static guint
count_packets (GstRtpQCELPDepay * depay, guint8 * data, guint size)
{
guint count = 0;
while (size > 0) {
gint frame_len;
frame_len = get_frame_len (depay, data[0]);
/* 0 is invalid and we throw away the remainder of the frames */
if (frame_len == 0)
break;
if (frame_len < 0)
frame_len = -frame_len;
if (frame_len > size)
break;
size -= frame_len;
data += frame_len;
count++;
}
return count;
}
void end_audio_frame() {
if (frame_offset == 0)
// No audio added; blip_end_frame() dislikes being called with an
// offset of 0
return;
assert(!(is_backwards_frame && frame_offset != get_frame_len()));
// Bring the signal level at the end of the frame to zero as outlined in
// set_audio_signal_level()
set_audio_signal_level(0);
blip_end_frame(blip, frame_offset);
if (playback_started) {
// Fudge playback rate by an amount proportional to the difference
// between the desired and current buffer fill levels to try to steer
// towards it
double const fudge_factor = 1.0 + 2*max_adjust*(0.5 - fill_level());
blip_set_rates(blip, cpu_clock_rate, sample_rate*fudge_factor);
}
else {
if (fill_level() >= 0.5) {
start_audio_playback();
playback_started = true;
}
}
int const n_samples = blip_read_samples(blip, blip_samples, ARRAY_LEN(blip_samples), 0);
// We expect to read all samples from blip_buf. If something goes wrong and
// we don't, clear the buffer to prevent data piling up in blip_buf's
// buffer (which lacks bounds checking).
int const avail = blip_samples_avail(blip);
if (avail != 0) {
printf("Warning: didn't read all samples from blip_buf (%d samples remain) - dropping samples\n",
avail);
blip_clear(blip);
}
#ifdef RECORD_MOVIE
add_movie_audio_frame(blip_samples, n_samples);
#endif
// Save the samples to the audio ring buffer
lock_audio();
write_samples(blip_samples, n_samples);
unlock_audio();
}
void set_audio_signal_level(int16_t level) {
// TODO: Do something to reduce the initial pop here?
static int16_t previous_signal_level = 0;
unsigned time = frame_offset;
int delta = level - previous_signal_level;
if (is_backwards_frame) {
// Flip deltas and add them from the end of the frame to reverse audio.
// Since the exact length of the frame can't be known in advance, the
// length of each frame is recorded when it is saved to the rewind
// buffer.
//
// This is easiest to visualize by thinking of deltas as fenceposts and
// the signal level as spans between them. While rewinding, the signal
// level that's being set should be considered the one to the left of
// the fencepost.
//
// One complication is the boundary between frames while rewinding -
// there the final sample added to one frame is not followed in time by
// the first sample of the next frame. To solve this, we bring the
// signal level down to zero at the end of each frame, and then adjust
// it to the correct value in the next frame (when rewinding, "to zero"
// becomes "from zero", and everything still works out). We also call
// begin_frame() between frames to invalidate the cached signal level
// in apu.cpp. Together this allows frames to be mixed-and-matched
// arbitrarily in time.
//
// Thanks to Blargg for help on this.
time = get_frame_len() - time;
delta = -delta;
}
blip_add_delta(blip, time, delta);
previous_signal_level = level;
}
static GstBuffer *
gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp)
{
GstRtpQCELPDepay *depay;
GstBuffer *outbuf;
GstClockTime timestamp;
guint payload_len, offset, index;
guint8 *payload;
guint LLL, NNN;
depay = GST_RTP_QCELP_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
if (payload_len < 2)
goto too_small;
timestamp = GST_BUFFER_PTS (rtp->buffer);
payload = gst_rtp_buffer_get_payload (rtp);
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |RR | LLL | NNN |
* +-+-+-+-+-+-+-+-+
*/
/* RR = payload[0] >> 6; */
LLL = (payload[0] & 0x38) >> 3;
NNN = (payload[0] & 0x07);
payload_len--;
payload++;
GST_DEBUG_OBJECT (depay, "LLL %u, NNN %u", LLL, NNN);
if (LLL > 5)
goto invalid_lll;
if (NNN > LLL)
goto invalid_nnn;
if (LLL != 0) {
/* we are interleaved */
if (!depay->interleaved) {
guint size;
GST_DEBUG_OBJECT (depay, "starting interleaving group");
/* bundling is not allowed to change in one interleave group */
depay->bundling = count_packets (depay, payload, payload_len);
GST_DEBUG_OBJECT (depay, "got bundling of %u", depay->bundling);
/* we have one bundle where NNN goes from 0 to L, we don't store the index
* 0 frames, so L+1 packets. Each packet has 'bundling - 1' packets */
size = (depay->bundling - 1) * (LLL + 1);
/* create the array to hold the packets */
if (depay->packets == NULL)
depay->packets = g_ptr_array_sized_new (size);
GST_DEBUG_OBJECT (depay, "created packet array of size %u", size);
g_ptr_array_set_size (depay->packets, size);
/* we were previously not interleaved, figure out how much space we
* need to deinterleave */
depay->interleaved = TRUE;
}
} else {
/* we are not interleaved */
if (depay->interleaved) {
GST_DEBUG_OBJECT (depay, "stopping interleaving");
/* flush packets if we were previously interleaved */
flush_packets (depay);
}
depay->bundling = 0;
}
index = 0;
offset = 1;
while (payload_len > 0) {
gint frame_len;
gboolean do_erasure;
frame_len = get_frame_len (depay, payload[0]);
GST_DEBUG_OBJECT (depay, "got frame len %d", frame_len);
if (frame_len == 0)
goto invalid_frame;
if (frame_len < 0) {
/* need to add an erasure frame but we can recover */
frame_len = -frame_len;
do_erasure = TRUE;
} else {
do_erasure = FALSE;
}
if (frame_len > payload_len)
goto invalid_frame;
if (do_erasure) {
/* create erasure frame */
outbuf = create_erasure_buffer (depay);
} else {
/* each frame goes into its buffer */
outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, frame_len);
}
GST_BUFFER_PTS (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = FRAME_DURATION;
gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
if (!depay->interleaved || index == 0) {
/* not interleaved or first frame in packet, just push */
gst_rtp_base_depayload_push (depayload, outbuf);
if (timestamp != -1)
timestamp += FRAME_DURATION;
} else {
/* put in interleave buffer */
add_packet (depay, LLL, NNN, index, outbuf);
if (timestamp != -1)
timestamp += (FRAME_DURATION * (LLL + 1));
}
payload_len -= frame_len;
payload += frame_len;
offset += frame_len;
index++;
/* discard excess packets */
if (depay->bundling > 0 && depay->bundling <= index)
break;
}
while (index < depay->bundling) {
GST_DEBUG_OBJECT (depay, "filling with erasure buffer");
/* fill remainder with erasure packets */
outbuf = create_erasure_buffer (depay);
add_packet (depay, LLL, NNN, index, outbuf);
index++;
}
if (depay->interleaved && LLL == NNN) {
GST_DEBUG_OBJECT (depay, "interleave group ended, flushing");
/* we have the complete interleave group, flush */
flush_packets (depay);
}
return NULL;
/* ERRORS */
too_small:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP payload too small (%d)", payload_len));
return NULL;
}
invalid_lll:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP invalid LLL received (%d)", LLL));
return NULL;
}
invalid_nnn:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP invalid NNN received (%d)", NNN));
return NULL;
}
invalid_frame:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP invalid frame received"));
return NULL;
}
}
