或者使用srslibrtmp里的rtmp server（可以在嵌入式平台上运行）。
二. 实现
这里不对rtmp的协议进行介绍（实际上只是调用api实现rtmp client的话不需要了解rtmp协议，只需要知道他是基于tcp协议端口号是1935即可，需要研究的读者可以查阅其他资源）。
rtmp在推送h264和aac流时，流的内容可以分为四类，首先在连接到rtmp server后需要传送的是h264的sequence header以及aac的sequence header（内容格式参考flv格式《Video File Format Specification Version 10》），通过这两包数据decoder才会清楚应该如何解码h264和aac数据，接下来的部分就是h264和aac 负载data部分了。
我们知道flv主要有flv file hrader/previous tag length/flv tag/previous tag length/组成，其中tag类型可分为video/audio/script三类，在rtmp传输时只需要传输flv tag即可，并且这个flv tag是去掉flv tag header的tag（以下我们统一称作rtmp tag），rtmp tag作为rtmp packet的body部分被封进了rtmp packet，rtmp传输的就是rtmp packet（当然就像前面讲的先传送h264 sequence header tag再传送aac sequence header tag，然后是h264和aac data tag），至于如何封装可以参考《Video File Format Specification Version 10》或者参考我的另外一篇blog flv封装H264+AAC[附完整代码] （当然我的flv muxer存在一些缺陷，只支持8k sample rate/bitformat 16bits,pcm buffer为1024bytes等，由于我比较懒所以就不打算改了，在源码里应该都用"TODO"标记出来了）。
这里有几点注意的得稍微提一下：
（1）RTMPPacket中的m_hasAbsTimestamp这个绝对/相对时间戳我不请出什么意思，我试了两个都可以用（我时间戳的基选的是0）；
（2）RTMP_Sendpacket里的发送到queue还是直接发送，没有研究过这两者的差异；
（3）在不同线程中调用RTMP_Sendpacket最好加锁处理，至于原因没有研究。
三. 例程
通过一个例程简单演示一下如何通过rtmpdump提供的lib书写rtmpclient代码，向rtmpserver推送h264和aac流；
详细资源请访问我的github： https://github.com/steveliu121/pistreamer/tree/master/rtmp
#include
#include
#include
#include
#include
#include
#include
#include
#include "myrtmp.h"
#include "aacenc.h"
#include "my_middle_media.h"
#include
/* XXX:WARNING the pcm period buf length should be the common factor of
* the aac input pcm frame length, or the aac timestamp will be wrong
* here the pcm period buf length == 1024 bytes,
* and the aac input pcm frame length == 2048
*/
#define SPS_LEN 28
#define PPS_LEN 6
#define RES_720P
#ifdef RES_720P
#define RESOLUTION_720P MY_VIDEO_RES_720P
#define RES_WIDTH 1280
#define RES_HEIGHT 720
#endif
#define VIDEO_FPS 15
#define VIDEO_TIME_SCALE 90000
#define VIDEO_SAMPLE_DURATION (VIDEO_TIME_SCALE / VIDEO_FPS)
#define AUDIO_SAMPLERATE 8000
#define AUDIO_CHANNELS 1
#define AUDIO_TIME_SCALE (AUDIO_SAMPLERATE * AUDIO_CHANNELS)
/* (AUDIO_TIME_SCALE / AUDIO_FPS)audio_period_time = 64ms, fps = 15.625 */
#define AUDIO_SAMPLE_DURATION 512
#define AAC_BITRATE 16000
static const uint8_t sps_buf[SPS_LEN] = {0x27, 0x64, 0x00, 0x29, 0xac, 0x1a, 0xd0, 0x0a,
0x00, 0xb7, 0x4d, 0xc0, 0x40, 0x40, 0x50, 0x00,
0x00, 0x03, 0x00, 0x10, 0x00 ,0x00, 0x03, 0x01,
0xe8, 0xf1 ,0x42, 0x2a};
static const uint8_t pps_buf[PPS_LEN] = {0x28, 0xee, 0x01, 0x34, 0x92, 0x24};
/*
static const uint8_t sps_buf[SPS_LEN + 4] = {0x00, 0x00, 0x00, 0x1c, 0x27, 0x64,
0x00, 0x29, 0xac, 0x1a, 0xd0, 0x0a,
0x00, 0xb7, 0x4d, 0xc0, 0x40, 0x40, 0x50, 0x00,
0x00, 0x03, 0x00, 0x10, 0x00 ,0x00, 0x03, 0x01,
0xe8, 0xf1 ,0x42, 0x2a};
static const uint8_t pps_buf[PPS_LEN + 4] = {0x00, 0x00, 0x00, 0x06, 0x28, 0xee,
0x01, 0x04, 0x92, 0x24};
*/
static int g_exit;
static HANDLE_AACENCODER aac_enc_hd;
static uint8_t aac_decoder_conf[64];
static int aac_decoder_conf_len;
uint32_t g_timestamp_begin;
RTMP *rtmp;
RTMPPacket video_pkt;
RTMPPacket audio_pkt;
pthread_mutex_t av_mutex;
void sig_handle(int sig)
{
g_exit = 1;
}
void h264_cb(const struct timeval *tv, const void *data,
const int len, const int keyframe)
{
int ret = 0;
uint8_t *buf = NULL;
int buf_len = 0;
uint32_t timestamp = 0;
int buf_payload_len = 0;
timestamp = (tv->tv_sec * 1000) + (tv->tv_usec / 1000);
if (g_timestamp_begin == 0)
g_timestamp_begin = timestamp;
/* strip sps/pps from I frame and
* replace NALU start flag '0x00/0x00/0x00/0x01' with
* the length of NALU in BIGENDIAN
*/
if (keyframe) {
buf = (uint8_t *)data + SPS_LEN + PPS_LEN + 2 * 4;
buf_len = len - SPS_LEN - PPS_LEN - 2 * 4;
} else {
buf = (uint8_t *)data;
buf_len = len;
}
buf_payload_len = buf_len - 4;
buf[0] = buf_payload_len >> 24;
buf[1] = buf_payload_len >> 16;
buf[2] = buf_payload_len >> 8;
buf[3] = buf_payload_len & 0xff;
video_pkt.m_headerType = RTMP_PACKET_SIZE_LARGE;
video_pkt.m_nTimeStamp = (timestamp - g_timestamp_begin);
video_pkt.m_nBodySize = buf_len + 5;//5bytes VIDEODATA tag header
rtmppacket_alloc(&video_pkt, video_pkt.m_nBodySize);
rtmp_write_avc_data_tag(video_pkt.m_body, buf, buf_len, keyframe);
ret = rtmp_isconnected(rtmp);
if (ret == true) {
/* true: send to outqueue;false: send directly */
pthread_mutex_lock(&av_mutex);
ret = rtmp_sendpacket(rtmp, &video_pkt, true);
if (ret == false)
printf("rtmp send video packet fail\n");
pthread_mutex_unlock(&av_mutex);
}
rtmppacket_free(&video_pkt);
}
void audio_cb(const struct timeval *tv, const void *pcm_buf,
const int pcm_len, const void *spk_buf)
{
int ret = 0;
uint8_t *aac_buf = NULL;
int aac_buf_len = 0;
uint32_t timestamp = 0;
timestamp = (tv->tv_sec * 1000) + (tv->tv_usec / 1000);
if (g_timestamp_begin == 0)
g_timestamp_begin = timestamp;
aac_buf_len = aac_encode(aac_enc_hd, pcm_buf, pcm_len, &aac_buf);
if (aac_buf_len == 0)
return;
audio_pkt.m_headerType = RTMP_PACKET_SIZE_LARGE;
audio_pkt.m_nTimeStamp = (timestamp - g_timestamp_begin);
audio_pkt.m_nBodySize = aac_buf_len - 7 + 2;//7bytes ADTS header & 2bytes AUDIODATA tag header
rtmppacket_alloc(&audio_pkt, audio_pkt.m_nBodySize);
rtmp_write_aac_data_tag(audio_pkt.m_body, aac_buf, aac_buf_len);
ret = rtmp_isconnected(rtmp);
if (ret == true) {
/* true: send to outqueue;false: send directly */
pthread_mutex_lock(&av_mutex);
ret = rtmp_sendpacket(rtmp, &audio_pkt, true);
if (ret == false)
printf("rtmp send audio packet fail\n");
pthread_mutex_unlock(&av_mutex);
}
rtmppacket_free(&audio_pkt);
}
static int __connect2rtmpsvr(char *url)
{
int ret = 0;
rtmp = rtmp_alloc();
rtmp_init(rtmp);
rtmp->Link.timeout=5; //default 30s
ret = rtmp_setupurl(rtmp, url);
if (ret == false) {
printf("rtmp setup url fail\n");
goto exit;
}
rtmp_enablewrite(rtmp);
ret = rtmp_connect(rtmp, NULL);
if (ret == false) {
printf("rtmp connect fail\n");
goto exit;
}
ret = rtmp_connectstream(rtmp, 0);
if (ret == false) {
printf("rtmp connect stream fail\n");
rtmp_close(rtmp);
goto exit;
}
return 0;
exit:
return -1;
}
static void __rtmp_send_sequence_header(void)
{
int ret = 0;
/* rtmp send audio/video sequence header frame */
rtmppacket_reset(&video_pkt);
rtmppacket_reset(&audio_pkt);
video_pkt.m_packetType = RTMP_PACKET_TYPE_VIDEO;
video_pkt.m_nChannel = 0x04;
video_pkt.m_nInfoField2 = rtmp->m_stream_id;
video_pkt.m_hasAbsTimestamp = false;
audio_pkt.m_packetType = RTMP_PACKET_TYPE_AUDIO;
audio_pkt.m_nChannel = 0x04;
audio_pkt.m_nInfoField2 = rtmp->m_stream_id;
audio_pkt.m_hasAbsTimestamp = false;
video_pkt.m_headerType = RTMP_PACKET_SIZE_LARGE;
video_pkt.m_nTimeStamp = 0;
video_pkt.m_nBodySize = SPS_LEN + PPS_LEN + 16;
rtmppacket_alloc(&video_pkt, video_pkt.m_nBodySize);
rtmp_write_avc_sequence_header_tag(video_pkt.m_body,
sps_buf, SPS_LEN,
pps_buf, PPS_LEN);
ret = rtmp_isconnected(rtmp);
if (ret == true) {
/* true: send to outqueue;false: send directly */
pthread_mutex_lock(&av_mutex);
ret = rtmp_sendpacket(rtmp, &video_pkt, true);
if (ret == false)
printf("rtmp send video packet fail\n");
pthread_mutex_unlock(&av_mutex);
}
rtmppacket_free(&video_pkt);
audio_pkt.m_headerType = RTMP_PACKET_SIZE_LARGE;
audio_pkt.m_nTimeStamp = 0;
audio_pkt.m_nBodySize = 4;
rtmppacket_alloc(&audio_pkt, audio_pkt.m_nBodySize);
rtmp_write_aac_sequence_header_tag(audio_pkt.m_body,
AUDIO_SAMPLERATE, AUDIO_CHANNELS);
ret = rtmp_isconnected(rtmp);
if (ret == true) {
/* true: send to outqueue;false: send directly */
pthread_mutex_lock(&av_mutex);
ret = rtmp_sendpacket(rtmp, &audio_pkt, true);
if (ret == false)
printf("rtmp send audio packet fail\n");
pthread_mutex_unlock(&av_mutex);
}
rtmppacket_free(&audio_pkt);/* rtmp send audio/video sequence header frame */
}
int main(int argc, char *argv[])
{
int ret = 0;
MYVideoInputChannel chn = {
.channelId = 0,
.res = RESOLUTION_720P,
.fps = VIDEO_FPS,
.bitrate = 1024,
.gop = 1,
.vbr = MY_BITRATE_MODE_CBR,
.cb = h264_cb
};
MYVideoInputOSD osd_info = {
.pic_enable = 0,
.pic_path = "/usr/osd_char_lib/argb_2222",
.pic_x = 200,
.pic_y = 200,
.time_enable = 1,
.time_x = 100,
.time_y = 100
};
MYAudioInputAttr_aec audio_in = {
.sampleRate = AUDIO_SAMPLERATE,
.sampleBit = 16,
.volume = 95,
.cb = audio_cb
};
signal(SIGTERM, sig_handle);
signal(SIGINT, sig_handle);
pthread_mutex_init(&av_mutex, NULL);
rtmp_logsetlevel(RTMP_LOGINFO);
if (argc <= 1) {
printf("Usage: %s \n"
" rtmp_url RTMP stream url to publish\n"
"For example:\n"
" %s rtmp://127.0.0.1:1935/live/livestream\n",
argv[0], argv[0]);
exit(-1);
}
ret = __connect2rtmpsvr(argv[1]);
if (ret < 0)
goto exit;
/* create aacencoder */
ret = create_aac_encoder(&aac_enc_hd,
AUDIO_CHANNELS, AUDIO_SAMPLERATE, AAC_BITRATE,
aac_decoder_conf, &aac_decoder_conf_len);
if (ret < 0)
goto exit;/* create aacencoder */
__rtmp_send_sequence_header();
/* start audio&video device and receive buffers, do muxer in callback */
MYAV_Context_Init();
ret = MYVideoInput_Init();
if (ret)
goto out;
ret = MYVideoInput_AddChannel(chn);
if (ret)
goto out;
ret = MYVideoInput_SetOSD(chn.channelId, &osd_info);
if (ret)
goto out;
ret = MYAudioInputOpen(&audio_in);
if (ret)
goto out;
ret = MYVideoInput_Start();
if (ret)
goto out;
ret = MYAudioInputStart();
if (ret)
goto out;/* start audio&video device and receive buffers, do muxer in callback */
while (!g_exit)
sleep(1);
out:
MYVideoInput_Uninit();
MYAudioInputStop();
MYAudioInputClose();
MYAV_Context_Release();
exit:
pthread_mutex_destroy(&av_mutex);
rtmp_close(rtmp);
rtmp_free(rtmp);
rtmp = NULL;
return ret;
}