Changeset 214 for pjproject/trunk

Timestamp:

Feb 22, 2006 9:21:09 AM (19 years ago)

Author:

bennylp

Message:

Putting initial DTMF efforts

Location:

pjproject/trunk/pjmedia

Files:

• build/pjmedia.dsp (modified) (1 diff)
• include/pjmedia/errno.h (modified) (1 diff)
• include/pjmedia/rtp.h (modified) (1 diff)
• include/pjmedia/stream.h (modified) (2 diffs)
• src/pjmedia/conference.c (modified) (1 diff)
• src/pjmedia/endpoint.c (modified) (1 diff)
• src/pjmedia/stream.c (modified) (9 diffs)

pjproject/trunk/pjmedia/build/pjmedia.dsp

@@ -66 +66 @@
 # PROP Target_Dir ""
 # ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
-# 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
+# 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
 # SUBTRACT CPP /YX
 # ADD BASE RSC /l 0x409 /d "_DEBUG"

pjproject/trunk/pjmedia/include/pjmedia/errno.h

@@ -355 +355 @@
  */
 #define PJMEDIA_RTP_ENOCONFIG       (PJMEDIA_ERRNO_START+134)    /* 220134 */
+/**
+ * @hideinitializer
+ * Invalid DTMF digit.
+ */
+#define PJMEDIA_RTP_EINDTMF         (PJMEDIA_ERRNO_START+135)    /* 220135 */
 
 

pjproject/trunk/pjmedia/include/pjmedia/rtp.h

@@ -123 +123 @@
 
 
+#pragma pack(1)
+
+/**
+ * Declaration for DTMF telephony-events (RFC2833).
+ */
+struct pjmedia_rtp_dtmf_event
+{
+    pj_uint8_t  event;
+    pj_uint8_t  e_vol;
+    pj_uint16_t duration;
+};
+
+/**
+ * @see pjmedia_rtp_dtmf_event
+ */
+typedef struct pjmedia_rtp_dtmf_event pjmedia_rtp_dtmf_event;
+
+#pragma pack()
+
+
 /**
  * A generic sequence number management, used by both RTP and RTCP.

pjproject/trunk/pjmedia/include/pjmedia/stream.h

@@ -72 +72 @@
     pj_sockaddr_in      rem_addr;   /**< Remote RTP address                 */
     pjmedia_codec_info  fmt;        /**< Codec format info.                 */
+    unsigned            tx_event_pt;/**< Outgoing pt for telephone-events.  */
+    unsigned            rx_event_pt;/**< Incoming pt for telephone-events.  */
     pj_uint32_t         ssrc;       /**< RTP SSRC.                          */
     int                 jb_min;     /**< Jitter buffer min delay.           */
@@ -184 +186 @@
                                            pjmedia_dir dir);
 
+/**
+ * Transmit DTMF to this stream. The DTMF will be transmitted uisng
+ * RTP telephone-events as described in RFC 2833. This operation is
+ * only valid for audio stream.
+ *
+ * @param stream        The media stream.
+ * @param digit         A single digit ('0123456789*#ABCD').
+ *
+ * @return              PJ_SUCCESS on success.
+ */
+PJ_DECL(pj_status_t) pjmedia_stream_dial_dtmf(pjmedia_stream *stream,
+                                              const pj_str_t *digit_char);
+
+
 
 /**

pjproject/trunk/pjmedia/src/pjmedia/conference.c

@@ -747 +747 @@
             continue;
 
+        //
+        // TODO:
+        //  When there's no source, not transmit the frame, but instead
+        //  transmit a 'silence' frame. This is to allow the 'port' to
+        //  do some processing, such as updating timestamp for RTP session
+        //  or transmit signal when it's in the middle of transmitting DTMF.
+        //
 
         target_buf = (conf_port->cur_tx_buf==conf_port->tx_buf1?

pjproject/trunk/pjmedia/src/pjmedia/endpoint.c

@@ -258 +258 @@
     }
 
-    /* Add sendrect attribute. */
+    /* Add sendrecv attribute. */
     attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
     attr->name = STR_SENDRECV;
     m->attr[m->attr_count++] = attr;
 
+#if 1
+    //
+    // Test: add telephony events
+    //
+    m->desc.fmt[m->desc.fmt_count++] = pj_str("101");
+    /* Add rtpmap. */
+    attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
+    attr->name = pj_str("rtpmap");
+    attr->value = pj_str(":101 telephone-event/8000");
+    m->attr[m->attr_count++] = attr;
+    /* Add fmtp */
+    attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
+    attr->name = pj_str("fmtp");
+    attr->value = pj_str(":101 0-15");
+    m->attr[m->attr_count++] = attr;
+#endif
 
     /* Done */

pjproject/trunk/pjmedia/src/pjmedia/stream.c

@@ -23 +23 @@
 #include <pjmedia/jbuf.h>
 #include <pj/os.h>
+#include <pj/ctype.h>
 #include <pj/log.h>
 #include <pj/string.h>      /* memcpy() */
@@ -40 +41 @@
 #define PJMEDIA_MAX_BUFFER_SIZE_MS      2000
 #define PJMEDIA_MAX_MTU                 1500
-
+#define PJMEDIA_DTMF_DURATION           1600    /* in timestamp */
 
 
@@ -64 +65 @@
 
 
+struct dtmf
+{
+    int             event;
+    pj_uint32_t     start_ts;
+    pj_uint32_t     end_ts;
+};
+
 /**
  * This structure describes media stream.
@@ -94 +102 @@
     pj_bool_t                quit_flag;     /**< To signal thread exit.     */
     pj_thread_t             *thread;        /**< Jitter buffer's thread.    */
+
+    /* RFC 2833 DTMF transmission queue: */
+    int                      dtmf_count;    /**< # of digits in queue.      */
+    struct dtmf              dtmf_queue[32];/**< Outgoing dtmf queue.       */
 };
 
@@ -179 +191 @@
 
 
+/*
+ * Transmit DTMF
+ */
+static void transmit_dtmf(pjmedia_stream *stream, 
+                          struct pjmedia_frame *frame_out)
+{
+    pjmedia_rtp_dtmf_event *event;
+    struct dtmf *digit = &stream->dtmf_queue[0];
+    pj_uint32_t cur_ts;
+
+    event = frame_out->buf;
+    cur_ts = pj_ntohl(stream->enc->rtp.out_hdr.ts);
+
+    event->event = (pj_uint8_t)digit->event;
+    event->e_vol = 10;
+    event->duration = pj_htonl(cur_ts - digit->start_ts);
+}
+
 /**
  * rec_callback()
@@ -203 +233 @@
         return -1;
 
-    /* Encode. */
+    /* Number of samples in the frame */
+    ts_len = frame->size / (channel->snd_info.bits_per_sample / 8);
+
+    /* Init frame_out buffer. */
     frame_out.buf = ((char*)channel->out_pkt) + sizeof(pjmedia_rtp_hdr);
-    status = stream->codec->op->encode( stream->codec, frame, 
-                                        channel->out_pkt_size - sizeof(pjmedia_rtp_hdr), 
-                                        &frame_out);
-    if (status != 0) {
-        TRACE_((THIS_FILE, "Codec encode() error", status));
-        return status;
+
+    /* If we have DTMF digits in the queue, transmit the digits. 
+     * Otherwise encode the PCM buffer.
+     */
+    if (stream->dtmf_count) {
+        transmit_dtmf(stream, &frame_out);
+    } else {
+        unsigned max_size;
+
+        max_size = channel->out_pkt_size - sizeof(pjmedia_rtp_hdr);
+        status = stream->codec->op->encode( stream->codec, frame, 
+                                            max_size, 
+                                            &frame_out);
+        if (status != 0) {
+            TRACE_((THIS_FILE, "Codec encode() error", status));
+            return status;
+        }
     }
 
     /* Encapsulate. */
-    ts_len = frame->size / (channel->snd_info.bits_per_sample / 8);
     status = pjmedia_rtp_encode_rtp( &channel->rtp, 
                                 channel->pt, 0, 
@@ -247 +290 @@
 }
 
+
+static void dump_bin(const char *buf, unsigned len)
+{
+    unsigned i;
+
+    PJ_LOG(3,(THIS_FILE, "begin dump"));
+    for (i=0; i<len; ++i) {
+        int j;
+        char bits[9];
+        unsigned val = buf[i] & 0xFF;
+
+        bits[8] = '\0';
+        for (j=0; j<8; ++j) {
+            if (val & (1 << (7-j)))
+                bits[j] = '1';
+            else
+                bits[j] = '0';
+        }
+
+        PJ_LOG(3,(THIS_FILE, "%2d %s [%d]", i, bits, val));
+    }
+    PJ_LOG(3,(THIS_FILE, "end dump"));
+}
 
 /*
@@ -311 +377 @@
         }
 
+#if 1
+        if (hdr->pt == 101) {
+            dump_bin((char*)payload, payloadlen);
+            continue;
+        }
+#endif
+
         status = pjmedia_rtp_session_update(&channel->rtp, hdr);
         if (status != 0 && 
@@ -739 +812 @@
 }
 
+/*
+ * Dial DTMF
+ */
+PJ_DEF(pj_status_t) pjmedia_stream_dial_dtmf( pjmedia_stream *stream,
+                                              const pj_str_t *digit_char)
+{
+    pj_status_t status = PJ_SUCCESS;
+
+    /* By convention we use jitter buffer mutex to access DTMF
+     * queue.
+     */
+    PJ_ASSERT_RETURN(stream && digit_char, PJ_EINVAL);
+
+    pj_mutex_lock(stream->jb_mutex);
+    
+    if (stream->dtmf_count+digit_char->slen >=
+        PJ_ARRAY_SIZE(stream->dtmf_queue))
+    {
+        status = PJ_ETOOMANY;
+    } else {
+        int i;
+
+        /* convert ASCII digits into payload type first, to make sure
+         * that all digits are valid. 
+         */
+        for (i=0; i<digit_char->slen; ++i) {
+            unsigned pt;
+
+            if (digit_char->ptr[i] >= '0' &&
+                digit_char->ptr[i] <= '9')
+            {
+                pt = digit_char->ptr[i] - '0';
+            } 
+            else if (pj_tolower(digit_char->ptr[i]) >= 'a' &&
+                     pj_tolower(digit_char->ptr[i]) <= 'd')
+            {
+                pt = pj_tolower(digit_char->ptr[i]) - 'a' + 12;
+            }
+            else if (digit_char->ptr[i] == '*')
+            {
+                pt = 10;
+            }
+            else if (digit_char->ptr[i] == '#')
+            {
+                pt = 11;
+            }
+            else
+            {
+                status = PJMEDIA_RTP_EINDTMF;
+                break;
+            }
+
+            stream->dtmf_queue[stream->dtmf_count+1].event = pt;
+            stream->dtmf_queue[stream->dtmf_count+1].start_ts = start_ts;
+            stream->dtmf_queue[stream->dtmf_count+1].end_ts = 
+                start_ts + PJMEDIA_DTMF_DURATION;
+
+            start_ts += PJMEDIA_DTMF_DURATION + 320;
+        }
+
+        if (status != PJ_SUCCESS)
+            goto on_return;
+
+        if (stream->dtmf_count ==0) {
+            pj_uint32_t start_ts;
+
+            start_ts = pj_ntohl(stream->enc->rtp.out_hdr.ts);
+            stream->dtmf_queue[0].start_ts = start_ts;
+            stream->dtmf_queue[0].end_ts = start_ts + PJMEDIA_DTMF_DURATION;
+        }
+
+        /* Increment digit count only if all digits are valid. */
+        stream->dtmf_count += digit_char->slen;
+
+    }
+
+on_return:
+    pj_mutex_unlock(stream->jb_mutex);
+
+    return status;
+}
+