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audio{en,de}coder: Track input and output segments separately

Browse source 
They can go out of sync for some time if processing of buffers
on the old segment happens after the segment was received.
This commit is contained in:
Sebastian Dröge 2012-03-30 13:21:09 +02:00
parent 9cd9f00799
commit a103fa85a9
4 changed files with 118 additions and 33 deletions
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67
gst-libs/gst/audio/gstaudiodecoder.c
View file

@ -460,7 +460,8 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
dec->priv->taglist = NULL;
}
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
gst_segment_init (&dec->input_segment, GST_FORMAT_TIME);
gst_segment_init (&dec->output_segment, GST_FORMAT_TIME);
g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (dec->priv->pending_events);
@ -644,7 +645,7 @@ gst_audio_decoder_push_forward (GstAudioDecoder * dec, GstBuffer * buf)
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* clip buffer */
buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->info.rate,
buf = gst_audio_buffer_clip (buf, &dec->output_segment, ctx->info.rate,
ctx->info.bpf);
if (G_UNLIKELY (!buf)) {
GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
@ -662,7 +663,7 @@ gst_audio_decoder_push_forward (GstAudioDecoder * dec, GstBuffer * buf)
if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
/* duration should always be valid for raw audio */
g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
dec->segment.position =
dec->output_segment.position =
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
}
@ -759,7 +760,7 @@ again:
}
if (G_LIKELY (buf)) {
if (dec->segment.rate > 0.0) {
if (dec->output_segment.rate > 0.0) {
ret = gst_audio_decoder_push_forward (dec, buf);
GST_LOG_OBJECT (dec, "buffer pushed: %s", gst_flow_get_name (ret));
} else {
@ -777,6 +778,29 @@ again:
return ret;
}
static gboolean
gst_audio_decoder_push_event (GstAudioDecoder * dec, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
GstSegment seg;
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_event_copy_segment (event, &seg);
GST_DEBUG_OBJECT (dec, "starting segment %" GST_SEGMENT_FORMAT, &seg);
dec->output_segment = seg;
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
default:
break;
}
return gst_pad_push_event (dec->srcpad, event);
}
/**
* gst_audio_decoder_finish_frame:
* @dec: a #GstAudioDecoder
@ -841,7 +865,7 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
GST_DEBUG_OBJECT (dec, "Pushing pending events");
for (l = pending_events; l; l = l->next)
gst_pad_push_event (dec->srcpad, l->data);
gst_audio_decoder_push_event (dec, l->data);
g_list_free (pending_events);
}
@ -926,7 +950,7 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
if (gst_tag_list_is_empty (priv->taglist)) {
gst_tag_list_free (priv->taglist);
} else {
gst_pad_push_event (dec->srcpad, gst_event_new_tag (priv->taglist));
gst_audio_decoder_push_event (dec, gst_event_new_tag (priv->taglist));
}
priv->taglist = NULL;
}
@ -1125,7 +1149,7 @@ gst_audio_decoder_drain (GstAudioDecoder * dec)
/* dispatch reverse pending buffers */
/* chain eventually calls upon drain as well, but by that time
* gather list should be clear, so ok ... */
if (dec->segment.rate < 0.0 && dec->priv->gather)
if (dec->output_segment.rate < 0.0 && dec->priv->gather)
gst_audio_decoder_chain_reverse (dec, NULL);
/* have subclass give all it can */
ret = gst_audio_decoder_push_buffers (dec, TRUE);
@ -1161,7 +1185,8 @@ gst_audio_decoder_flush (GstAudioDecoder * dec, gboolean hard)
ret = gst_audio_decoder_drain (dec);
} else {
gst_audio_decoder_clear_queues (dec);
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
gst_segment_init (&dec->input_segment, GST_FORMAT_TIME);
gst_segment_init (&dec->output_segment, GST_FORMAT_TIME);
dec->priv->error_count = 0;
}
/* only bother subclass with flushing if known it is already alive
@ -1433,14 +1458,15 @@ gst_audio_decoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
/* buffer may claim DISCONT loudly, if it can't tell us where we are now,
* we'll stick to where we were ...
* Particularly useful/needed for upstream BYTE based */
if (dec->segment.rate > 0.0 && !GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
if (dec->input_segment.rate > 0.0
&& !GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
GST_DEBUG_OBJECT (dec, "... but restoring previous ts tracking");
dec->priv->base_ts = ts;
dec->priv->samples = samples;
}
}
if (dec->segment.rate > 0.0)
if (dec->input_segment.rate > 0.0)
ret = gst_audio_decoder_chain_forward (dec, buffer);
else
ret = gst_audio_decoder_chain_reverse (dec, buffer);
@ -1524,16 +1550,17 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
GST_DEBUG_OBJECT (dec,
"segment update: plc %d, do_plc %d, position %" GST_TIME_FORMAT,
dec->priv->plc, dec->priv->ctx.do_plc,
GST_TIME_ARGS (dec->segment.position));
GST_TIME_ARGS (dec->input_segment.position));
if (dec->priv->plc && dec->priv->ctx.do_plc &&
dec->segment.rate > 0.0 && dec->segment.position < start) {
dec->input_segment.rate > 0.0
&& dec->input_segment.position < start) {
GstAudioDecoderClass *klass;
GstBuffer *buf;
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
/* hand subclass empty frame with duration that needs covering */
buf = gst_buffer_new ();
GST_BUFFER_DURATION (buf) = start - dec->segment.position;
GST_BUFFER_DURATION (buf) = start - dec->input_segment.position;
/* best effort, not much error handling */
gst_audio_decoder_handle_frame (dec, klass, buf);
}
@ -1552,7 +1579,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
}
/* and follow along with segment */
dec->segment = seg;
dec->input_segment = seg;
dec->priv->pending_events =
g_list_append (dec->priv->pending_events, event);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
@ -1573,7 +1600,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
/* Forward FLUSH_STOP, it is expected to be forwarded immediately
* and no buffers are queued anyway. */
ret = gst_pad_push_event (dec->srcpad, event);
ret = gst_audio_decoder_push_event (dec, event);
break;
case GST_EVENT_EOS:
@ -1583,7 +1610,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
/* Forward EOS because no buffer or serialized event will come after
* EOS and nothing could trigger another _finish_frame() call. */
ret = gst_pad_push_event (dec->srcpad, event);
ret = gst_audio_decoder_push_event (dec, event);
break;
case GST_EVENT_CAPS:
@ -1671,7 +1698,7 @@ gst_audio_decoder_do_seek (GstAudioDecoder * dec, GstEvent * event)
return FALSE;
}
memcpy (&seek_segment, &dec->segment, sizeof (seek_segment));
memcpy (&seek_segment, &dec->output_segment, sizeof (seek_segment));
gst_segment_do_seek (&seek_segment, rate, format, flags, start_type,
start_time, end_type, end_time, NULL);
start_time = seek_segment.position;
@ -1956,9 +1983,11 @@ gst_audio_decoder_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
}
/* we start from the last seen time */
time = dec->segment.position;
time = dec->output_segment.position;
/* correct for the segment values */
time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time);
time =
gst_segment_to_stream_time (&dec->output_segment, GST_FORMAT_TIME,
time);
GST_LOG_OBJECT (dec,
"query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time));

23
gst-libs/gst/audio/gstaudiodecoder.h
View file

@ -84,6 +84,26 @@ G_BEGIN_DECLS
#define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock)
#define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock)
/**
* GST_AUDIO_DECODER_INPUT_SEGMENT:
* @obj: base parse instance
*
* Gives the input segment of the element.
*
* Since: 0.10.36
*/
#define GST_AUDIO_DECODER_INPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->input_segment)
/**
* GST_AUDIO_DECODER_OUTPUT_SEGMENT:
* @obj: base parse instance
*
* Gives the output segment of the element.
*
* Since: 0.10.36
*/
#define GST_AUDIO_DECODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->output_segment)
typedef struct _GstAudioDecoder GstAudioDecoder;
typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
@ -161,7 +181,8 @@ struct _GstAudioDecoder
GRecMutex stream_lock;
/* MT-protected (with STREAM_LOCK) */
GstSegment segment;
GstSegment input_segment;
GstSegment output_segment;
/*< private >*/
GstAudioDecoderPrivate *priv;

42
gst-libs/gst/audio/gstaudioencoder.c
View file

@ -456,7 +456,8 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
enc->priv->pending_events = NULL;
}
gst_segment_init (&enc->segment, GST_FORMAT_TIME);
gst_segment_init (&enc->input_segment, GST_FORMAT_TIME);
gst_segment_init (&enc->output_segment, GST_FORMAT_TIME);
gst_adapter_clear (enc->priv->adapter);
enc->priv->got_data = FALSE;
@ -525,6 +526,29 @@ close_failed:
}
}
static gboolean
gst_audio_encoder_push_event (GstAudioEncoder * enc, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
GstSegment seg;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
gst_event_copy_segment (event, &seg);
GST_DEBUG_OBJECT (enc, "starting segment %" GST_SEGMENT_FORMAT, &seg);
enc->output_segment = seg;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
}
default:
break;
}
return gst_pad_push_event (enc->srcpad, event);
}
/**
* gst_audio_encoder_finish_frame:
* @enc: a #GstAudioEncoder
@ -585,7 +609,7 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
GST_DEBUG_OBJECT (enc, "Pushing pending events");
for (l = pending_events; l; l = l->next)
gst_pad_push_event (enc->srcpad, l->data);
gst_audio_encoder_push_event (enc, l->data);
g_list_free (pending_events);
}
@ -607,7 +631,7 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
caps);
#endif
GST_DEBUG_OBJECT (enc, "sending tags %" GST_PTR_FORMAT, tags);
gst_pad_push_event (enc->srcpad, gst_event_new_tag (tags));
gst_audio_encoder_push_event (enc, gst_event_new_tag (tags));
}
/* remove corresponding samples from input */
@ -940,7 +964,7 @@ gst_audio_encoder_set_base_gp (GstAudioEncoder * enc)
/* use running time for granule */
/* incoming data is clipped, so a valid input should yield a valid output */
ts = gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME,
ts = gst_segment_to_running_time (&enc->input_segment, GST_FORMAT_TIME,
enc->priv->base_ts);
if (GST_CLOCK_TIME_IS_VALID (ts)) {
enc->priv->base_gp =
@ -1018,7 +1042,7 @@ gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
/* clip to segment */
/* NOTE: slightly painful linking -laudio only for this one ... */
buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->info.rate,
buffer = gst_audio_buffer_clip (buffer, &enc->input_segment, ctx->info.rate,
ctx->info.bpf);
if (G_UNLIKELY (!buffer)) {
GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
@ -1368,7 +1392,7 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
/* reset partially for new segment */
gst_audio_encoder_reset (enc, FALSE);
/* and follow along with segment */
enc->segment = seg;
enc->input_segment = seg;
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
@ -1379,7 +1403,7 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
}
case GST_EVENT_FLUSH_START:
res = gst_pad_push_event (enc->srcpad, event);
res = gst_audio_encoder_push_event (enc, event);
break;
case GST_EVENT_FLUSH_STOP:
@ -1396,7 +1420,7 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
enc->priv->pending_events = NULL;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = gst_pad_push_event (enc->srcpad, event);
res = gst_audio_encoder_push_event (enc, event);
break;
case GST_EVENT_EOS:
@ -1407,7 +1431,7 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
/* forward immediately because no buffer or serialized event
* will come after EOS and nothing could trigger another
* _finish_frame() call. */
res = gst_pad_push_event (enc->srcpad, event);
res = gst_audio_encoder_push_event (enc, event);
break;
case GST_EVENT_TAG:

19
gst-libs/gst/audio/gstaudioencoder.h
View file

@ -73,14 +73,24 @@ G_BEGIN_DECLS
#define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
/**
* GST_AUDIO_ENCODER_SEGMENT:
* GST_AUDIO_ENCODER_INPUT_SEGMENT:
* @obj: base parse instance
*
* Gives the segment of the element.
* Gives the input segment of the element.
*
* Since: 0.10.36
*/
#define GST_AUDIO_ENCODER_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->segment)
#define GST_AUDIO_ENCODER_INPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->input_segment)
/**
* GST_AUDIO_ENCODER_OUTPUT_SEGMENT:
* @obj: base parse instance
*
* Gives the output segment of the element.
*
* Since: 0.10.36
*/
#define GST_AUDIO_ENCODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->output_segment)
#define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)
@ -111,7 +121,8 @@ struct _GstAudioEncoder {
GRecMutex stream_lock;
/* MT-protected (with STREAM_LOCK) */
GstSegment segment;
GstSegment input_segment;
GstSegment output_segment;
/*< private >*/
GstAudioEncoderPrivate *priv;