static struct ast_frame *g723_read(struct ast_filestream *s, int *whennext)
{
unsigned short size;
int res;
int delay;
/* Read the delay for the next packet, and schedule again if necessary */
/* XXX is this ignored ? */
if (fread(&delay, 1, 4, s->f) == 4)
delay = ntohl(delay);
else
delay = -1;
if (fread(&size, 1, 2, s->f) != 2) {
/* Out of data, or the file is no longer valid. In any case
go ahead and stop the stream */
return NULL;
}
/* Looks like we have a frame to read from here */
size = ntohs(size);
if (size > G723_MAX_SIZE) {
ast_log(LOG_WARNING, "Size %d is invalid\n", size);
/* The file is apparently no longer any good, as we
shouldn't ever get frames even close to this
size. */
return NULL;
}
/* Read the data into the buffer */
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, size);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != size) {
ast_log(LOG_WARNING, "Short read (%d of %d bytes) (%s)!\n", res, size, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = 240;
return &s->fr;
}
static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext)
{
/* Send a frame from the file to the appropriate channel */
struct wavg_desc *fs = (struct wavg_desc *)s->_private;
s->fr.frametype = AST_FRAME_VOICE;
ast_format_set(&s->fr.subclass.format, AST_FORMAT_GSM, 0);
s->fr.offset = AST_FRIENDLY_OFFSET;
s->fr.samples = GSM_SAMPLES;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, GSM_FRAME_SIZE);
if (fs->secondhalf) {
/* Just return a frame based on the second GSM frame */
s->fr.data.ptr = (char *)s->fr.data.ptr + GSM_FRAME_SIZE;
s->fr.offset += GSM_FRAME_SIZE;
} else {
/* read and convert */
unsigned char msdata[MSGSM_FRAME_SIZE];
int res;
if ((res = fread(msdata, 1, MSGSM_FRAME_SIZE, s->f)) != MSGSM_FRAME_SIZE) {
if (res && (res != 1))
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
/* Convert from MS format to two real GSM frames */
conv65(msdata, s->fr.data.ptr);
}
fs->secondhalf = !fs->secondhalf;
*whennext = GSM_SAMPLES;
return &s->fr;
}
static int fax_generator_generate(struct ast_channel *chan, void *data, int len, int samples)
{
fax_state_t *fax = (fax_state_t*) data;
uint8_t buffer[AST_FRIENDLY_OFFSET + MAX_SAMPLES * sizeof(uint16_t)];
int16_t *buf = (int16_t *) (buffer + AST_FRIENDLY_OFFSET);
struct ast_frame outf = {
.frametype = AST_FRAME_VOICE,
.src = __FUNCTION__,
};
ast_format_set(&outf.subclass.format, AST_FORMAT_SLINEAR, 0);
if (samples > MAX_SAMPLES) {
ast_log(LOG_WARNING, "Only generating %d samples, where %d requested\n", MAX_SAMPLES, samples);
samples = MAX_SAMPLES;
}
if ((len = fax_tx(fax, buf, samples)) > 0) {
outf.samples = len;
AST_FRAME_SET_BUFFER(&outf, buffer, AST_FRIENDLY_OFFSET, len * sizeof(int16_t));
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", ast_channel_name(chan), strerror(errno));
return -1;
}
}
return 0;
}
static struct ast_frame *ilbc_read(struct ast_filestream *s, int *whennext)
{
int res;
/* Send a frame from the file to the appropriate channel */
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, ILBC_BUF_SIZE);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = ILBC_SAMPLES;
return &s->fr;
}
static struct ast_frame *generic_read(struct ast_filestream *s, int *whennext, unsigned int buf_size)
{
int res;
/* Send a frame from the file to the appropriate channel */
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, buf_size);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) < 1) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = res/2;
s->fr.datalen = res;
return &s->fr;
}
static struct ast_frame *gsm_read(struct ast_filestream *s, int *whennext)
{
int res;
s->fr.frametype = AST_FRAME_VOICE;
ast_format_set(&s->fr.subclass.format, AST_FORMAT_GSM, 0);
AST_FRAME_SET_BUFFER(&(s->fr), s->buf, AST_FRIENDLY_OFFSET, GSM_FRAME_SIZE)
s->fr.mallocd = 0;
if ((res = fread(s->fr.data.ptr, 1, GSM_FRAME_SIZE, s->f)) != GSM_FRAME_SIZE) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = GSM_SAMPLES;
return &s->fr;
}
static struct ast_frame *g726_read(struct ast_filestream *s, int *whennext)
{
int res;
struct g726_desc *fs = (struct g726_desc *)s->_private;
/* Send a frame from the file to the appropriate channel */
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, frame_size[fs->rate]);
s->fr.samples = 8 * FRAME_TIME;
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples;
return &s->fr;
}
/*!
* \brief Read a frame full of audio data from the filestream.
* \param fs The filestream.
* \param whennext Number of sample times to schedule the next call.
* \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
*/
static struct ast_frame *ogg_speex_read(struct ast_filestream *fs,
int *whennext)
{
struct speex_desc *s = (struct speex_desc *)fs->_private;
if (read_packet(fs) < 0) {
return NULL;
}
AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
memcpy(fs->fr.data.ptr, s->op.packet, s->op.bytes);
fs->fr.datalen = s->op.bytes;
fs->fr.samples = *whennext = ast_codec_samples_count(&fs->fr);
return &fs->fr;
}
static struct ast_frame *mp3_read(struct ast_filestream *s, int *whennext)
{
struct mp3_private *p = s->_private;
int delay =0;
int save=0;
/* Pre-populate the buffer that holds audio to be returned (dbuf) */
if (mp3_queue(s)) {
return NULL;
}
if (p->dbuflen) {
/* Read out what's waiting in dbuf */
for (p->buflen = 0; p->buflen < MP3_BUFLEN && p->buflen < p->dbuflen; p->buflen++) {
s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[p->buflen + p->dbufoffset];
}
p->dbufoffset += p->buflen;
p->dbuflen -= p->buflen;
}
if (p->buflen < MP3_BUFLEN) {
/* dbuf didn't have enough, so reset dbuf, fill it back up and continue */
p->dbuflen = p->dbufoffset = 0;
if (mp3_queue(s)) {
return NULL;
}
/* Make sure dbuf has enough to complete this read attempt */
if (p->dbuflen >= (MP3_BUFLEN - p->buflen)) {
for (save = p->buflen; p->buflen < MP3_BUFLEN; p->buflen++) {
s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[(p->buflen - save) + p->dbufoffset];
}
p->dbufoffset += (MP3_BUFLEN - save);
p->dbuflen -= (MP3_BUFLEN - save);
}
}
p->offset += p->buflen;
delay = p->buflen / 2;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, p->buflen);
s->fr.samples = delay;
*whennext = delay;
return &s->fr;
}
static struct ast_frame *g719read(struct ast_filestream *s, int *whennext)
{
size_t res;
/* Send a frame from the file to the appropriate channel */
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
if (res) {
ast_log(LOG_WARNING, "Short read of %s data (expected %d bytes, read %zu): %s\n",
ast_format_get_name(s->fr.subclass.format), s->fr.datalen, res,
strerror(errno));
}
return NULL;
}
*whennext = s->fr.samples = BYTES_TO_SAMPLES(res);
return &s->fr;
}
static struct ast_frame *siren14read(struct ast_filestream *s, int *whennext)
{
int res;
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass.codec = AST_FORMAT_SIREN14;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = BYTES_TO_SAMPLES(res);
return &s->fr;
}
static struct ast_frame *g729_read(struct ast_filestream *s, int *whennext)
{
int res;
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_G729A;
s->fr.mallocd = 0;
s->fr.samples = G729A_SAMPLES;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
if (res && (res != 10)) /* XXX what for ? */
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples;
return &s->fr;
}
static struct ast_frame *mp3_read(struct ast_filestream *s, int *whennext)
{
struct mp3_private *p = s->_private;
int delay =0;
int save=0;
/* Send a frame from the file to the appropriate channel */
if(mp3_queue(s))
return NULL;
if(p->dbuflen) {
for(p->buflen=0; p->buflen < MP3_BUFLEN && p->buflen < p->dbuflen; p->buflen++) {
s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[p->buflen+p->dbufoffset];
p->sbufoffset++;
}
p->dbufoffset += p->buflen;
p->dbuflen -= p->buflen;
if(p->buflen < MP3_BUFLEN) {
if(mp3_queue(s))
return NULL;
for(save = p->buflen; p->buflen < MP3_BUFLEN; p->buflen++) {
s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[(p->buflen-save)+p->dbufoffset];
p->sbufoffset++;
}
p->dbufoffset += (MP3_BUFLEN - save);
p->dbuflen -= (MP3_BUFLEN - save);
}
}
p->offset += p->buflen;
delay = p->buflen/2;
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_SLINEAR;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, p->buflen);
s->fr.mallocd = 0;
s->fr.samples = delay;
*whennext = delay;
return &s->fr;
}
static struct ast_frame *slinear_read(struct ast_filestream *s, int *whennext)
{
int res;
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_SLINEAR;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = res/2;
s->fr.datalen = res;
return &s->fr;
}
static struct ast_frame *generic_read(struct ast_filestream *s, int *whennext, unsigned int buf_size, enum ast_format_id id)
{
int res;
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
ast_format_set(&s->fr.subclass.format, id, 0);
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, buf_size);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) < 1) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = res/2;
s->fr.datalen = res;
return &s->fr;
}
/*!
* \brief Read a frame full of audio data from the filestream.
* \param fs The filestream.
* \param whennext Number of sample times to schedule the next call.
* \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
*/
static struct ast_frame *ogg_vorbis_read(struct ast_filestream *fs,
int *whennext)
{
struct ogg_vorbis_desc *desc = (struct ogg_vorbis_desc *) fs->_private;
int current_bitstream = -10;
char *out_buf;
long bytes_read;
if (desc->writing) {
ast_log(LOG_WARNING, "Reading is not supported on OGG/Vorbis on write files.\n");
return NULL;
}
/* initialize frame */
fs->fr.frametype = AST_FRAME_VOICE;
ast_format_set(&fs->fr.subclass.format, AST_FORMAT_SLINEAR, 0);
fs->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
out_buf = (char *) (fs->fr.data.ptr); /* SLIN data buffer */
/* read samples from OV interface */
bytes_read = ov_read(
&desc->ov_f,
out_buf, /* Buffer to write data */
BUF_SIZE, /* Size of buffer */
(__BYTE_ORDER == __BIG_ENDIAN), /* Endianes (0 for little) */
2, /* 1 = 8bit, 2 = 16bit */
1, /* 0 = unsigned, 1 = signed */
¤t_bitstream /* Returns the current bitstream section */
);
/* check returned data */
if (bytes_read <= 0) {
/* End of stream */
return NULL;
}
/* Return decoded bytes */
fs->fr.datalen = bytes_read;
fs->fr.samples = bytes_read / 2;
*whennext = fs->fr.samples;
return &fs->fr;
}
static struct ast_frame *g726_read(struct ast_filestream *s, int *whennext)
{
size_t res;
struct g726_desc *fs = (struct g726_desc *)s->_private;
/* Send a frame from the file to the appropriate channel */
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, frame_size[fs->rate]);
s->fr.samples = 8 * FRAME_TIME;
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
if (res) {
ast_log(LOG_WARNING, "Short read of %s data (expected %d bytes, read %zu): %s\n",
ast_format_get_name(s->fr.subclass.format), s->fr.datalen, res,
strerror(errno));
}
return NULL;
}
*whennext = s->fr.samples;
return &s->fr;
}
static struct ast_frame *h263_read(struct ast_filestream *s, int *whennext)
{
int res;
uint32_t mark;
unsigned short len;
unsigned int ts;
struct h263_desc *fs = (struct h263_desc *)s->_private;
/* Send a frame from the file to the appropriate channel */
if ((res = fread(&len, 1, sizeof(len), s->f)) < 1)
return NULL;
len = ntohs(len);
mark = (len & 0x8000) ? 1 : 0;
len &= 0x7fff;
if (len > BUF_SIZE) {
ast_log(LOG_WARNING, "Length %d is too long\n", len);
return NULL;
}
s->fr.frametype = AST_FRAME_VIDEO;
ast_format_set(&s->fr.subclass.format, AST_FORMAT_H263, 0);
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, len);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
s->fr.samples = fs->lastts; /* XXX what ? */
s->fr.datalen = len;
if (mark) {
ast_format_set_video_mark(&s->fr.subclass.format);
}
s->fr.delivery.tv_sec = 0;
s->fr.delivery.tv_usec = 0;
if ((res = fread(&ts, 1, sizeof(ts), s->f)) == sizeof(ts)) {
fs->lastts = ntohl(ts);
*whennext = fs->lastts * 4/45;
} else
*whennext = 0;
return &s->fr;
}
static int t38_tx_packet_handler(t38_core_state_t *s, void *user_data, const uint8_t *buf, int len, int count)
{
struct ast_channel *chan = (struct ast_channel *) user_data;
struct ast_frame outf = {
.frametype = AST_FRAME_MODEM,
.subclass = AST_MODEM_T38,
.src = __FUNCTION__,
};
/* TODO: Asterisk does not provide means of resending the same packet multiple
times so count is ignored at the moment */
AST_FRAME_SET_BUFFER(&outf, buf, 0, len);
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno));
return -1;
}
return 0;
}
static struct ast_frame *pcm_read(struct ast_filestream *s, int *whennext)
{
int res;
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
ast_format_copy(&s->fr.subclass.format, &s->fmt->format);
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) < 1) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
s->fr.datalen = res;
if (s->fmt->format.id == AST_FORMAT_G722)
*whennext = s->fr.samples = res * 2;
else
*whennext = s->fr.samples = res;
return &s->fr;
}
static int t38_tx_packet_handler(t38_core_state_t *s, void *user_data, const uint8_t *buf, int len, int count)
{
struct ast_channel *chan = (struct ast_channel *) user_data;
struct ast_frame outf = {
.frametype = AST_FRAME_MODEM,
.subclass.integer = AST_MODEM_T38,
.src = __FUNCTION__,
};
/* TODO: Asterisk does not provide means of resending the same packet multiple
times so count is ignored at the moment */
AST_FRAME_SET_BUFFER(&outf, buf, 0, len);
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno));
return -1;
}
return 0;
}
static void phase_e_handler(t30_state_t *f, void *user_data, int result)
{
RAII_VAR(struct ast_json *, json_object, NULL, ast_json_unref);
RAII_VAR(struct ast_json *, json_filenames, NULL, ast_json_unref);
RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
const char *local_ident;
const char *far_ident;
char buf[20];
fax_session *s = (fax_session *) user_data;
t30_stats_t stat;
int pages_transferred;
ast_debug(1, "Fax phase E handler. result=%d\n", result);
t30_get_transfer_statistics(f, &stat);
s = (fax_session *) user_data;
if (result != T30_ERR_OK) {
s->finished = -1;
/* FAXSTATUS is already set to FAILED */
pbx_builtin_setvar_helper(s->chan, "FAXERROR", t30_completion_code_to_str(result));
ast_log(LOG_WARNING, "Error transmitting fax. result=%d: %s.\n", result, t30_completion_code_to_str(result));
return;
}
s->finished = 1;
local_ident = S_OR(t30_get_tx_ident(f), "");
far_ident = S_OR(t30_get_rx_ident(f), "");
pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "SUCCESS");
pbx_builtin_setvar_helper(s->chan, "FAXERROR", NULL);
pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", far_ident);
#if SPANDSP_RELEASE_DATE >= 20090220
pages_transferred = (s->direction) ? stat.pages_tx : stat.pages_rx;
#else
pages_transferred = stat.pages_transferred;
#endif
snprintf(buf, sizeof(buf), "%d", pages_transferred);
pbx_builtin_setvar_helper(s->chan, "FAXPAGES", buf);
snprintf(buf, sizeof(buf), "%d", stat.y_resolution);
pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", buf);
snprintf(buf, sizeof(buf), "%d", stat.bit_rate);
pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", buf);
ast_debug(1, "Fax transmitted successfully.\n");
ast_debug(1, " Remote station ID: %s\n", far_ident);
ast_debug(1, " Pages transferred: %d\n", pages_transferred);
ast_debug(1, " Image resolution: %d x %d\n", stat.x_resolution, stat.y_resolution);
ast_debug(1, " Transfer Rate: %d\n", stat.bit_rate);
json_filenames = ast_json_pack("[s]", s->file_name);
if (!json_filenames) {
return;
}
ast_json_ref(json_filenames);
json_object = ast_json_pack("{s: s, s: s, s: s, s: i, s: i, s: i, s: o}",
"type", s->direction ? "send" : "receive",
"remote_station_id", far_ident,
"local_station_id", local_ident,
"fax_pages", pages_transferred,
"fax_resolution", stat.y_resolution,
"fax_bitrate", stat.bit_rate,
"filenames", json_filenames);
message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(s->chan), ast_channel_fax_type(), json_object);
if (!message) {
return;
}
stasis_publish(ast_channel_topic(s->chan), message);
}
static int t38_tx_packet_handler(t38_core_state_t *s, void *user_data, const uint8_t *buf, int len, int count)
{
struct ast_channel *chan = (struct ast_channel *) user_data;
struct ast_frame outf = {
.frametype = AST_FRAME_MODEM,
.subclass.integer = AST_MODEM_T38,
.src = __FUNCTION__,
};
/* TODO: Asterisk does not provide means of resending the same packet multiple
times so count is ignored at the moment */
AST_FRAME_SET_BUFFER(&outf, buf, 0, len);
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno));
return -1;
}
return 0;
}
static void phase_e_handler(t30_state_t *f, void *user_data, int result)
{
const char *local_ident;
const char *far_ident;
char buf[20];
fax_session *s = (fax_session *) user_data;
t30_stats_t stat;
int pages_transferred;
ast_debug(1, "Fax phase E handler. result=%d\n", result);
t30_get_transfer_statistics(f, &stat);
s = (fax_session *) user_data;
if (result != T30_ERR_OK) {
s->finished = -1;
/* FAXSTATUS is already set to FAILED */
pbx_builtin_setvar_helper(s->chan, "FAXERROR", t30_completion_code_to_str(result));
ast_log(LOG_WARNING, "Error transmitting fax. result=%d: %s.\n", result, t30_completion_code_to_str(result));
return;
}
s->finished = 1;
local_ident = S_OR(t30_get_tx_ident(f), "");
far_ident = S_OR(t30_get_rx_ident(f), "");
pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "SUCCESS");
pbx_builtin_setvar_helper(s->chan, "FAXERROR", NULL);
pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", far_ident);
#if SPANDSP_RELEASE_DATE >= 20090220
pages_transferred = (s->direction) ? stat.pages_tx : stat.pages_rx;
#else
pages_transferred = stat.pages_transferred;
#endif
snprintf(buf, sizeof(buf), "%d", pages_transferred);
pbx_builtin_setvar_helper(s->chan, "FAXPAGES", buf);
snprintf(buf, sizeof(buf), "%d", stat.y_resolution);
pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", buf);
snprintf(buf, sizeof(buf), "%d", stat.bit_rate);
pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", buf);
ast_debug(1, "Fax transmitted successfully.\n");
ast_debug(1, " Remote station ID: %s\n", far_ident);
ast_debug(1, " Pages transferred: %d\n", pages_transferred);
ast_debug(1, " Image resolution: %d x %d\n", stat.x_resolution, stat.y_resolution);
ast_debug(1, " Transfer Rate: %d\n", stat.bit_rate);
ast_manager_event(s->chan, EVENT_FLAG_CALL,
s->direction ? "FaxSent" : "FaxReceived",
"Channel: %s\r\n"
"Exten: %s\r\n"
"CallerID: %s\r\n"
"CallerIDName: %s\r\n"
"ConnectedLineNum: %s\r\n"
"ConnectedLineName: %s\r\n"
"RemoteStationID: %s\r\n"
"LocalStationID: %s\r\n"
"PagesTransferred: %d\r\n"
"Resolution: %d\r\n"
"TransferRate: %d\r\n"
"FileName: %s\r\n",
ast_channel_name(s->chan),
ast_channel_exten(s->chan),
S_COR(ast_channel_caller(s->chan)->id.number.valid, ast_channel_caller(s->chan)->id.number.str, ""),
S_COR(ast_channel_caller(s->chan)->id.name.valid, ast_channel_caller(s->chan)->id.name.str, ""),
S_COR(ast_channel_connected(s->chan)->id.number.valid, ast_channel_connected(s->chan)->id.number.str, ""),
S_COR(ast_channel_connected(s->chan)->id.name.valid, ast_channel_connected(s->chan)->id.name.str, ""),
far_ident,
local_ident,
pages_transferred,
stat.y_resolution,
stat.bit_rate,
s->file_name);
}
/* === Helper functions to configure fax === */
/* Setup SPAN logging according to Asterisk debug level */
static int set_logging(logging_state_t *state)
{
int level = SPAN_LOG_WARNING + option_debug;
span_log_set_message_handler(state, span_message);
span_log_set_level(state, SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | level);
return 0;
}
/*!
* \brief Read a frame full of audio data from the filestream.
* \param fs The filestream.
* \param whennext Number of sample times to schedule the next call.
* \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
*/
static struct ast_frame *ogg_vorbis_read(struct ast_filestream *fs,
int *whennext)
{
int clipflag = 0;
int i;
int j;
double accumulator[SAMPLES_MAX];
int val;
int samples_in;
int samples_out = 0;
struct vorbis_desc *s = (struct vorbis_desc *)fs->_private;
short *buf; /* SLIN data buffer */
fs->fr.frametype = AST_FRAME_VOICE;
fs->fr.subclass = AST_FORMAT_SLINEAR;
fs->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
buf = (short *)(fs->fr.data); /* SLIN data buffer */
while (samples_out != SAMPLES_MAX) {
float **pcm;
int len = SAMPLES_MAX - samples_out;
/* See ifVorbis decoder has some audio data for us ... */
samples_in = read_samples(fs, &pcm);
if (samples_in <= 0)
break;
/* Got some audio data from Vorbis... */
/* Convert the float audio data to 16-bit signed linear */
clipflag = 0;
if (samples_in > len)
samples_in = len;
for (j = 0; j < samples_in; j++)
accumulator[j] = 0.0;
for (i = 0; i < s->vi.channels; i++) {
float *mono = pcm[i];
for (j = 0; j < samples_in; j++)
accumulator[j] += mono[j];
}
for (j = 0; j < samples_in; j++) {
val = accumulator[j] * 32767.0 / s->vi.channels;
if (val > 32767) {
val = 32767;
clipflag = 1;
} else if (val < -32768) {
val = -32768;
clipflag = 1;
}
buf[samples_out + j] = val;
}
if (clipflag)
ast_log(LOG_WARNING, "Clipping in frame %ld\n", (long) (s->vd.sequence));
/* Tell the Vorbis decoder how many samples we actually used. */
vorbis_synthesis_read(&s->vd, samples_in);
samples_out += samples_in;
}
if (samples_out > 0) {
fs->fr.datalen = samples_out * 2;
fs->fr.samples = samples_out;
*whennext = samples_out;
return &fs->fr;
} else {
return NULL;
}
}
