Putting initial DTMF efforts
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@214 74dad513-b988-da41-8d7b-12977e46ad98
This commit is contained in:
Benny Prijono
parent
f256ae7b37
commit
349a0f25f3
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@ -65,7 +65,7 @@ LIB32=link.exe -lib
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# PROP Intermediate_Dir ".\output\pjmedia-i386-win32-vc6-debug"
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# PROP Target_Dir ""
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# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
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# ADD CPP /nologo /MTd /W4 /Gm /GX /ZI /Od /I "../include" /I "../../pjlib/include" /I "../../pjlib-util/include" /I "../src/pjmedia/portaudio" /D "_DEBUG" /D "PA_NO_ASIO" /D "PA_NO_WIN_DS" /D "WIN32" /D "_MBCS" /D "_LIB" /D PJ_WIN32=1 /D PJ_M_I386=1 /FR /FD /GZ /c
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# ADD CPP /nologo /MTd /W4 /Gm /GX /ZI /Od /I "../include" /I "../../pjlib/include" /I "../../pjlib-util/include" /I "../src/pjmedia/portaudio" /D "_DEBUG" /D "PA_NO_ASIO" /D "WIN32" /D "_MBCS" /D "_LIB" /D PJ_WIN32=1 /D PJ_M_I386=1 /FR /FD /GZ /c
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# SUBTRACT CPP /YX
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# ADD BASE RSC /l 0x409 /d "_DEBUG"
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# ADD RSC /l 0x409 /d "_DEBUG"
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@ -354,6 +354,11 @@ PJ_BEGIN_DECL
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* RTP is not configured.
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*/
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#define PJMEDIA_RTP_ENOCONFIG (PJMEDIA_ERRNO_START+134) /* 220134 */
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/**
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* @hideinitializer
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* Invalid DTMF digit.
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*/
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#define PJMEDIA_RTP_EINDTMF (PJMEDIA_ERRNO_START+135) /* 220135 */
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/************************************************************
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@ -122,6 +122,26 @@ struct pjmedia_rtp_ext_hdr
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typedef struct pjmedia_rtp_ext_hdr pjmedia_rtp_ext_hdr;
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#pragma pack(1)
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/**
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* Declaration for DTMF telephony-events (RFC2833).
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*/
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struct pjmedia_rtp_dtmf_event
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{
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pj_uint8_t event;
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pj_uint8_t e_vol;
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pj_uint16_t duration;
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};
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/**
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* @see pjmedia_rtp_dtmf_event
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*/
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typedef struct pjmedia_rtp_dtmf_event pjmedia_rtp_dtmf_event;
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#pragma pack()
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/**
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* A generic sequence number management, used by both RTP and RTCP.
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*/
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@ -71,6 +71,8 @@ struct pjmedia_stream_info
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pjmedia_sock_info sock_info; /**< Media transport (RTP/RTCP sockets) */
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pj_sockaddr_in rem_addr; /**< Remote RTP address */
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pjmedia_codec_info fmt; /**< Codec format info. */
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unsigned tx_event_pt;/**< Outgoing pt for telephone-events. */
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unsigned rx_event_pt;/**< Incoming pt for telephone-events. */
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pj_uint32_t ssrc; /**< RTP SSRC. */
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int jb_min; /**< Jitter buffer min delay. */
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int jb_max; /**< Jitter buffer max delay. */
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@ -183,6 +185,20 @@ PJ_DECL(pj_status_t) pjmedia_stream_pause( pjmedia_stream *stream,
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PJ_DECL(pj_status_t) pjmedia_stream_resume(pjmedia_stream *stream,
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pjmedia_dir dir);
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/**
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* Transmit DTMF to this stream. The DTMF will be transmitted uisng
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* RTP telephone-events as described in RFC 2833. This operation is
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* only valid for audio stream.
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*
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* @param stream The media stream.
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* @param digit A single digit ('0123456789*#ABCD').
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*
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* @return PJ_SUCCESS on success.
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*/
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PJ_DECL(pj_status_t) pjmedia_stream_dial_dtmf(pjmedia_stream *stream,
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const pj_str_t *digit_char);
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/**
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* @}
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@ -746,6 +746,13 @@ static pj_status_t play_cb( /* in */ void *user_data,
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if (!conf_port)
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continue;
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//
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// TODO:
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// When there's no source, not transmit the frame, but instead
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// transmit a 'silence' frame. This is to allow the 'port' to
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// do some processing, such as updating timestamp for RTP session
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// or transmit signal when it's in the middle of transmitting DTMF.
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//
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target_buf = (conf_port->cur_tx_buf==conf_port->tx_buf1?
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conf_port->tx_buf2 : conf_port->tx_buf1);
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@ -257,11 +257,27 @@ PJ_DEF(pj_status_t) pjmedia_endpt_create_sdp( pjmedia_endpt *endpt,
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}
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/* Add sendrect attribute. */
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/* Add sendrecv attribute. */
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attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
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attr->name = STR_SENDRECV;
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m->attr[m->attr_count++] = attr;
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#if 1
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//
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// Test: add telephony events
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//
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m->desc.fmt[m->desc.fmt_count++] = pj_str("101");
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/* Add rtpmap. */
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attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
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attr->name = pj_str("rtpmap");
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attr->value = pj_str(":101 telephone-event/8000");
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m->attr[m->attr_count++] = attr;
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/* Add fmtp */
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attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
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attr->name = pj_str("fmtp");
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attr->value = pj_str(":101 0-15");
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m->attr[m->attr_count++] = attr;
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#endif
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/* Done */
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*p_sdp = sdp;
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@ -22,6 +22,7 @@
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#include <pjmedia/rtcp.h>
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#include <pjmedia/jbuf.h>
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#include <pj/os.h>
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#include <pj/ctype.h>
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#include <pj/log.h>
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#include <pj/string.h> /* memcpy() */
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#include <pj/pool.h>
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@ -39,7 +40,7 @@
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#define PJMEDIA_MAX_FRAME_DURATION_MS 200
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#define PJMEDIA_MAX_BUFFER_SIZE_MS 2000
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#define PJMEDIA_MAX_MTU 1500
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#define PJMEDIA_DTMF_DURATION 1600 /* in timestamp */
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/**
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@ -63,6 +64,13 @@ struct pjmedia_channel
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};
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struct dtmf
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{
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int event;
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pj_uint32_t start_ts;
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pj_uint32_t end_ts;
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};
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/**
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* This structure describes media stream.
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* A media stream is bidirectional media transmission between two endpoints.
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@ -93,6 +101,10 @@ struct pjmedia_stream
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pj_bool_t quit_flag; /**< To signal thread exit. */
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pj_thread_t *thread; /**< Jitter buffer's thread. */
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/* RFC 2833 DTMF transmission queue: */
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int dtmf_count; /**< # of digits in queue. */
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struct dtmf dtmf_queue[32];/**< Outgoing dtmf queue. */
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};
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@ -178,6 +190,24 @@ static pj_status_t get_frame( pjmedia_port *port, pjmedia_frame *frame)
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}
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/*
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* Transmit DTMF
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*/
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static void transmit_dtmf(pjmedia_stream *stream,
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struct pjmedia_frame *frame_out)
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{
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pjmedia_rtp_dtmf_event *event;
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struct dtmf *digit = &stream->dtmf_queue[0];
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pj_uint32_t cur_ts;
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event = frame_out->buf;
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cur_ts = pj_ntohl(stream->enc->rtp.out_hdr.ts);
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event->event = (pj_uint8_t)digit->event;
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event->e_vol = 10;
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event->duration = pj_htonl(cur_ts - digit->start_ts);
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}
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/**
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* rec_callback()
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*
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@ -202,18 +232,31 @@ static pj_status_t put_frame( pjmedia_port *port,
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if (stream->quit_flag)
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return -1;
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/* Encode. */
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/* Number of samples in the frame */
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ts_len = frame->size / (channel->snd_info.bits_per_sample / 8);
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/* Init frame_out buffer. */
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frame_out.buf = ((char*)channel->out_pkt) + sizeof(pjmedia_rtp_hdr);
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status = stream->codec->op->encode( stream->codec, frame,
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channel->out_pkt_size - sizeof(pjmedia_rtp_hdr),
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&frame_out);
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if (status != 0) {
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TRACE_((THIS_FILE, "Codec encode() error", status));
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return status;
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/* If we have DTMF digits in the queue, transmit the digits.
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* Otherwise encode the PCM buffer.
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*/
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if (stream->dtmf_count) {
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transmit_dtmf(stream, &frame_out);
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} else {
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unsigned max_size;
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max_size = channel->out_pkt_size - sizeof(pjmedia_rtp_hdr);
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status = stream->codec->op->encode( stream->codec, frame,
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max_size,
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&frame_out);
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if (status != 0) {
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TRACE_((THIS_FILE, "Codec encode() error", status));
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return status;
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}
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}
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/* Encapsulate. */
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ts_len = frame->size / (channel->snd_info.bits_per_sample / 8);
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status = pjmedia_rtp_encode_rtp( &channel->rtp,
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channel->pt, 0,
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frame_out.size, ts_len,
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@ -247,6 +290,29 @@ static pj_status_t put_frame( pjmedia_port *port,
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}
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static void dump_bin(const char *buf, unsigned len)
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{
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unsigned i;
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PJ_LOG(3,(THIS_FILE, "begin dump"));
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for (i=0; i<len; ++i) {
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int j;
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char bits[9];
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unsigned val = buf[i] & 0xFF;
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bits[8] = '\0';
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for (j=0; j<8; ++j) {
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if (val & (1 << (7-j)))
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bits[j] = '1';
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else
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bits[j] = '0';
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}
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PJ_LOG(3,(THIS_FILE, "%2d %s [%d]", i, bits, val));
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}
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PJ_LOG(3,(THIS_FILE, "end dump"));
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}
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/*
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* This thread will poll the socket for incoming packets, and put
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* the packets to jitter buffer.
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@ -310,6 +376,13 @@ static int PJ_THREAD_FUNC jitter_buffer_thread (void*arg)
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continue;
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}
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#if 1
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if (hdr->pt == 101) {
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dump_bin((char*)payload, payloadlen);
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continue;
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}
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#endif
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status = pjmedia_rtp_session_update(&channel->rtp, hdr);
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if (status != 0 &&
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status != PJMEDIA_RTP_ESESSPROBATION &&
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@ -738,3 +811,85 @@ PJ_DEF(pj_status_t) pjmedia_stream_resume( pjmedia_stream *stream,
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return PJ_SUCCESS;
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}
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/*
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* Dial DTMF
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*/
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PJ_DEF(pj_status_t) pjmedia_stream_dial_dtmf( pjmedia_stream *stream,
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const pj_str_t *digit_char)
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{
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pj_status_t status = PJ_SUCCESS;
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/* By convention we use jitter buffer mutex to access DTMF
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* queue.
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*/
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PJ_ASSERT_RETURN(stream && digit_char, PJ_EINVAL);
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pj_mutex_lock(stream->jb_mutex);
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if (stream->dtmf_count+digit_char->slen >=
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PJ_ARRAY_SIZE(stream->dtmf_queue))
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{
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status = PJ_ETOOMANY;
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} else {
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int i;
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/* convert ASCII digits into payload type first, to make sure
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* that all digits are valid.
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*/
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for (i=0; i<digit_char->slen; ++i) {
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unsigned pt;
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if (digit_char->ptr[i] >= '0' &&
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digit_char->ptr[i] <= '9')
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{
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pt = digit_char->ptr[i] - '0';
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}
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else if (pj_tolower(digit_char->ptr[i]) >= 'a' &&
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pj_tolower(digit_char->ptr[i]) <= 'd')
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{
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pt = pj_tolower(digit_char->ptr[i]) - 'a' + 12;
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}
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else if (digit_char->ptr[i] == '*')
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{
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pt = 10;
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}
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else if (digit_char->ptr[i] == '#')
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{
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pt = 11;
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}
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else
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{
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status = PJMEDIA_RTP_EINDTMF;
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break;
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}
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stream->dtmf_queue[stream->dtmf_count+1].event = pt;
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stream->dtmf_queue[stream->dtmf_count+1].start_ts = start_ts;
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stream->dtmf_queue[stream->dtmf_count+1].end_ts =
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start_ts + PJMEDIA_DTMF_DURATION;
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start_ts += PJMEDIA_DTMF_DURATION + 320;
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}
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if (status != PJ_SUCCESS)
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goto on_return;
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if (stream->dtmf_count ==0) {
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pj_uint32_t start_ts;
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start_ts = pj_ntohl(stream->enc->rtp.out_hdr.ts);
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stream->dtmf_queue[0].start_ts = start_ts;
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stream->dtmf_queue[0].end_ts = start_ts + PJMEDIA_DTMF_DURATION;
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}
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/* Increment digit count only if all digits are valid. */
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stream->dtmf_count += digit_char->slen;
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}
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on_return:
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pj_mutex_unlock(stream->jb_mutex);
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return status;
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}
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