樹莓派sip視訊電話-3:exosip2+硬體h264+g711編碼初步實現
阿新 • • 發佈：2019-01-19
之前使用python語音實現，但是python下的exosip2庫部分功能不能實現，現改為c語音的exsip方式，初步實現sip視訊電話功能。
測試環境：樹莓派------------elastix---------------ekiga（pc端） 視訊為h.264 音訊為G711u
存在問題：1.原始碼中有幾處告警未處理；2.執行緒終止沒有實現，呼叫結束後，視訊執行緒仍在傳送（不能呼入兩次），音訊執行緒因對方埠關閉終止退出；3.接收視音訊還沒有實現，採用外部程式omxplayer 播放sdp檔案方式實現。4.只做了呼入視訊 音訊傳送，撥出未做。（休假快結束了，暫時就到這裡了，有時間再完善！）
1.解除安裝exosip 4.1版本
sudo apt-get remove libosip2-11 libosip2-dev libexosip2-11 libexosip2-dev
2.編譯安裝3.6版本
libosip2-3.6.0 ./configure&&make&&sudo make install
libexosip2-3.6.0 ./configure&&make&&sudo make install
3.安裝編譯視訊編碼動態連結庫
樹莓派Camkit，原始碼百度查詢，感謝作者！
4.編譯sip原始碼
（1）sip部分：使用了exsip2實現，開啟了3個執行緒，事件處理,音訊傳送,視訊傳送3個執行緒。
（2）音訊部分：為g711U編碼，在原始碼中可以改為g711a。
（3）視訊部分：使用了omx硬體編碼方式，呼叫了牛人的Camkit原始碼，動態連結方式。
5.原始碼：
（1）sip.c
#include <eXosip2/eXosip.h>
#include <stdio.h>
#include <stdlib.h>
#include <netinet/in.h>
#include <sys/socket.h>
#include <sys/types.h>
#include <pthread.h>
#include "audio_rtp.h"
#include "video_rtp.h"
eXosip_event_t *je;
osip_message_t *reg = NULL;
osip_message_t *invite = NULL;
osip_message_t *ack = NULL;
osip_message_t *info = NULL;
osip_message_t *message = NULL;
osip_message_t *answer = NULL;
sdp_message_t *remote_sdp = NULL;
sdp_connection_t * con_req = NULL;
sdp_media_t * md_audio_req = NULL;
sdp_media_t * md_video_req = NULL;
int call_id, dialog_id,calling ;
int i,flag;
int flag1 = 1;
int id;
struct osip_thread *event_thread;
struct osip_thread *audio_thread;
struct osip_thread *video_thread;
struct audio_param_t audioparam;
struct video_param_t videoparam;
char *identity = "sip: （2）video_rtp.h
資料結構為引數傳遞部分，很多設定沒有傳遞，下步再繼續。
typedef struct video_param_t
{
char *video_hw;
char *dest_ip ;
int dest_port;
} video_param_t;
void video_rtp(struct video_param_t *videoparam) ;
（3）video_rtp.c
原作者的程式，稍作改動。
#include <stdio.h>
#include <signal.h>
#include <string.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <linux/videodev2.h>
#include <sys/time.h>
#include <pthread.h>
#include <assert.h>
#include <stdint.h>
#include "camkit.h"
#include "video_rtp.h"
#define MAX_RTP_SIZE 1420
int debug = 0;
void video_rtp(struct video_param_t *videoparam)
{
printf("video_rtp:video_hw=%s,dest_ip=%s,dest_port=%d\n",videoparam->video_hw,videoparam->dest_ip,videoparam->dest_port);
struct cap_handle *caphandle = NULL;
struct cvt_handle *cvthandle = NULL;
struct enc_handle *enchandle = NULL;
struct pac_handle *pachandle = NULL;
struct net_handle *nethandle = NULL;
struct cap_param capp;
struct cvt_param cvtp;
struct enc_param encp;
struct pac_param pacp;
struct net_param netp;
U32 vfmt = V4L2_PIX_FMT_YUYV;
U32 ofmt = V4L2_PIX_FMT_YUV420;
// set default values
capp.dev_name = "/dev/video0";
capp.width = 640;
capp.height = 480;
capp.pixfmt = vfmt;
capp.rate = 15;
cvtp.inwidth = 640;
cvtp.inheight = 480;
cvtp.inpixfmt = vfmt;
cvtp.outwidth = 640;
cvtp.outheight = 480;
cvtp.outpixfmt = ofmt;
encp.src_picwidth = 640;
encp.src_picheight = 480;
encp.enc_picwidth = 640;
encp.enc_picheight = 480;
encp.chroma_interleave = 0;
encp.fps = 15;
encp.gop = 12;
encp.bitrate = 1000;
pacp.max_pkt_len = 1400;
pacp.ssrc = 1234;
netp.serip = videoparam->dest_ip;//ip地址
netp.serport =videoparam->dest_port;//埠
netp.type = UDP;
capp.width = cvtp.inwidth = cvtp.outwidth = encp.src_picwidth =encp.enc_picwidth = 640;
capp.height = cvtp.inheight = cvtp.outheight =encp.src_picheight = encp.enc_picheight = 480;
caphandle = capture_open(capp);
if (!caphandle)
{
printf("--- Open capture failed\n");
exit(1);
}
cvthandle = convert_open(cvtp);
if (!cvthandle)
{
printf("--- Open convert failed\n");
exit(1);
}
enchandle = encode_open(encp);
if (!enchandle)
{
printf("--- Open encode failed\n");
exit(1);
}
pachandle = pack_open(pacp);
if (!pachandle)
{
printf("--- Open pack failed\n");
exit(1);
}
if (netp.serip == NULL || netp.serport == -1)
{
printf("--- Server ip and port must be specified when using network\n");
exit(1);
}
nethandle = net_open(netp);
if (!nethandle)
{
printf("--- Open network failed\n");
exit(1);
}
// start capture encode loop
int ret;
void *cap_buf, *cvt_buf, *hd_buf, *enc_buf;
char *pac_buf = (char *) malloc(MAX_RTP_SIZE);
int cap_len, cvt_len, hd_len, enc_len, pac_len;
enum pic_t ptype;
struct timeval ctime, ltime;
unsigned long fps_counter = 0;
int sec, usec;
double stat_time = 0;
capture_start(caphandle); // !!! need to start capture stream!
gettimeofday(<ime, NULL);
while (1)
{
if (debug) // print fps
{
gettimeofday(&ctime, NULL);
sec = ctime.tv_sec - ltime.tv_sec;
usec = ctime.tv_usec - ltime.tv_usec;
if (usec < 0)
{
sec--;
usec = usec + 1000000;
}
stat_time = (sec * 1000000) + usec; // diff in microsecond
if (stat_time >= 1000000) // >= 1s
{
printf("\n*** FPS: %ld\n", fps_counter);
fps_counter = 0;
ltime = ctime;
}
fps_counter++;
}
ret = capture_get_data(caphandle, &cap_buf, &cap_len);
if (ret != 0)
{
if (ret < 0) // error
{
printf("--- capture_get_data failed\n");
break;
}
else // again
{
usleep(10000);
continue;
}
}
if (cap_len <= 0)
{
printf("!!! No capture data\n");
continue;
}
if (debug)
fputc('.', stdout);
// convert
if (capp.pixfmt == V4L2_PIX_FMT_YUV420) // no need to convert
{
cvt_buf = cap_buf;
cvt_len = cap_len;
}
else // do convert: YUYV => YUV420
{
ret = convert_do(cvthandle, cap_buf, cap_len, &cvt_buf, &cvt_len);
if (ret < 0)
{
printf("--- convert_do failed\n");
break;
}
if (cvt_len <= 0)
{
printf("!!! No convert data\n");
continue;
}
}
if (debug)
fputc('-', stdout);
// encode
// fetch h264 headers first!
while ((ret = encode_get_headers(enchandle, &hd_buf, &hd_len, &ptype))
!= 0)
{
if (debug)
fputc('S', stdout);
// pack headers
pack_put(pachandle, hd_buf, hd_len);
while (pack_get(pachandle, pac_buf, MAX_RTP_SIZE, &pac_len) == 1)
{
if (debug)
fputc('#', stdout);
// network
ret = net_send(nethandle, pac_buf, pac_len);
if (ret != pac_len)
{
printf("send pack failed, size: %d, err: %s\n", pac_len,
strerror(errno));
}
if (debug)
fputc('>', stdout);
}
}
ret = encode_do(enchandle, cvt_buf, cvt_len, &enc_buf, &enc_len,
&ptype);
if (ret < 0)
{
printf("--- encode_do failed\n");
break;
}
if (enc_len <= 0)
{
printf("!!! No encode data\n");
continue;
}
if (debug)
{
char c;
switch (ptype)
{
case PPS:
c = 'S';
break;
case SPS:
c = 'S';
break;
case I:
c = 'I';
break;
case P:
c = 'P';
break;
case B:
c = 'B';
break;
default:
c = 'N';
break;
}
fputc(c, stdout);
}
// pack
pack_put(pachandle, enc_buf, enc_len);
while (pack_get(pachandle, pac_buf, MAX_RTP_SIZE, &pac_len) == 1)
{
if (debug)
fputc('#', stdout);
// network
ret = net_send(nethandle, pac_buf, pac_len);
if (ret != pac_len)
{
printf("send pack failed, size: %d, err: %s\n", pac_len,
strerror(errno));
}
if (debug)
fputc('>', stdout);
}
}
capture_stop(caphandle);
free(pac_buf);
net_close(nethandle);
pack_close(pachandle);
encode_close(enchandle);
convert_close(cvthandle);
capture_close(caphandle);
exit(0);
}
（4）audio_rtp.h
typedef struct
{
/** byte 0 */
unsigned char csrc_len:4; /** expect 0 */
unsigned char extension:1; /** expect 1, see RTP_OP below */
unsigned char padding:1; /** expect 0 */
unsigned char version:2; /** expect 2 */
/** byte 1 */
unsigned char payload:7; /** stream type */
unsigned char marker:1; /** when send the first framer,set it */
/** bytes 2, 3 */
unsigned short seq_no;
/** bytes 4-7 */
unsigned long timestamp;
/** bytes 8-11 */
unsigned long ssrc; /** stream number is used here. */
} RTP_FIXED_HEADER;
typedef struct audio_param_t
{
char *audio_hw;
char *dest_ip ;
int dest_port;
} audio_param_t;
void audio_rtp(struct audio_param_t *audioparam) ;
（5）audio_rtp.c
/*
This example reads from the default PCM device
and G711 code and rtp send.
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <memory.h>
#include <netdb.h>
#include <time.h>
#include <sys/types.h>
#include <netinet/in.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <unistd.h>
#include <alsa/asoundlib.h>
#include <math.h>
#include "g711codec.h"
#include "audio_rtp.h"
#define BUFFERSIZE 4096
#define PERIOD_SIZE 1024
#define PERIODS 2
#define SAMPLE_RATE 8000
#define CHANNELS 1
#define FSIZE 2*CHANNELS
/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API
void audio_rtp(struct audio_param_t *audioparam) {
//char *audio_hw, char *dest_ip, int dest_port
printf("audio_rtp:audio_hw=%s,dest_ip=%s,dest_port=%d\n",audioparam->audio_hw,audioparam->dest_ip,audioparam->dest_port);
int rc; //return code.
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
snd_pcm_uframes_t frames;
char *buffer;
int err;
/* Open PCM device for recording (capture). */
err = snd_pcm_open(&handle, audioparam->audio_hw , SND_PCM_STREAM_CAPTURE, 0);
if (err < 0) {
fprintf(stderr,"unable to open pcm device: %s\n",snd_strerror(err));
exit(1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
err=snd_pcm_hw_params_any(handle, params);
if (err < 0) {
fprintf(stderr, "Can not configure this PCM device: %s\n",snd_strerror(err));
exit(1);
}
/* Set the desired hardware parameters. */
/* Interleaved mode */
err=snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
fprintf(stderr, "Failed to set PCM device to interleaved: %s\n", snd_strerror(err));
exit(1);
}
/* Signed 16-bit little-endian format */
//err=snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_MU_LAW);
err=snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_S16_LE);
if (err < 0) {
fprintf(stderr,"Failed to set PCM device to 16-bit signed PCM: %s\n", snd_strerror(err));
exit(1);
}
/* One channels (mono) */
err=snd_pcm_hw_params_set_channels(handle, params, CHANNELS);
if (err < 0) {
fprintf(stderr, "Failed to set PCM device to mono: %s\n",snd_strerror(err));
exit(1);
}
/* 8000 bits/second sampling rate (CD quality) */
val = SAMPLE_RATE;//這裡修改為8000
err=snd_pcm_hw_params_set_rate_near(handle, params,&val, &dir);
if (err < 0) {
fprintf(stderr, "Failed to set PCM device to sample rate =%d: %s\n",val,snd_strerror(err));
exit(1);
}
/* Set buffer time 500000. */
unsigned int buffer_time,period_time;
snd_pcm_hw_params_get_buffer_time_max(params, &buffer_time, 0);
if ( buffer_time >500000)
buffer_time = 80000;//這裡可修改
period_time = buffer_time/4;//這裡可修改 size = frames * FSIZE;
err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, 0);
if (err < 0) {
fprintf(stderr, "Failed to set PCM device to buffer time =%d: %s\n", buffer_time,snd_strerror(err));
exit(1);
}
err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, 0);
if (err < 0) {
fprintf(stderr, "Failed to set PCM device to period time =%d: %s\n",period_time,snd_strerror(err));
exit(1);
}
/* Write the parameters to the driver */
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
fprintf(stderr,"unable to set hw parameters: %s\n",snd_strerror(err));
exit(1);
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params,&frames, &dir);
size = frames * FSIZE; /* 2 bytes/sample, 1 channels *///這裡應該=320 FSIZE=2 frames=160
buffer = (char *) malloc(size);
printf("read buffer size = %d\n",size);
printf("period size = %d frames\n", (int)frames);
/*print alsa config parameter*/
snd_pcm_hw_params_get_period_time(params,&val, &dir);
printf("period time is: %d\n",val);
snd_pcm_hw_params_get_buffer_time(params, &val, &dir);
printf("buffer time = %d us\n", val);
snd_pcm_hw_params_get_buffer_size(params, (snd_pcm_uframes_t *) &val);
printf("buffer size = %d frames\n", val);
snd_pcm_hw_params_get_periods(params, &val, &dir);
printf("periods per buffer = %d frames\n", val);
int M_bit=1;
char sendbuf[1500];
memset(sendbuf,0,1500);
unsigned short seq_num = 0;
RTP_FIXED_HEADER *rtp_hdr;
int rtp_socket;
struct sockaddr_in server;
int len = sizeof(server);
unsigned int timestamp_increse = 0,ts_current = 0;
timestamp_increse = 160;
server.sin_family = AF_INET;
server.sin_port = htons(audioparam->dest_port);
server.sin_addr.s_addr = inet_addr(audioparam->dest_ip);
rtp_socket = socket(AF_INET,SOCK_DGRAM,0);
connect(rtp_socket, (struct sockaddr *)&server, len);
while (1) {
//1.採集
rc = snd_pcm_readi(handle, buffer, frames);//採集音訊資料
if (rc == -EPIPE) {
fprintf(stderr, "overrun occurred\n");
err=snd_pcm_prepare(handle);
if( err <0){
fprintf(stderr, "Failed to recover form overrun : %s\n",
snd_strerror(err));
exit(1);
}
}
else if (rc < 0) {
fprintf(stderr,"error from read: %s\n",snd_strerror(rc));
exit(1);
}
else if (rc != (int)frames) {
fprintf(stderr, "short read, read %d frames\n", rc);
}
//2.編碼
rc = PCM2G711u( (char *)buffer, (char *)&sendbuf[12], size, 0 );//pcm轉g711a
if(rc<0) fprintf(stderr,"PCM2G711u error:rc=%d\n",rc);
//3.打包
rtp_hdr =(RTP_FIXED_HEADER*)&sendbuf[0];
rtp_hdr->payload = 0; //負載型別號，
rtp_hdr->version = 2; //版本號，此版本固定為2
if(1 == M_bit) {
rtp_hdr->marker = 1; //標誌位，由具體協議規定其值。
M_bit = 0;
}
else{
rtp_hdr->marker = 0; //標誌位，由具體協議規定其值。
}
rtp_hdr->ssrc = htonl(10); //隨機指定為10，並且在本RTP會話中全域性唯一
rtp_hdr->seq_no = htons(seq_num ++);//rtp包序號
ts_current = ts_current+timestamp_increse;
rtp_hdr->timestamp=htonl(ts_current);//rtp傳輸時間戳，增量為timestamp_increse=160
//4.傳送
rc = send( rtp_socket, sendbuf, rc+12, 0 );//開始傳送rtp包，+12是rtp的包頭+g711荷載
if(rc<0) {
fprintf(stderr,"net send error=%d\n",rc);
break;
}
memset(sendbuf,0,1500);//清空sendbuf；此時會將上次的時間戳清空，因此需要ts_current來儲存上次的時間戳值
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
}
/*
* G711 encode decode HEADER.
*/
#ifndef __G711CODEC_H__
#define __G711CODEC_H__
/*
* u-law, A-law and linear PCM conversions.
*/
#define SIGN_BIT (0x80) /* Sign bit for a A-law byte. */
#define QUANT_MASK (0xf) /* Quantization field mask. */
#define NSEGS (8) /* Number of A-law segments. */
#define SEG_SHIFT (4) /* Left shift for segment number. */
#define SEG_MASK (0x70) /* Segment field mask. */
#define BIAS (0x84) /* Bias for linear code. */
int PCM2G711a( char *InAudioData, char *OutAudioData, int DataLen, int reserve );
int PCM2G711u( char *InAudioData, char *OutAudioData, int DataLen, int reserve );
int G711a2PCM( char *InAudioData, char *OutAudioData, int DataLen, int reserve );
int G711u2PCM( char *InAudioData, char *OutAudioData, int DataLen, int reserve );
int g711a_decode(short amp[], const unsigned char g711a_data[], int g711a_bytes);
int g711u_decode(short amp[], const unsigned char g711u_data[], int g711u_bytes);
int g711a_encode(unsigned char g711_data[], const short amp[], int len);
int g711u_encode(unsigned char g711_data[], const short amp[], int len);
#endif /* g711codec.h */
#include "g711codec.h"
static short seg_end[8] = {0xFF, 0x1FF, 0x3FF, 0x7FF,
0xFFF, 0x1FFF, 0x3FFF, 0x7FFF};
static int search(int val, short *table, int size)
{
int i;
for (i = 0; i < size; i++) {
if (val <= *table++)
return (i);
}
return (size);
}
/*
* alaw2linear() - Convert an A-law value to 16-bit linear PCM
*
*/
static int alaw2linear( unsigned char a_val )
{
int t;
int seg;
a_val ^= 0x55;
t = (a_val & QUANT_MASK) << 4;
seg = ( (unsigned)a_val & SEG_MASK ) >> SEG_SHIFT;
switch (seg)
{
case 0:
t += 8;
break;
case 1:
t += 0x108;
break;
default:
t += 0x108;
t <<= seg - 1;
}
return ((a_val & SIGN_BIT) ? t : -t);
}
/*
* ulaw2linear() - Convert a u-law value to 16-bit linear PCM
*
* First, a biased linear code is derived from the code word. An unbiased
* output can then be obtained by subtracting 33 from the biased code.
*
* Note that this function expects to be passed the complement of the
* original code word. This is in keeping with ISDN conventions.
*/
static int ulaw2linear(unsigned char u_val)
{
int t;
/* Complement to obtain normal u-law value. */
u_val = ~u_val;
/*
* Extract and bias the quantization bits. Then
* shift up by the segment number and subtract out the bias.
*/
t = ((u_val & QUANT_MASK) << 3) + BIAS;
t <<= ((unsigned)u_val & SEG_MASK) >> SEG_SHIFT;
return ((u_val & SIGN_BIT) ? (BIAS - t) : (t - BIAS));
}
/*
* linear2alaw() - Convert a 16-bit linear PCM value to 8-bit A-law
*
*/
unsigned char linear2alaw(int pcm_val) /* 2's complement (16-bit range) */
{
int mask;
int seg;
unsigned char aval;
if (pcm_val >= 0) {
mask = 0xD5; /* sign (7th) bit = 1 */
} else {
mask = 0x55; /* sign bit = 0 */
pcm_val = -pcm_val - 8;
}
/* Convert the scaled magnitude to segment number. */
seg = search(pcm_val, seg_end, 8);
/* Combine the sign, segment, and quantization bits. */
if (seg >= 8) /* out of range, return maximum value. */
return (0x7F ^ mask);
else {
aval = seg << SEG_SHIFT;
if (seg < 2)
aval |= (pcm_val >> 4) & QUANT_MASK;
else
aval |= (pcm_val >> (seg + 3)) & QUANT_MASK;
return (aval ^ mask);
}
}
/*
* linear2ulaw() - Convert a linear PCM value to u-law
*
*/
unsigned char linear2ulaw(int pcm_val) /* 2's complement (16-bit range) */
{
int mask;
int seg;
unsigned char uval;
/* Get the sign and the magnitude of the value. */
if (pcm_val < 0) {
pcm_val = BIAS - pcm_val;
mask = 0x7F;
} else {
pcm_val += BIAS;
mask = 0xFF;
}
/* Convert the scaled magnitude to segment number. */
seg = search(pcm_val, seg_end, 8);
/*
* Combine the sign, segment, quantization bits;
* and complement the code word.
*/
if (seg >= 8) /* out of range, return maximum value. */
return (0x7F ^ mask);
else {
uval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0xF);
return (uval ^ mask);
}
}
int g711a_decode( short amp[], const unsigned char g711a_data[], int g711a_bytes )
{
int i;
int samples;
unsigned char code;
int sl;
for ( samples = i = 0; ; )
{
if (i >= g711a_bytes)
break;
code = g711a_data[i++];
sl = alaw2linear( code );
amp[samples++] = (short) sl;
}
return samples*2;
}
int g711u_decode(short amp[], const unsigned char g711u_data[], int g711u_bytes)
{
int i;
int samples;
unsigned char code;
int sl;
for (samples = i = 0;;)
{
if (i >= g711u_bytes)
break;
code = g711u_data[i++];
sl = ulaw2linear(code);
amp[samples++] = (short) sl;
}
return samples*2;
}
int g711a_encode(unsigned char g711_data[], const short amp[], int len)
{
int i;
for (i = 0; i < len; i++)
{
g711_data[i] = linear2alaw(amp[i]);
}
return len;
}
int g711u_encode(unsigned char g711_data[], const short amp[], int len)
{
int i;
for (i = 0; i < len; i++)
{
g711_data[i] = linear2ulaw(amp[i]);
}
return len;
}
（8）g711.c
#include <stdio.h>
#include "g711codec.h"
/*
* function: convert PCM audio format to g711 alaw/ulaw.(zqj)
* InAudioData: PCM data prepared for encoding to g711 alaw/ulaw.
* OutAudioData: encoded g711 alaw/ulaw.
* DataLen: PCM data size.
* reserve: reserved param, no use.
*/
/*alaw*/
int PCM2G711a( char *InAudioData, char *OutAudioData, int DataLen, int reserve )
{
//check params.
if( (NULL == InAudioData) && (NULL == OutAudioData) && (0 == DataLen) )
{
printf("Error, empty data or transmit failed, exit !\n");
return -1;
}
//printf("DataLen = %d, %s, %d\n", DataLen, __func__, __LINE__);
int Retaen = 0;
//printf("G711a encode start......\n");
Retaen = g711a_encode( (unsigned char *)OutAudioData, (short*)InAudioData, DataLen/2 );
//printf("Retaen = %d, %s, %d\n", Retaen, __func__, __LINE__);
return Retaen; //index successfully encoded data len.
}
/*ulaw*/
int PCM2G711u( char *InAudioData, char *OutAudioData, int DataLen, int reserve )
{
//check params.
if( (NULL == InAudioData) && (NULL == OutAudioData) && (0 == DataLen) )
{
printf("Error, empty data or transmit failed, exit !\n");
return -1;
}
//printf("DataLen = %d, %s, %d\n", DataLen, __func__, __LINE__);
int Retuen = 0;
//printf("G711u encode start......\n");
Retuen = g711u_encode( (unsigned char *)OutAudioData, (short*)InAudioData, DataLen/2 );
//printf("Retuen = %d, %s, %d\n", Retuen, __func__, __LINE__);
return Retuen;
}
/*
* function: convert g711 alaw audio format to PCM.(zqj)
* InAudioData: g711 alaw data prepared for encoding to PCM.
* OutAudioData: encoded PCM audio data.
* DataLen: g711a data size.
* reserve: reserved param, no use.
*/
/*alaw*/
int G711a2PCM( char *InAudioData, char *OutAudioData, int DataLen, int reserve )
{
//check param.
if( (NULL == InAudioData) && (NULL == OutAudioData) && (0 == DataLen) )
{
printf("Error, empty data or transmit failed, exit !\n");
return -1;
}
printf("DataLen = %d, %s, %d\n", DataLen, __func__, __LINE__);
int Retade = 0;
printf("G711a decode start......\n");
Retade = g711a_decode( (short*)OutAudioData, (unsigned char *)InAudioData, DataLen );
printf("Retade = %d, %s, %d\n", Retade, __func__, __LINE__);
return Retade; //index successfully decoded data len.
}
/*ulaw*/
int G711u2PCM( char *InAudioData, char *OutAudioData, int DataLen, int reserve )
{
//check param.
if( (NULL == InAudioData) && (NULL == OutAudioData) && (0 == DataLen) )
{
printf("Error, empty data or transmit failed, exit !\n");
return -1;
}
printf("DataLen = %d, %s, %d\n", DataLen, __func__, __LINE__);
int Retude = 0;
printf("G711u decode start......\n");
Retude = g711u_decode( (short*)OutAudioData, (unsigned char *)InAudioData, DataLen );
printf("Retude = %d, %s, %d\n", Retude, __func__, __LINE__);
return Retude;
}
（9）makefile
CC=gcc
CCFLAGS=-g -Wall
LDFLAGS=-leXosip2 -losip2 -losipparser2 -lpthread -lasound -lcamkit
all:sip
sip:sip.c
$(CC) sip.c video_rtp.c audio_rtp.c g711.c g711codec.c $(CCFLAGS) $(LDFLAGS) -o sip
clean:
rm sip6.執行測試
（1）執行後，r 註冊伺服器；（2）ekiga呼入，自動應答建立連結。
[email protected]:~/project$ ./sip
r 向伺服器註冊
c 取消註冊
i 發起呼叫請求
h 結束通話
q 退出程式
s 執行方法INFO
m 執行方法MESSAGE
event_thread created!
please input the comand:
EXOSIP_MESSAGE_NEW!
EXOSIP_MESSAGE_NEW!
r
start register!
please input the comand:
EXOSIP_REGISTRATION_FAILURE---401 error!
registrered successfully
EXOSIP_MESSAGE_NEW!
options
EXOSIP_MESSAGE_NEW!
EXOSIP_MESSAGE_NEW!
Received a INVITE msg from 192.168.1.101:5066, UserName is 8006, password is (null)
send 200 over!
the INFO is :
remote_sdp_str=======================
v=0
o=root 120365001 120365001 IN IP4 192.168.1.200
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.1.200
b=CT:8096
t=0 0
m=audio 19874 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 11520 RTP/AVP 99 34
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801E;max-mbps=190080;max-fs=6336;packetization-mode=1
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
a=nortpproxy:yes
audio info:----------------
payload_str=0,m_media=0 PCMU/8000
payload_str=8,m_media=8 PCMA/8000
payload_str=3,m_media=3 GSM/8000
payload_str=9,m_media=9 G722/8000
payload_str=101,m_media=101 telephone-event/8000
video info:----------------
payload_str=99,m_media=99 H264/90000
payload_str=34,m_media=99 profile-level-id=42801E;max-mbps=190080;max-fs=6336;packetization-mode=1
audio video port info:--------------
conn_add=192.168.1.200,audio_port=19874,video_port=11520
ACK received!
conn_add=192.168.1.200,audio_port=19874
audio_thread created!
audio_rtp:audio_hw=hw:1,0,dest_ip=192.168.1.200,dest_port=19874
conn_add=192.168.1.200,audio_port=19874
video_thread created!
video_rtp:video_hw=/dev/video0,dest_ip=192.168.1.200,dest_port=11520
+++ Capture opened
!!! Set crop to (0, 0, 640, 480) failed. Don't panic, not all capture device support crop!
!!! VIDIOC_S_INPUT failed
+++ Device initialized
+++ Capture Opened
+++ Convert Opened
read buffer size = 320
period size = 160 frames
period time is: 20000
buffer time = 80000 us
buffer size = 640 frames
periods per buffer = 4 frames
[libx264 @ 0x73d13830] using cpu capabilities: ARMv6 NEON
[libx264 @ 0x73d13830] profile Main, level 2.2
+++ Encode Opened
+++ Pack Opened
+++ Network Opened
+++ Capture Started
!!! encoded frame delayed!
!!! No encode data
other response!
other response!
other response!
other response!
other response!
other response!
other response!
other response!
other response!
the call sid closed!
net send error=-1
send pack failed, size: 157, err: Connection refused
send pack failed, size: 1193, err: Connection refused
^C
問題：
the call sid closed! //呼叫結束
net send error=-1 //音訊執行緒出錯退出
send pack failed, size: 157, err: Connection refused//視訊執行緒遠端埠被關閉，連線拒絕。
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