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流媒体之RTMP——librtmp推流测试

程序员文章站 2022-07-08 19:46:40
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文章目录


作者：一步(Reser)
日期：2019.10.9

一：LibRTMP推流测试

测试使用 FFMPEG 从MP4文件中解析出H264裸流，之后按照固定帧率将视频流推送到RTMP服务器。

  • H264流支持从MP4文件中解析，或者从H264文件中读取(未测试)；
  • 暂未加入音频测试。

新建项目，加入FFMPEG和LibRTMP相关依赖。



二：时间控制

编写简单的时间控制类 CTimeStatistics，这个类主要负责RTMP发送时候的帧控制。

#pragma once
#define WIN32_LEAN_AND_MEAN
#include <Windows.h>


typedef long long tick_t;

#define GET_TIME(T,S,F) ((double)((T)-(S))/(double)(F/1000))


class CTimeStatistics
{
public:
	CTimeStatistics()
	{
		_start = 0;
		_stop = 0;
	}
	virtual ~CTimeStatistics() {};

	inline void reset()
	{
		_start = 0;
		_stop = 0;
	}
	inline void start()
	{
		_start = _get_tick();
	}
	inline void stop()
	{
		_stop = _get_tick();
	}
	inline double get_delta()
	{
		return GET_TIME(_get_tick(), _start, _get_frequency());
	}
	inline double get_total()
	{
		return GET_TIME(_stop, _start, _get_frequency());
	}

protected:
	tick_t _get_tick()
	{
		LARGE_INTEGER t1;
		QueryPerformanceCounter(&t1);
		return t1.QuadPart;
	}
	tick_t _get_frequency()
	{
		LARGE_INTEGER t1;
		QueryPerformanceFrequency(&t1);
		return t1.QuadPart;
	}

private:
	tick_t _start;
	tick_t _stop;
};

包含简单的Start/Stop/Reset和获得时间间隔接口。



三：FFMPEG从MP4文件解析出H264

这里假定对FFMPEG的使用有些基本了解。
主要结构体定义：

// For ffmpeg demux
typedef enum _stream_type
{
	STREAM_FILE_MP4 = 0,
	STREAM_H264_RAW
} stream_type_t;

typedef struct _fmt_manage
{
	AVFormatContext *context;
	AVStream *vstream;
	AVStream *astream;
} fmt_manage_t;

从文件中解析出流信息：

// Ffmpeg demux, parse streams from file
av_register_all();
if (avformat_open_input(&_manage.context, file.c_str(), 0, 0) < 0)
	break;
if (avformat_find_stream_info(_manage.context, 0) < 0)
	break;
if (!_parse_streams(meta, type))
	break;

其中主要函数 _parse_stream：

bool CTestLibRTMPPusher::_parse_streams(metadata_t &meta, stream_type_t type)
{
	for (int i = 0; i < _manage.context->nb_streams; i++) {
		// Video stream
		if (AVMEDIA_TYPE_VIDEO == _manage.context->streams[i]->codecpar->codec_type) {
			_manage.vstream = _manage.context->streams[i];

			// Parse sps/pps from extradata
			// If MP4,extradata stores 'avcCfg'; or stores 'sps/pps'
			if (_manage.context->streams[i]->codecpar->extradata_size > 0) {
				uint32_t size = _manage.context->streams[i]->codecpar->extradata_size;
				uint8_t *ptr = _manage.context->streams[i]->codecpar->extradata;

				switch (type)
				{
				case STREAM_FILE_MP4:
				{
					// Parse SPS/PPS from avcCfg
					uint32_t offset = 5;
					uint32_t num_sps = ptr[offset++] & 0x1f;
					for (uint32_t j = 0; j < num_sps; j++) {
						meta.param.size_sps = (ptr[offset++] << 8);
						meta.param.size_sps |= ptr[offset++];
						memcpy(meta.param.data_sps, ptr + offset, meta.param.size_sps);
						offset += meta.param.size_sps;
					}
					uint32_t num_pps = ptr[offset++];
					for (uint32_t j = 0; j < num_pps; j++) {
						meta.param.size_pps = (ptr[offset++] << 8);
						meta.param.size_pps |= ptr[offset++];
						memcpy(meta.param.data_pps, ptr + offset, meta.param.size_pps);
						offset += meta.param.size_pps;
					}
				}
				break;
				case STREAM_H264_RAW:
				{
					// Parse SPS/PPS from 'sps/pps'
					uint32_t offset = 0;
					if (ptr[offset] != 0x00 || ptr[offset + 1] != 0x00 || ptr[offset + 2] != 0x00 || ptr[offset + 3] != 0x01) {
						// No valid data...
					}
					else {
						// Find next pos
						offset++;
						while ((ptr[offset] != 0x00 || ptr[offset + 1] != 0x00 || ptr[offset + 2] != 0x00 || ptr[offset + 3] != 0x01) && (offset < size - 3))
							offset++;

						if ((ptr[4] & 0x1f) == 7) { // SPS first
							meta.param.size_sps = offset - 4;
							memcpy(meta.param.data_sps, ptr + 4, meta.param.size_sps);
							meta.param.size_pps = size - offset - 4;
							memcpy(meta.param.data_pps, ptr + offset + 4, meta.param.size_pps);
						}
						else if ((ptr[4] & 0x1f) == 8) { // PPS first
							meta.param.size_pps = offset - 4;
							memcpy(meta.param.data_pps, ptr + 4, meta.param.size_pps);
							meta.param.size_sps = size - offset - 4;
							memcpy(meta.param.data_sps, ptr + offset + 4, meta.param.size_sps);
						}
					}
				}
				break;
				default:
					break;
				}
			}
		}
		// Audio stream
		else if (AVMEDIA_TYPE_AUDIO == _manage.context->streams[i]->codecpar->codec_type) {
			_manage.astream = _manage.context->streams[i];

			// parse esds from extra data
			if (_manage.context->streams[i]->codecpar->extradata_size > 0) {
				uint32_t size = _manage.context->streams[i]->codecpar->extradata_size;
				uint8_t *ptr = _manage.context->streams[i]->codecpar->extradata;

				// Ignore audio
				// ...
			}
		}
	}

	return true;
}
  • 根据 _manage.context->streams[i]->codecpar->codec_type 判断流类型；
  • 视频流的extradata保存视频配置数据，当为MP4文件时，保存的是 avcCfg 结构体，需要从中解析出SPS和PPS；当为H264裸流文件时，保存的是以 0x00,0x00,0x00,0x01 开头的SPS和PPS，需要从中解析出SPS和PPS；

循环解析和发送数据：

// Read frames from file by ffmpeg
AVPacket pkt = { 0 };
if (av_read_frame(_manage.context, &pkt) < 0)
	break;

// Video frame
if (pkt.stream_index == _manage.vstream->index) {
	uint64_t pts = period_ms * video_frame_count;
	bool keyframe = pkt.flags & AV_PKT_FLAG_KEY;

	// Replace size-4-bytes with 0x00,0x00,0x00,0x01
	pkt.data[0] = 0x00;
	pkt.data[1] = 0x00;
	pkt.data[2] = 0x00;
	pkt.data[3] = 0x01;
	_send_video(pkt.size, pkt.data, pts, keyframe);

	video_frame_count++;
	if (video_frame_count % 100 == 0) {
		printf("Send video frames: %d\n", video_frame_count);
	}
}
// Audio frame
else if (pkt.stream_index == _manage.astream->index) {
	// Ignore audio
	// ...

	audio_frame_count++;
}
  • 读取的帧类型可由 pkt.stream_index 判断；
  • 视频关键帧由 pkt.flags & AV_PKT_FLAG_KEY 判断；
  • 解析出的视频帧并没有以 0x00,0x00,x00,0x01 分割，而是以4个字节的size打头，因此需要将4个字节的size替换为 0x00,0x00,x00,0x01 起始分隔符;
  • 解析出的pts可能为0，因此使用自定义的pts。对于某些RTMP服务器，需要pts从0开始。


四：LibRTMP的使用

使用的结构体定义：

// For rtmp structure
#define H264_PARAM_LEN		512
typedef struct _h264_param
{
	uint32_t size_sps;
	uint8_t	data_sps[H264_PARAM_LEN];
	uint32_t size_pps;
	uint8_t	data_pps[H264_PARAM_LEN];
} h264_param_t;

typedef struct _metadata
{
	// Video
	uint32_t width;
	uint32_t height;
	uint32_t fps;
	uint32_t bitrate_kpbs;
	h264_param_t param;

	// Audio
	bool has_audio;
	uint32_t channels;
	uint32_t samplerate;
	uint32_t samplesperframe;
	uint32_t datarate;
} metadata_t;

LibRTMP在Windows下的使用前和使用后要进行Socket的初始化和反初始化：

bool CTestLibRTMPPusher::_init_sockets()
{
	WORD version;
	WSADATA wsaData;
	version = MAKEWORD(2, 2);
	return (0 == WSAStartup(version, &wsaData));
}

void CTestLibRTMPPusher::_cleanup_sockets()
{
	WSACleanup();
}

RTMP资源初始化：

...
_rtmp_ptr = RTMP_Alloc();
if (NULL == _rtmp_ptr)
	break;
RTMP_Init(_rtmp_ptr);
...

// Parse rtmp url
_rtmp_ptr->Link.timeout = timeout_secs;
_rtmp_ptr->Link.lFlags |= RTMP_LF_LIVE;
if (RTMP_SetupURL(_rtmp_ptr, (char *)url.c_str()) < 0)
	break;

// Pusher mode
RTMP_EnableWrite(_rtmp_ptr);

// Socket connection
// Handshakes and connect command
if (RTMP_Connect(_rtmp_ptr, NULL) < 0)
	break;

// Setup stream and stream settings
if (RTMP_ConnectStream(_rtmp_ptr, 0) < 0)
	break;

// Send metadata(video and audio settings)
if (!_send_metadata(_metadata))
	break;

RTMP为Adobe公司开发，其上层传输封装主要基于 FLV 格式，这样可以使用Flash插件直接播放。因此，在发送RTMP的包时必须要属性FLV的封装规范。

FLV由Header和Body组成；Body由多组Tag组成；Tag又由TagHeader和TagData组成。由于RTMP协议的包已经包含了TagHeader的信息，因此，在推流时没必要附加上TagHeader，即实际发送的RTMP packet：

A/V data
    |
添加上FLV的TagData头部分数据
    |
添加包信息，用于底层分包使用
    |
底层拆包发送

4.1 发送Metadata

onMetaData 为FLV的第一个Tag。在RTMP的网络和流通道建立完毕后，需要上层发送的第一个包就是Metadata包。Metadata包主要是键值对形式，指明音视频的格式和解码信息。详细可参见文末 FLV文件的第一个Tag: onMetaData 。
流媒体之RTMP——librtmp推流测试
代码：

RTMPPacket packet;
RTMPPacket_Alloc(&packet, RTMP_METADATA_SIZE);
RTMPPacket_Reset(&packet);
packet.m_packetType = RTMP_PACKET_TYPE_INFO;
packet.m_nChannel = 0x04;
packet.m_headerType = RTMP_PACKET_SIZE_LARGE;
packet.m_nTimeStamp = 0;
packet.m_nInfoField2 = _rtmp_ptr->m_stream_id;

/////////////////////////////////////////////
// Send media info
char *ptr = packet.m_body;
ptr = _put_byte(ptr, AMF_STRING);
ptr = _put_amf_string(ptr, "@setDataFrame");
ptr = _put_byte(ptr, AMF_STRING);
ptr = _put_amf_string(ptr, "onMetaData");
ptr = _put_byte(ptr, AMF_OBJECT);
ptr = _put_amf_string(ptr, "copyright");
ptr = _put_byte(ptr, AMF_STRING);
ptr = _put_amf_string(ptr, "firehood");
ptr = _put_amf_string(ptr, "width");
ptr = _put_amf_double(ptr, meta.width);
ptr = _put_amf_string(ptr, "height");
ptr = _put_amf_double(ptr, meta.height);
ptr = _put_amf_string(ptr, "framerate");
ptr = _put_amf_double(ptr, meta.fps);
ptr = _put_amf_string(ptr, "videodatarate");
ptr = _put_amf_double(ptr, meta.bitrate_kpbs);
double vcodec_ID = 7;
ptr = _put_amf_string(ptr, "videocodecid");
ptr = _put_amf_double(ptr, vcodec_ID);
if (meta.has_audio) {
	ptr = _put_amf_string(ptr, "audiodatarate");
	ptr = _put_amf_double(ptr, meta.datarate);
	ptr = _put_amf_string(ptr, "audiosamplerate");
	ptr = _put_amf_double(ptr, meta.samplerate);
	ptr = _put_amf_string(ptr, "audiosamplesize");
	ptr = _put_amf_double(ptr, meta.samplesperframe);
	ptr = _put_amf_string(ptr, "stereo");
	ptr = _put_amf_double(ptr, meta.channels);
	double acodec_ID = 10;
	ptr = _put_amf_string(ptr, "audiocodecid");
	ptr = _put_amf_double(ptr, acodec_ID);
}
ptr = _put_amf_string(ptr, "");
ptr = _put_byte(ptr, AMF_OBJECT_END);
packet.m_nBodySize = ptr - packet.m_body;
if (RTMP_SendPacket(_rtmp_ptr, &packet, 0) < 0) {
	RTMPPacket_Free(&packet);
	return false;
}
  • RTMP发送的参数设置使用 AMF/AMF3 编码方式；主要以键值对方式；
  • 由官方文档：H264编码ID为7，AAC编码ID为10；
  • 音频可选。

4.2 发送视频

视频帧数据需要打上FLV的TagData头数据。详细可参见文末 librtmp获取视频流和音频流1 。视频包有两种，一种为视频同步数据 AVCDecoderConfigurationRecord （解码信息包），一种为H264帧数据 One or more NALUs （内容视频包）。

Video TagData：

Field Type Comment
Frame Type UB [4] Type of video frame. The following values are defined:
1 = key frame (for AVC, a seekable frame)
2 = inter frame (for AVC, a non-seekable frame)
3 = disposable inter frame (H.263 only)
4 = generated key frame (reserved for server use only)
5 = video info/command frame
CodecID UB [4] Codec Identifier. The following values are defined:
2 = Sorenson H.263
3 = Screen video
4 = On2 VP6
5 = On2 VP6 with alpha channel
6 = Screen video version 2
7 = AVC
AVCPacketType IF CodecID == 7
UI8		 The following values are defined:
0 = AVC sequence header
1 = AVC NALU
2 = AVC end of sequence (lower level NALU sequence ender is not required or supported)
CompositionTime IF CodecID == 7
SI24		 IF AVCPacketType == 1
Composition time offset
ELSE
0
See ISO 14496-12, 8.15.3 for an explanation of compositiontimes. The offset in an FLV file is always in milliseconds.

F AVCPacketType == 0 AVCDecoderConfigurationRecord（AVC sequence header）
IF AVCPacketType == 1 One or more NALUs (Full frames are required)

4.2.1 发送视频信息包

/////////////////////////////////////////////
// Send decode info
// FLV video sequence format:
// Frame type(4 bits) + codecID(4 bits) + AVCPacketType(1 bytes) + CompositionTime + AVCDecoderConfiguration
//
uint32_t offset = 0;
packet.m_body[offset++] = 0x17;
packet.m_body[offset++] = 0x00;
packet.m_body[offset++] = 0x00;
packet.m_body[offset++] = 0x00;
packet.m_body[offset++] = 0x00;
// AVCDecoderConfiguration  
packet.m_body[offset++] = 0x01;
packet.m_body[offset++] = meta.param.data_sps[1];
packet.m_body[offset++] = meta.param.data_sps[2];
packet.m_body[offset++] = meta.param.data_sps[3];
packet.m_body[offset++] = 0xff;
// SPS
packet.m_body[offset++] = 0xE1;
packet.m_body[offset++] = meta.param.size_sps >> 8;
packet.m_body[offset++] = meta.param.size_sps & 0xff;
memcpy(&packet.m_body[offset], meta.param.data_sps, meta.param.size_sps);
offset += meta.param.size_sps;
// PPS
packet.m_body[offset++] = 0x01;
packet.m_body[offset++] = meta.param.size_pps >> 8;
packet.m_body[offset++] = meta.param.size_pps & 0xff;
memcpy(&packet.m_body[offset], meta.param.data_pps, meta.param.size_pps);
offset += meta.param.size_pps;
packet.m_packetType = RTMP_PACKET_TYPE_VIDEO;
packet.m_nBodySize = offset;
if (RTMP_SendPacket(_rtmp_ptr, &packet, 0) < 0) {
	RTMPPacket_Free(&packet);
	return false;
}
  • 包数据部分按 AVCDecoderConfiguration 格式即可。

4.2.2 发送视频数据包

RTMPPacket packet;
RTMPPacket_Alloc(&packet, size + RTMP_RESERVED_HEAD_SIZE * 2);
RTMPPacket_Reset(&packet);
packet.m_packetType = RTMP_PACKET_TYPE_VIDEO;
packet.m_nChannel = 0x04;
packet.m_headerType = RTMP_PACKET_SIZE_LARGE;
packet.m_nTimeStamp = pts;
packet.m_nInfoField2 = _rtmp_ptr->m_stream_id;

uint32_t offset = 0;
if (keyframe)
	packet.m_body[offset++] = 0x17;
else
	packet.m_body[offset++] = 0x27;
packet.m_body[offset++] = 0x01;
packet.m_body[offset++] = 0x00;
packet.m_body[offset++] = 0x00;
packet.m_body[offset++] = 0x00;
packet.m_body[offset++] = size >> 24;
packet.m_body[offset++] = size >> 16;
packet.m_body[offset++] = size >> 8;
packet.m_body[offset++] = size & 0xff;
memcpy(packet.m_body + offset, data_ptr, size);
packet.m_nBodySize = offset + size;
if (RTMP_SendPacket(_rtmp_ptr, &packet, 0) < 0) {
	RTMPPacket_Free(&packet);
	return false;
}

RTMPPacket_Free(&packet);
return true;

4.3 发送音频

同样，音频也分为音频信息包和音频数据包。

Audio TagData:

名称 二进制值 介绍
音频格式 4 bits[AAC]1010 0 = Linear PCM, platform endian
1 = ADPCM
2 = MP3
3 = Linear PCM, little endian
4 = Nellymoser 16-kHz mono
5 = Nellymoser 8-kHz mono
6 = Nellymoser
7 = G.711 A-law logarithmic PCM
8 = G.711 mu-law logarithmic PCM
9 = reserved
10 = AAC
11 = Speex
14 = MP3 8-Khz
15 = Device-specific sound
采样率 2 bits[44kHZ]11 0 = 5.5-kHz
1 = 11-kHz
2 = 22-kHz
3 = 44-kHz
对于AAC总是3
采样的长度 1 bit 0 = snd8Bit
1 = snd16Bit
压缩过的音频都是16bit
音频类型 1 bit 0 = sndMono，单声道
1 = sndStereo，立体声
对于AAC总是1
ACCPacketType 8bit 00000000 只有SoundFormat == 10时才有此8bi的字段 0x00,表示音频同步包；0x01，表示音频raw数据。
AudioObjectType 5bits [AAC LC]00010
SampleRateIndex 4bits [44100]0100
ChannelConfig 4bits [Stereo]0010
FrameLengthFlag 1bit
dependOnCoreCoder 1bit 0
extensionFlag 1bit 0

4.3.1 发送音频信息包

音频信息包不额外发送也可以，如需发送需要根据上表生成，一般共4个字节（2bytes AACDecoderSpecificInfo和2bytesAudioSpecificConfig）。

4.3.2 发送音频数据包

RTMPPacket packet;
RTMPPacket_Alloc(&packet, size + RTMP_RESERVED_HEAD_SIZE * 2);
RTMPPacket_Reset(&packet);
packet.m_packetType = RTMP_PACKET_TYPE_AUDIO;
packet.m_nChannel = 0x04;
packet.m_headerType = RTMP_PACKET_SIZE_MEDIUM;
packet.m_nTimeStamp = pts;
packet.m_hasAbsTimestamp = 0;
packet.m_nInfoField2 = _rtmp_ptr->m_stream_id;

packet.m_body[0] = 0xAF;
packet.m_body[1] = 0x01;
memcpy(packet.m_body + 2, data_ptr, size);
packet.m_nBodySize = size + 2;
if (RTMP_SendPacket(_rtmp_ptr, &packet, FALSE) < 0) {
	RTMPPacket_Free(&packet);
	return false;
}

RTMPPacket_Free(&packet);
return true;
  • RTMP包信息很多都是固定字段，使用时注意填充；
  • 为了减少内存拷贝，RTMPPacket 内存在分配时预留了 RTMP_RESERVED_HEAD_SIZE 大小；这样，在底层拆包后填充包头时之间在预留内存部分填充就可以了。



references：
FLV文件的第一个Tag: onMetaData
librtmp获取视频流和音频流1

相关标签： rtmp librtmp 直播 推流 rtmp推流h264

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