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|
/*
* FILE: transmit.c
* PROGRAM: RAT
* AUTHOR: Orion Hodson / Isidor Kouvelas
* MODIFIED: Colin Perkins
*
* Copyright (c) 1995-2001 University College London
* All rights reserved.
*/
#ifndef HIDE_SOURCE_STRINGS
static const char cvsid[] =
"$Id: transmit.c,v 1.131 2002/03/15 17:57:19 ucacoxh Exp $";
#endif /* HIDE_SOURCE_STRINGS */
#include "config_unix.h"
#include "config_win32.h"
#include "memory.h"
#include "debug.h"
#include "audio_types.h"
#include "codec_types.h"
#include "codec.h"
#include "codec_state.h"
#include "playout.h"
#include "channel_types.h"
#include "channel.h"
#include "session.h"
#include "audio.h"
#include "audio_util.h"
#include "sndfile.h"
#include "converter.h"
#include "parameters.h"
#include "pdb.h"
#include "ui_send_rtp.h"
#include "ui_send_audio.h"
#include "ui_send_prefs.h"
#include "rtp.h"
#include "transmit.h"
#include "util.h"
#define TX_BUFFER_MAGIC 0x12345678
/* All this code can be greatly simplified and reduced by making
* better use of the playout buffer structure in playout.h.
*/
typedef struct s_tx_unit {
sample *data; /* pointer to raw data in read_buf */
uint32_t dur_used; /* number of time intervals filled */
uint16_t energy; /* */
char silence; /* First pass */
u_char send; /* Silence second pass */
u_char encoded; /* TRUE if this unit has been encoded for transmission */
u_char live_capture; /* TRUE if this unit came from a microphone, FALSE if it came from a file */
} tx_unit;
typedef struct s_tx_buffer {
struct s_session *sp;
struct s_vad *vad;
struct s_agc *agc;
struct s_bias_ctl *bc;
struct s_pb *media_buffer;
struct s_pb *channel_buffer;
struct s_pb *audio_buffer; /* Audio buffer and it's iterators... */
struct s_pb_iterator *reading; /* ...current read point iterator */
struct s_pb_iterator *silence; /* ...silence classification iterator */
struct s_pb_iterator *transmit; /* ...transmission point iterator */
struct s_codec_state_store *state_store; /* Encoder states */
uint32_t sending_audio:1;
uint32_t sample_rate;
uint16_t channels;
uint16_t unit_dur; /* dur. in sampling intervals (excludes channels) */
/* Statistics log */
double mean_read_dur;
/* These are a hack because we use playout buffer
* which expects time units of type timestamp_t so we need
* to be able to map to and from 32 bit no for
* packet timestamp */
ts_sequencer down_seq; /* used for 32 -> timestamp_t */
ts_sequencer up_seq; /* used for timestamp_t -> 32 */
/* place for the samples */
sample samples[DEVICE_REC_BUF];
int last_sample; /* Stores the index of the last read buffer */
/* bandwidth estimate parameters */
int bps_bytes_sent;
timestamp_t bps_last_update;
/* For debugging... */
uint32_t magic;
} tx_buffer;
static sample dummy_buf[DEVICE_REC_BUF];
static void tx_read_sndfile(session_t *sp, uint16_t tx_freq, uint16_t tx_channels, tx_unit *u);
static void
tx_buffer_validate(tx_buffer *tb)
{
#ifdef DEBUG
session_validate(tb->sp);
if (tb->sending_audio) {
assert(pb_iterator_count(tb->audio_buffer) == 3);
} else {
assert(pb_iterator_count(tb->audio_buffer) == 0);
}
assert(tb->channels > 0 && tb->channels <= 2);
assert(tb->last_sample < DEVICE_REC_BUF);
#endif
assert(tb->magic == TX_BUFFER_MAGIC);
}
static int
tx_unit_create(tx_buffer *tb, tx_unit **ptu, int n_samples)
{
tx_unit *tu;
tx_buffer_validate(tb);
tu = xmalloc(sizeof(tx_unit));
if (tu) {
*ptu = tu;
/* Position sample pointer */
if (tb->last_sample + n_samples >= DEVICE_REC_BUF) {
tb->last_sample = 0;
}
tu->data = tb->samples + tb->last_sample;
tu->dur_used = 0;
tu->energy = 555;
tu->silence = -1; /* -1 == not yet performed silence detection */
tu->send = 0;
tu->encoded = FALSE;
tu->live_capture = TRUE;
tb->last_sample += n_samples;
tx_buffer_validate(tb);
return TRUE;
}
debug_msg("Failed to allocate tx_unit\n");
return FALSE;
}
static void
tx_unit_destroy(tx_unit **ptu, uint32_t len)
{
tx_unit *tu = *ptu;
assert(tu != NULL);
assert(len == sizeof(tx_unit));
xfree(tu);
*ptu = NULL;
}
int
tx_create(tx_buffer **ntb,
session_t *sp,
uint32_t sample_rate,
uint16_t channels,
uint16_t unit_dur)
{
tx_buffer *tb;
tb = (tx_buffer*)xmalloc(sizeof(tx_buffer));
if (tb) {
memset(tb, 0, sizeof(tx_buffer));
debug_msg("Unit duration %d channels %d\n", unit_dur, channels);
tb->magic = TX_BUFFER_MAGIC;
tb->sp = sp;
tb->sending_audio = FALSE;
tb->sample_rate = sample_rate;
tb->channels = channels;
tb->unit_dur = unit_dur;
tb->mean_read_dur = unit_dur;
tb->bc = bias_ctl_create(channels, sample_rate);
tb->vad = vad_create(unit_dur, sample_rate);
tb->agc = agc_create(sp);
sp->auto_sd = sd_init(unit_dur, sample_rate);
sp->manual_sd = manual_sd_init(unit_dur, sample_rate, sp->manual_sd_thresh);
pb_create(&tb->audio_buffer, (playoutfreeproc)tx_unit_destroy);
pb_create(&tb->media_buffer, (playoutfreeproc)media_data_destroy);
pb_create(&tb->channel_buffer, (playoutfreeproc)channel_data_destroy);
tx_buffer_validate(tb);
*ntb = tb;
return TRUE;
}
return FALSE;
}
void
tx_destroy(tx_buffer **ptb)
{
tx_buffer *tb;
assert(ptb != NULL);
tb = *ptb;
assert(tb != NULL);
tx_buffer_validate(tb);
bias_ctl_destroy(tb->bc);
sd_destroy(tb->sp->auto_sd);
manual_sd_destroy(tb->sp->manual_sd);
vad_destroy(tb->vad);
agc_destroy(tb->agc);
pb_destroy(&tb->audio_buffer);
pb_destroy(&tb->media_buffer);
pb_destroy(&tb->channel_buffer);
xfree(tb);
*ptb = NULL;
}
/* These routines are called when the button on the interface is toggled */
void
tx_start(tx_buffer *tb)
{
tx_unit *tu_new;
timestamp_t unit_start;
debug_msg("Starting to transmit...\n");
tx_buffer_validate(tb);
if (tb->sending_audio) {
debug_msg("Already sending... why? Fix me!");
abort();
}
tb->sending_audio = TRUE;
/* Turn off auto lecture */
tb->sp->auto_lecture = 1;
/* Reset signal classification and auto-scaling */
sd_reset(tb->sp->auto_sd);
vad_reset(tb->vad);
agc_reset(tb->agc);
/* Attach iterator for silence classification */
pb_iterator_create(tb->audio_buffer, &tb->transmit);
pb_iterator_create(tb->audio_buffer, &tb->silence);
pb_iterator_create(tb->audio_buffer, &tb->reading);
assert(pb_iterator_count(tb->audio_buffer) == 3);
/* Add one unit to media buffer to kick off audio reading */
unit_start = tb->sp->cur_ts;
tx_unit_create(tb, &tu_new, tb->unit_dur * tb->channels);
assert(ts_valid(unit_start));
pb_add(tb->audio_buffer, (u_char*)tu_new, sizeof(tx_unit), unit_start);
/* And then put reading iterator on it */
pb_iterator_advance(tb->reading);
assert(tb->state_store == NULL);
codec_state_store_create(&tb->state_store, ENCODER);
tx_update_ui(tb);
tb->bps_last_update = tb->sp->cur_ts;
}
void
tx_stop(tx_buffer *tb)
{
struct timeval tv;
tx_buffer_validate(tb);
if (tb->sending_audio == FALSE) {
return;
}
gettimeofday(&tv, NULL);
tb->sp->auto_lecture = tv.tv_sec;
codec_state_store_destroy(&tb->state_store);
channel_encoder_reset(tb->sp->channel_coder);
ui_send_audio_input_powermeter(tb->sp, tb->sp->mbus_ui_addr, 0);
tb->sending_audio = FALSE;
/* Detach iterators */
assert(pb_iterator_count(tb->audio_buffer) == 3);
pb_iterator_destroy(tb->audio_buffer, &tb->transmit);
pb_iterator_destroy(tb->audio_buffer, &tb->silence);
pb_iterator_destroy(tb->audio_buffer, &tb->reading);
assert(pb_iterator_count(tb->audio_buffer) == 0);
/* Drain playout buffers */
pb_flush(tb->audio_buffer);
pb_flush(tb->media_buffer);
pb_flush(tb->channel_buffer);
tb->bps_bytes_sent = 0;
tx_buffer_validate(tb);
tx_update_ui(tb);
}
int
tx_read_audio(tx_buffer *tb)
{
session_t *sp;
tx_unit *u;
timestamp_t u_ts;
uint32_t read_dur = 0, this_read, ulen;
tx_buffer_validate(tb);
sp = tb->sp;
if (tb->sending_audio) {
int filled_unit;
assert(pb_iterator_count(tb->audio_buffer) == 3);
do {
if (pb_iterator_get_at(tb->reading, (u_char**)&u, &ulen, &u_ts) == FALSE) {
debug_msg("Reading iterator failed to get unit!\n");
}
assert(u != NULL);
this_read = audio_read(sp->audio_device,
u->data + u->dur_used * tb->channels,
(tb->unit_dur - u->dur_used) * tb->channels) / tb->channels;
assert(this_read <= tb->unit_dur - u->dur_used);
filled_unit = FALSE;
u->dur_used += this_read;
if (u->dur_used == tb->unit_dur) {
read_dur += tb->unit_dur;
if (sp->in_file) {
/* Reading from a file overwrites any audio we've captured... */
tx_read_sndfile(sp, tb->sample_rate, tb->channels, u);
}
sp->cur_ts = ts_add(sp->cur_ts, ts_map32(tb->sample_rate, tb->unit_dur));
u_ts = sp->cur_ts;
filled_unit = TRUE;
/* We've filled one unit, so create the next one... */
tx_unit_create(tb, &u, tb->unit_dur * tb->channels);
pb_add(tb->audio_buffer, (u_char*)u, ulen, u_ts);
pb_iterator_advance(tb->reading);
}
} while (filled_unit == TRUE);
assert(pb_iterator_count(tb->audio_buffer) == 3);
} else {
int this_read = 0;
/* We're not sending, but have access to the audio device.
* Read the audio anyway to get exact values, and then
* throw the data we've just read away...
*/
do {
this_read = audio_read(sp->audio_device, dummy_buf, DEVICE_REC_BUF / 4) / sp->tb->channels;
read_dur += this_read;
} while (this_read > 0);
sp->cur_ts = ts_add(sp->cur_ts, ts_map32(tb->sample_rate, read_dur));
}
if (read_dur >= (uint32_t)(DEVICE_REC_BUF / (4 * tb->channels))) {
debug_msg("Read a lot of audio %d\n", read_dur);
if (tb->sending_audio) {
debug_msg("Resetting transmitter\n");
tx_stop(tb);
tx_start(tb);
}
}
if (read_dur) {
sp->tb->mean_read_dur += ((double)read_dur - sp->tb->mean_read_dur) / 8.0;
}
assert(read_dur < 0x7fffffff);
return read_dur;
}
int
tx_process_audio(tx_buffer *tb)
{
struct s_pb_iterator *marker;
tx_unit *u;
uint32_t u_len;
timestamp_t u_ts;
int to_send;
tx_buffer_validate(tb);
assert(tb->sending_audio);
/* Do signal classification up until read point, that
* is not a complete audio frame so cannot be done
*/
assert(pb_iterator_count(tb->audio_buffer) == 3);
pb_iterator_get_at(tb->silence, (u_char**)&u, &u_len, &u_ts);
while (pb_iterators_equal(tb->silence, tb->reading) == FALSE) {
assert(u->dur_used == tb->unit_dur);
if (u->live_capture) {
bias_remove(tb->bc, u->data, u->dur_used * tb->channels);
} else {
debug_msg("Unit came from a file, no need for bias removal\n");
}
u->energy = audio_avg_energy(u->data, u->dur_used * tb->channels, tb->channels);
u->send = FALSE;
/* Silence classification on this block */
assert(u->silence == -1); /* We should only do this once per block... */
switch(tb->sp->silence_detection) {
case SILENCE_DETECTION_AUTO:
u->silence = sd(tb->sp->auto_sd, (uint16_t)u->energy);
break;
case SILENCE_DETECTION_MANUAL:
u->silence = manual_sd(tb->sp->manual_sd,
(uint16_t)u->energy,
audio_abs_max(u->data, u->dur_used * tb->channels));
break;
case SILENCE_DETECTION_OFF:
u->silence = 0;
break;
}
assert((u->silence == 0) || (u->silence == 1));
/* Pass decision to voice activity detector (damps transients, etc) */
to_send = vad_to_get(tb->vad,
(u_char)u->silence,
(u_char)((tb->sp->lecture) ? VAD_MODE_LECT : VAD_MODE_CONF));
agc_update(tb->agc, (uint16_t)u->energy, vad_talkspurt_no(tb->vad));
if (tb->sp->silence_detection != SILENCE_DETECTION_OFF) {
if (to_send != 0) {
pb_iterator_dup(&marker, tb->silence);
while(u != NULL && to_send != 0) {
u->send = TRUE;
to_send --;
pb_iterator_retreat(marker);
pb_iterator_get_at(marker, (u_char**)&u, &u_len, &u_ts);
}
pb_iterator_destroy(tb->audio_buffer, &marker);
}
assert(pb_iterator_count(tb->audio_buffer) == 3);
} else {
u->silence = FALSE;
u->send = TRUE;
}
pb_iterator_advance(tb->silence);
pb_iterator_get_at(tb->silence, (u_char**)&u, &u_len, &u_ts);
}
if (tb->sp->agc_on == TRUE && agc_apply_changes(tb->agc) == TRUE) {
ui_send_audio_input_gain(tb->sp, tb->sp->mbus_ui_addr);
}
tx_buffer_validate(tb);
return TRUE;
}
static int
tx_encode(struct s_codec_state_store *css,
sample *buf,
uint32_t dur_used,
uint32_t encoding,
u_char *payloads,
coded_unit **coded)
{
codec_id_t id;
uint32_t i;
id = codec_get_by_payload(payloads[encoding]);
assert(id);
/* Look to see if we have already coded this unit,
* i.e. we are using redundancy. Don't want to code
* twice since it screws up encoder state.
*/
for (i = 0; i < encoding; i++) {
if (coded[i]->id == id) {
break;
}
}
if (i == encoding) {
const codec_format_t *cf;
coded_unit native;
codec_state *cs;
/* Unit does not exist already */
cf = codec_get_format(id);
/* native is a temporary coded_unit that we use to pass to
* codec_encode since this take a 'native' (raw) coded unit as
* input and fills in coded with the transformed data.
*/
native.id = codec_get_native_coding((uint32_t)cf->format.sample_rate,
(uint16_t)cf->format.channels);
native.state = NULL;
native.state_len = 0;
native.data = (u_char*)buf;
native.data_len = (uint16_t)(dur_used * sizeof(sample) * cf->format.channels);
/* Get codec state from those stored for us */
cs = codec_state_store_get(css, id);
return codec_encode(cs, &native, coded[encoding]);
} else {
/* duplicate coded unit */
return coded_unit_dup(coded[encoding], coded[i]);
}
}
void
tx_send(tx_buffer *tb)
{
struct s_pb_iterator *cpos;
channel_data *cd;
channel_unit *cu;
tx_unit *u;
timestamp_t u_ts, u_sil_ts, delta;
timestamp_t time_ts;
uint32_t time_32, cd_len;
uint32_t u_len, units, i, j, k, n, send, encoding;
int success;
char *extn;
uint16_t extn_len, extn_type;
tx_buffer_validate(tb);
assert(pb_iterator_count(tb->audio_buffer) == 3);
if (pb_iterators_equal(tb->silence, tb->transmit)) {
return;
}
pb_iterator_get_at(tb->silence, (u_char**)&u, &u_len, &u_sil_ts);
pb_iterator_get_at(tb->transmit, (u_char**)&u, &u_len, &u_ts);
assert(ts_gt(u_sil_ts, u_ts));
delta = ts_sub(u_sil_ts, u_ts);
n = delta.ticks / tb->unit_dur;
units = channel_encoder_get_units_per_packet(tb->sp->channel_coder);
while(n >= units) {
/* We have accumulated at least enough audio to fill an RTP packet. */
/* The following code processes one packet's worth of audio, and if */
/* any of the units are marked `to send' then the entire packet is */
/* encoded and placed onto the transmission queue: tb->media_buffer */
send = FALSE;
for (i = 0; i < units; i++) {
pb_iterator_get_at(tb->transmit, (u_char**)&u, &u_len, &u_ts);
assert((u->silence == 0) || (u->silence == 1)); /* We MUST have done silence detection before transmit */
if (u->send) {
send = TRUE;
break;
}
pb_iterator_advance(tb->transmit);
}
/* Rewind transmit point to where it was before we did last check */
while(i > 0) {
pb_iterator_retreat(tb->transmit);
i--;
}
for (i = 0;i < units; i++) {
media_data *m;
success = pb_iterator_get_at(tb->transmit, (u_char**)&u, &u_len, &u_ts);
assert(success);
assert((u->silence == 0) || (u->silence == 1)); /* We MUST have done silence detection before transmit */
if (send) {
assert(u->encoded == FALSE);
media_data_create(&m, tb->sp->num_encodings);
for(encoding = 0; encoding < (uint32_t)tb->sp->num_encodings; encoding ++) {
tx_encode(tb->state_store, u->data, u->dur_used, encoding, tb->sp->encodings, m->rep);
}
/* At this point, the uncompressed data `u' is no longer needed since its */
/* coded form is in tb->media_buffer. The tb->transmit is audited later. */
/* We mark it as being encoded, as a debugging check to ensure that it is */
/* not accidently re-sent later. */
u->encoded = TRUE;
} else {
media_data_create(&m, 0);
}
assert(m != NULL);
success = pb_add(tb->media_buffer, (u_char*)m, sizeof(media_data), u_ts);
assert(success);
success = pb_iterator_advance(tb->transmit);
assert(success);
}
n -= units;
}
/* This does any necessary channel coding... The channel coder takes units from */
/* tb->media_buffer and moves then (eventually) onto tb->channel_buffer. They */
/* be delayed, reordered, aggregated, etc, in the process, but we don't worry */
/* about that here. tb->media_buffer is drained by the channel encoding stage. */
channel_encoder_encode(tb->sp->channel_coder, tb->media_buffer, tb->channel_buffer);
/* Pull units out of tb->channel_buffer and transmit them... */
pb_iterator_create(tb->channel_buffer, &cpos);
pb_iterator_advance(cpos);
while(pb_iterator_detach_at(cpos, (u_char**)&cd, &cd_len, &time_ts)) {
uint32_t csrc[16];
char *data, pt;
int data_len, done;
int marker;
/* Set up fields for RTP header */
cu = cd->elem[0];
pt = channel_coder_get_payload(tb->sp->channel_coder, cu->pt);
time_32 = ts_seq32_out(&tb->up_seq, tb->sample_rate, time_ts);
if (time_32 - tb->sp->last_depart_ts != units * tb->unit_dur) {
marker = 1;
debug_msg("new talkspurt (%d - %d != %d)\n", time_32, tb->sp->last_depart_ts, units * tb->unit_dur);
} else {
marker = 0;
}
/* layer loop starts here */
for(j = 0; j < (uint32_t)tb->sp->layers; j++) {
assert(tb->sp->layers == 1); /* FIXME */
assert(j == 0); /* FIXME */
#ifdef DEBUG_HEADER_EXTN
extn = (char *) xmalloc(5);
sprintf(extn, "test");
extn_len = 1; /* 32 bit words of extn data */
extn_type = 1;
#else
extn = NULL;
extn_len = 0;
extn_type = 0;
#endif
data_len = 0;
/* determine data length for packet. This is a */
/* little over complicated because of layering... */
for(i = j, k=0; i < cd->nelem; i += tb->sp->layers) {
data_len += (int) cd->elem[i]->data_len;
k++;
}
/* Copy all out going data into one block (no scatter) */
data = (char*)block_alloc(data_len);
done = 0;
for(i = j; i < cd->nelem; i += tb->sp->layers) {
memcpy(data + done, cd->elem[i]->data, cd->elem[i]->data_len);
done += cd->elem[i]->data_len;
}
rtp_send_data(tb->sp->rtp_session[j], time_32, pt, marker, 0, csrc, data, data_len, extn, extn_len, extn_type);
block_free(data, data_len);
tb->bps_bytes_sent += data_len;
if (extn != NULL) {
xfree(extn);
}
}
/* layer loop ends here */
tb->sp->last_depart_ts = time_32;
channel_data_destroy(&cd, sizeof(channel_data));
}
pb_iterator_destroy(tb->channel_buffer, &cpos);
/* Drain tb->audio, remove every older than silence position
* by two packets worth of audio. Note tb->media_buffer is drained
* by the channel encoding stage and tb->channel_buffer is drained
* in the act of transmission with pbi_detach_at call.
*/
u_ts = ts_map32(tb->sample_rate, 2 * units * tb->unit_dur);
{
struct s_pb *buf;
buf = pb_iterator_get_playout_buffer(tb->transmit);
assert(pb_iterator_count(buf) == 3);
}
assert(pb_iterator_count(tb->audio_buffer) == 3);
n = pb_iterator_audit(tb->transmit, u_ts);
}
void
tx_update_ui(tx_buffer *tb)
{
session_t *sp = tb->sp;
tx_buffer_validate(tb);
if (sp->meter && tb->sending_audio) {
struct s_pb_iterator *prev;
tx_unit *u;
uint32_t u_len;
timestamp_t u_ts;
/* Silence point should be upto read point here so use last
* completely read unit.
*/
assert(pb_iterator_count(tb->audio_buffer) == 3);
pb_iterator_dup(&prev, tb->silence);
pb_iterator_retreat(prev);
if (pb_iterators_equal(tb->silence, prev)) {
pb_iterator_destroy(tb->audio_buffer, &prev);
return;
}
if (pb_iterator_get_at(prev, (u_char**)&u, &u_len, &u_ts) &&
(vad_in_talkspurt(sp->tb->vad) == TRUE || sp->silence_detection == SILENCE_DETECTION_OFF)) {
ui_send_audio_input_powermeter(sp, sp->mbus_ui_addr, lin2vu(u->energy, 100, VU_INPUT));
} else {
ui_send_audio_input_powermeter(sp, sp->mbus_ui_addr, 0);
}
pb_iterator_destroy(tb->audio_buffer, &prev);
assert(pb_iterator_count(tb->audio_buffer) == 3);
}
/* This next routine is really inefficient - we only need do ui_info_activate() */
/* when the state changes, else we flood the mbus with redundant messages. */
if (sp->silence_detection != SILENCE_DETECTION_OFF) {
if (vad_in_talkspurt(sp->tb->vad) == TRUE) {
if (sp->ui_activated == FALSE) {
ui_send_rtp_active(sp, sp->mbus_ui_addr, rtp_my_ssrc(sp->rtp_session[0]));
sp->ui_activated = TRUE;
}
} else if (sp->ui_activated == TRUE) {
ui_send_rtp_inactive(sp, sp->mbus_ui_addr, rtp_my_ssrc(sp->rtp_session[0]));
sp->ui_activated = FALSE;
}
if (sp->lecture) {
sp->lecture = FALSE;
ui_send_lecture_mode(sp, sp->mbus_ui_addr);
}
} else if (sp->silence_detection == SILENCE_DETECTION_OFF) {
if (tb->sending_audio == TRUE && sp->ui_activated == FALSE) {
ui_send_rtp_active(sp, sp->mbus_ui_addr, rtp_my_ssrc(sp->rtp_session[0]));
sp->ui_activated = TRUE;
}
}
if (tb->sending_audio == FALSE && sp->ui_activated == TRUE) {
ui_send_rtp_inactive(sp, sp->mbus_ui_addr, rtp_my_ssrc(sp->rtp_session[0]));
sp->ui_activated = FALSE;
}
}
void
tx_igain_update(tx_buffer *tb)
{
tx_buffer_validate(tb);
sd_reset(tb->sp->auto_sd);
agc_reset(tb->agc);
}
int
tx_is_sending(tx_buffer *tb)
{
return tb->sending_audio;
}
double
tx_get_bps(tx_buffer *tb)
{
tx_buffer_validate(tb);
if (tb->bps_bytes_sent == 0) {
return 0.0;
} else {
uint32_t dms;
double bps;
timestamp_t delta = ts_abs_diff(tb->bps_last_update, tb->sp->cur_ts);
dms = timestamp_to_us(delta);
bps = tb->bps_bytes_sent * 8e6 / (double)dms;
tb->bps_bytes_sent = 0;
tb->bps_last_update = tb->sp->cur_ts;
return bps;
}
}
static void
tx_read_sndfile(session_t *sp, uint16_t tx_freq, uint16_t tx_channels, tx_unit *u)
{
sndfile_fmt_t sfmt;
int samples_read, dst_samples;
snd_get_format(sp->in_file, &sfmt);
if (sfmt.channels != tx_channels || sfmt.sample_rate != tx_freq) {
converter_fmt_t target;
const converter_fmt_t *actual;
coded_unit in, out;
target.src_channels = (uint16_t)sfmt.channels;
target.src_freq = (uint32_t)sfmt.sample_rate;
target.dst_channels = (uint16_t)tx_channels;
target.dst_freq = tx_freq;
/* Check if existing converter exists and whether valid */
if (sp->in_file_converter != NULL) {
actual = converter_get_format(sp->in_file_converter);
if (memcmp(actual, &target, sizeof(converter_fmt_t)) != 0) {
converter_destroy(&sp->in_file_converter);
}
}
/* Create relevent converter if necessary */
if (sp->in_file_converter == NULL) {
const converter_details_t *details = NULL;
uint32_t i, n;
/* We iterate through available converters
* since they have different capabilities,
* specifically MS-ACM does m*8:n*11025 and
* the RAT ones don't at time of writing.
*/
n = converter_get_count();
for(i = 0; i < n; i++) {
details = converter_get_details(i);
if (converter_create(details->id, &target, &sp->in_file_converter)) {
debug_msg("Created converter %s for sound file conversion\n", details->name);
break;
}
}
if (i == n) {
debug_msg("Could not create suitable converter for sound file\n");
snd_read_close(&sp->in_file);
return;
}
}
dst_samples = u->dur_used * tx_channels;
/* Prepare block to read audio into */
in.id = codec_get_native_coding(target.src_freq, target.src_channels);
in.state = NULL;
in.state_len = 0;
in.data_len = sizeof(sample) * dst_samples *
(target.src_freq * target.src_channels) /
(target.dst_freq * target.dst_channels);
in.data = (u_char*)block_alloc(in.data_len);
/* Get the sound from file */
samples_read = snd_read_audio(&sp->in_file, (sample*)in.data, (uint16_t)(in.data_len / sizeof(sample)));
if (samples_read == 0) {
/* File is paused */
codec_clear_coded_unit(&in);
return;
}
/* Prepare output block */
memset(&out, 0, sizeof(out));
converter_process(sp->in_file_converter, &in, &out);
assert((uint32_t)dst_samples == out.data_len / sizeof(sample));
memcpy(u->data, out.data, dst_samples * sizeof(sample));
/* Tidy up */
codec_clear_coded_unit(&in);
codec_clear_coded_unit(&out);
} else {
samples_read = snd_read_audio(&sp->in_file, u->data, (uint16_t)(u->dur_used * tx_channels));
}
if (samples_read > 0) {
u->live_capture = FALSE;
}
}
uint32_t
tx_get_rtp_time(session_t *sp)
{
return ts_seq32_out(&sp->tb->up_seq, sp->tb->sample_rate, sp->cur_ts);
}
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