static GstClock *
gst_base_audio_src_provide_clock (GstElement * elem)
{
GstBaseAudioSrc *src;
GstClock *clock;
src = GST_BASE_AUDIO_SRC (elem);
/* we have no ringbuffer (must be NULL state) */
if (src->ringbuffer == NULL)
goto wrong_state;
if (!gst_ring_buffer_is_acquired (src->ringbuffer))
goto wrong_state;
GST_OBJECT_LOCK (src);
if (!src->priv->provide_clock)
goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (src->clock));
GST_OBJECT_UNLOCK (src);
return clock;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer not acquired");
return NULL;
}
clock_disabled:
{
GST_DEBUG_OBJECT (src, "clock provide disabled");
GST_OBJECT_UNLOCK (src);
return NULL;
}
}
static GstFlowReturn
gst_audio_ringbuffer_render (GstAudioRingbuffer * ringbuffer, GstBuffer * buf)
{
GstRingBuffer *rbuf;
gint bps, accum;
guint size;
guint samples, written, out_samples;
gint64 diff, align, ctime, cstop;
guint8 *data;
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
gboolean align_next;
rbuf = ringbuffer->buffer;
/* can't do anything when we don't have the device */
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (rbuf)))
goto wrong_state;
bps = rbuf->spec.bytes_per_sample;
size = GST_BUFFER_SIZE (buf);
if (G_UNLIKELY (size % bps) != 0)
goto wrong_size;
samples = size / bps;
out_samples = samples;
in_offset = GST_BUFFER_OFFSET (buf);
time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (ringbuffer,
"time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT
", samples %u", GST_TIME_ARGS (time), in_offset,
GST_TIME_ARGS (ringbuffer->sink_segment.start), samples);
data = GST_BUFFER_DATA (buf);
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
rbuf->spec.rate);
if (!gst_segment_clip (&ringbuffer->sink_segment, GST_FORMAT_TIME, time, stop,
&ctime, &cstop))
goto out_of_segment;
/* see if some clipping happened */
diff = ctime - time;
if (diff > 0) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (ringbuffer, "clipping start to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
samples -= diff;
data += diff * bps;
time = ctime;
}
diff = stop - cstop;
if (diff > 0) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (ringbuffer, "clipping stop to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
samples -= diff;
stop = cstop;
}
/* bring buffer start and stop times to running time */
render_start =
gst_segment_to_running_time (&ringbuffer->sink_segment, GST_FORMAT_TIME,
time);
render_stop =
gst_segment_to_running_time (&ringbuffer->sink_segment, GST_FORMAT_TIME,
stop);
GST_DEBUG_OBJECT (ringbuffer,
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start,
rbuf->spec.rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop,
rbuf->spec.rate, GST_SECOND);
/* positive playback rate, first sample is render_start, negative rate, first
* sample is render_stop. When no rate conversion is active, render exactly
* the amount of input samples to avoid aligning to rounding errors. */
if (ringbuffer->sink_segment.rate >= 0.0) {
sample_offset = render_start;
if (ringbuffer->sink_segment.rate == 1.0)
render_stop = sample_offset + samples;
} else {
sample_offset = render_stop;
if (ringbuffer->sink_segment.rate == -1.0)
render_start = sample_offset + samples;
}
/* always resync after a discont */
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
GST_DEBUG_OBJECT (ringbuffer, "resync after discont");
goto no_align;
}
/* resync when we don't know what to align the sample with */
if (G_UNLIKELY (ringbuffer->next_sample == -1)) {
GST_DEBUG_OBJECT (ringbuffer,
"no align possible: no previous sample position known");
goto no_align;
}
/* now try to align the sample to the previous one, first see how big the
* difference is. */
if (sample_offset >= ringbuffer->next_sample)
diff = sample_offset - ringbuffer->next_sample;
else
diff = ringbuffer->next_sample - sample_offset;
/* we tollerate half a second diff before we start resyncing. This
* should be enough to compensate for various rounding errors in the timestamp
* and sample offset position. We always resync if we got a discont anyway and
* non-discont should be aligned by definition. */
if (G_LIKELY (diff < rbuf->spec.rate / DIFF_TOLERANCE)) {
/* calc align with previous sample */
align = ringbuffer->next_sample - sample_offset;
GST_DEBUG_OBJECT (ringbuffer,
"align with prev sample, ABS (%" G_GINT64_FORMAT ") < %d", align,
rbuf->spec.rate / DIFF_TOLERANCE);
} else {
/* bring sample diff to seconds for error message */
diff = gst_util_uint64_scale_int (diff, GST_SECOND, rbuf->spec.rate);
/* timestamps drifted apart from previous samples too much, we need to
* resync. We log this as an element warning. */
GST_ELEMENT_WARNING (ringbuffer, CORE, CLOCK,
("Compensating for audio synchronisation problems"),
("Unexpected discontinuity in audio timestamps of more "
"than half a second (%" GST_TIME_FORMAT "), resyncing",
GST_TIME_ARGS (diff)));
align = 0;
}
ringbuffer->last_align = align;
/* apply alignment */
render_start += align;
render_stop += align;
no_align:
/* number of target samples is difference between start and stop */
out_samples = render_stop - render_start;
/* we render the first or last sample first, depending on the rate */
if (ringbuffer->sink_segment.rate >= 0.0)
sample_offset = render_start;
else
sample_offset = render_stop;
GST_DEBUG_OBJECT (ringbuffer, "rendering at %" G_GUINT64_FORMAT " %d/%d",
sample_offset, samples, out_samples);
/* we need to accumulate over different runs for when we get interrupted */
accum = 0;
align_next = TRUE;
do {
written =
gst_ring_buffer_commit_full (rbuf, &sample_offset, data, samples,
out_samples, &accum);
GST_DEBUG_OBJECT (ringbuffer, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
if (written == samples)
break;
GST_OBJECT_LOCK (ringbuffer);
if (ringbuffer->flushing)
goto flushing;
GST_OBJECT_UNLOCK (ringbuffer);
/* if we got interrupted, we cannot assume that the next sample should
* be aligned to this one */
align_next = FALSE;
samples -= written;
data += written * bps;
} while (TRUE);
if (align_next)
ringbuffer->next_sample = sample_offset;
else
ringbuffer->next_sample = -1;
GST_DEBUG_OBJECT (ringbuffer, "next sample expected at %" G_GUINT64_FORMAT,
ringbuffer->next_sample);
if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= ringbuffer->sink_segment.stop) {
GST_DEBUG_OBJECT (ringbuffer,
"start playback because we are at the end of segment");
gst_ring_buffer_start (rbuf);
}
return GST_FLOW_OK;
/* SPECIAL cases */
out_of_segment:
{
GST_DEBUG_OBJECT (ringbuffer,
"dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
GST_TIME_FORMAT, GST_TIME_ARGS (time),
GST_TIME_ARGS (ringbuffer->sink_segment.start));
return GST_FLOW_OK;
}
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (ringbuffer, "ringbuffer not negotiated");
GST_ELEMENT_ERROR (ringbuffer, STREAM, FORMAT, (NULL),
("ringbuffer not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
wrong_size:
{
GST_DEBUG_OBJECT (ringbuffer, "wrong size");
GST_ELEMENT_ERROR (ringbuffer, STREAM, WRONG_TYPE,
(NULL), ("ringbuffer received buffer of wrong size."));
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (ringbuffer, "ringbuffer is flushing");
GST_OBJECT_UNLOCK (ringbuffer);
return GST_FLOW_WRONG_STATE;
}
}
static GstFlowReturn
gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstBuffer ** outbuf)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstBuffer *buf;
guchar *data;
guint samples, total_samples;
guint64 sample;
gint bps;
GstRingBuffer *ringbuffer;
GstRingBufferSpec *spec;
guint read;
GstClockTime timestamp, duration;
GstClock *clock;
ringbuffer = src->ringbuffer;
spec = &ringbuffer->spec;
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer)))
goto wrong_state;
bps = spec->bytes_per_sample;
if ((length == 0 && bsrc->blocksize == 0) || length == -1)
/* no length given, use the default segment size */
length = spec->segsize;
else
/* make sure we round down to an integral number of samples */
length -= length % bps;
/* figure out the offset in the ringbuffer */
if (G_UNLIKELY (offset != -1)) {
sample = offset / bps;
/* if a specific offset was given it must be the next sequential
* offset we expect or we fail for now. */
if (src->next_sample != -1 && sample != src->next_sample)
goto wrong_offset;
} else {
/* calculate the sequentially next sample we need to read. This can jump and
* create a DISCONT. */
sample = gst_base_audio_src_get_offset (src);
}
GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT, sample);
/* get the number of samples to read */
total_samples = samples = length / bps;
/* FIXME, using a bufferpool would be nice here */
buf = gst_buffer_new_and_alloc (length);
data = GST_BUFFER_DATA (buf);
do {
read = gst_ring_buffer_read (ringbuffer, sample, data, samples);
GST_DEBUG_OBJECT (src, "read %u of %u", read, samples);
/* if we read all, we're done */
if (read == samples)
break;
/* else something interrupted us and we wait for playing again. */
GST_DEBUG_OBJECT (src, "wait playing");
if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
goto stopped;
GST_DEBUG_OBJECT (src, "continue playing");
/* read next samples */
sample += read;
samples -= read;
data += read * bps;
} while (TRUE);
/* mark discontinuity if needed */
if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {
GST_WARNING_OBJECT (src,
"create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
G_GUINT64_FORMAT, sample - src->next_sample, sample);
GST_ELEMENT_WARNING (src, CORE, CLOCK,
(_("Can't record audio fast enough")),
("dropped %" G_GUINT64_FORMAT " samples", sample - src->next_sample));
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
src->next_sample = sample + samples;
/* get the normal timestamp to get the duration. */
timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
duration = gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
spec->rate) - timestamp;
GST_OBJECT_LOCK (src);
if (!(clock = GST_ELEMENT_CLOCK (src)))
goto no_sync;
if (clock != src->clock) {
/* we are slaved, check how to handle this */
switch (src->priv->slave_method) {
case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
/* not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produces more or less samples based
* on the clock drift */
case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
{
GstClockTime running_time;
GstClockTime base_time;
GstClockTime current_time;
guint64 running_time_sample;
gint running_time_segment;
gint current_segment;
gint segment_skew;
gint sps;
/* samples per segment */
sps = ringbuffer->samples_per_seg;
/* get the current time */
current_time = gst_clock_get_time (clock);
/* get the basetime */
base_time = GST_ELEMENT_CAST (src)->base_time;
/* get the running_time */
running_time = current_time - base_time;
/* the running_time converted to a sample (relative to the ringbuffer) */
running_time_sample =
gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND);
/* the segmentnr corrensponding to running_time, round down */
running_time_segment = running_time_sample / sps;
/* the segment currently read from the ringbuffer */
current_segment = sample / sps;
/* the skew we have between running_time and the ringbuffertime */
segment_skew = running_time_segment - current_segment;
GST_DEBUG_OBJECT (bsrc, "\n running_time = %" GST_TIME_FORMAT
"\n timestamp = %" GST_TIME_FORMAT
"\n running_time_segment = %d"
"\n current_segment = %d"
"\n segment_skew = %d",
GST_TIME_ARGS (running_time),
GST_TIME_ARGS (timestamp),
running_time_segment, current_segment, segment_skew);
/* Resync the ringbuffer if:
* 1. We get one segment into the future.
* This is clearly a lie, because we can't
* possibly have a buffer with timestamp 1 at
* time 0. (unless it has time-travelled...)
*
* 2. We are more than the length of the ringbuffer behind.
* The length of the ringbuffer then gets to dictate
* the threshold for what is concidered "too late"
*
* 3. If this is our first buffer.
* We know that we should catch up to running_time
* the first time we are ran.
*/
if ((segment_skew < 0) ||
(segment_skew >= ringbuffer->spec.segtotal) ||
(current_segment == 0)) {
gint segments_written;
gint first_segment;
gint last_segment;
gint new_last_segment;
gint segment_diff;
gint new_first_segment;
guint64 new_sample;
/* we are going to say that the last segment was captured at the current time
(running_time), minus one segment of creation-latency in the ringbuffer.
This can be thought of as: The segment arrived in the ringbuffer at time X, and
that means it was created at time X - (one segment). */
new_last_segment = running_time_segment - 1;
/* for better readablity */
first_segment = current_segment;
/* get the amount of segments written from the device by now */
segments_written = g_atomic_int_get (&ringbuffer->segdone);
/* subtract the base to segments_written to get the number of the
last written segment in the ringbuffer (one segment written = segment 0) */
last_segment = segments_written - ringbuffer->segbase - 1;
/* we see how many segments the ringbuffer was timeshifted */
segment_diff = new_last_segment - last_segment;
/* we move the first segment an equal amount */
new_first_segment = first_segment + segment_diff;
/* and we also move the segmentbase the same amount */
ringbuffer->segbase -= segment_diff;
/* we calculate the new sample value */
new_sample = ((guint64) new_first_segment) * sps;
/* and get the relative time to this -> our new timestamp */
timestamp =
gst_util_uint64_scale_int (new_sample, GST_SECOND, spec->rate);
/* we update the next sample accordingly */
src->next_sample = new_sample + samples;
GST_DEBUG_OBJECT (bsrc,
"Timeshifted the ringbuffer with %d segments: "
"Updating the timestamp to %" GST_TIME_FORMAT ", "
"and src->next_sample to %" G_GUINT64_FORMAT, segment_diff,
GST_TIME_ARGS (timestamp), src->next_sample);
}
break;
}
case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
{
GstClockTime base_time, latency;
/* We are slaved to another clock, take running time of the pipeline clock and
* timestamp against it. Somebody else in the pipeline should figure out the
* clock drift. We keep the duration we calculated above. */
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
/* subtract latency */
latency =
gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
if (timestamp > latency)
timestamp -= latency;
else
timestamp = 0;
}
case GST_BASE_AUDIO_SRC_SLAVE_NONE:
break;
}
} else {
GstClockTime base_time;
/* we are not slaved, subtract base_time */
base_time = GST_ELEMENT_CAST (src)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
}
no_sync:
GST_OBJECT_UNLOCK (src);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_OFFSET (buf) = sample;
GST_BUFFER_OFFSET_END (buf) = sample + samples;
*outbuf = buf;
return GST_FLOW_OK;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
wrong_offset:
{
GST_ELEMENT_ERROR (src, RESOURCE, SEEK,
(NULL), ("resource can only be operated on sequentially but offset %"
G_GUINT64_FORMAT " was given", offset));
return GST_FLOW_ERROR;
}
stopped:
{
gst_buffer_unref (buf);
GST_DEBUG_OBJECT (src, "ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
