2 * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
3 * Copyright (C) 2011 Nokia Corporation. All rights reserved.
4 * Contact: Stefan Kost <stefan.kost@nokia.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
19 * Boston, MA 02111-1307, USA.
23 * SECTION:gstaudioencoder
24 * @short_description: Base class for audio encoders
25 * @see_also: #GstBaseTransform
28 * This base class is for audio encoders turning raw audio samples into
31 * GstAudioEncoder and subclass should cooperate as follows.
34 * <itemizedlist><title>Configuration</title>
36 * Initially, GstAudioEncoder calls @start when the encoder element
37 * is activated, which allows subclass to perform any global setup.
40 * GstAudioEncoder calls @set_format to inform subclass of the format
41 * of input audio data that it is about to receive. Subclass should
42 * setup for encoding and configure various base class parameters
43 * appropriately, notably those directing desired input data handling.
44 * While unlikely, it might be called more than once, if changing input
45 * parameters require reconfiguration.
48 * GstAudioEncoder calls @stop at end of all processing.
52 * As of configuration stage, and throughout processing, GstAudioEncoder
53 * maintains various parameters that provide required context,
54 * e.g. describing the format of input audio data.
55 * Conversely, subclass can and should configure these context parameters
56 * to inform base class of its expectation w.r.t. buffer handling.
59 * <title>Data processing</title>
61 * Base class gathers input sample data (as directed by the context's
62 * frame_samples and frame_max) and provides this to subclass' @handle_frame.
65 * If codec processing results in encoded data, subclass should call
66 * @gst_audio_encoder_finish_frame to have encoded data pushed
67 * downstream. Alternatively, it might also call to indicate dropped
68 * (non-encoded) samples.
71 * Just prior to actually pushing a buffer downstream,
72 * it is passed to @pre_push.
75 * During the parsing process GstAudioEncoderClass will handle both
76 * srcpad and sinkpad events. Sink events will be passed to subclass
77 * if @event callback has been provided.
82 * <itemizedlist><title>Shutdown phase</title>
84 * GstAudioEncoder class calls @stop to inform the subclass that data
85 * parsing will be stopped.
91 * Subclass is responsible for providing pad template caps for
92 * source and sink pads. The pads need to be named "sink" and "src". It also
93 * needs to set the fixed caps on srcpad, when the format is ensured. This
94 * is typically when base class calls subclass' @set_format function, though
95 * it might be delayed until calling @gst_audio_encoder_finish_frame.
97 * In summary, above process should have subclass concentrating on
98 * codec data processing while leaving other matters to base class,
99 * such as most notably timestamp handling. While it may exert more control
100 * in this area (see e.g. @pre_push), it is very much not recommended.
102 * In particular, base class will either favor tracking upstream timestamps
103 * (at the possible expense of jitter) or aim to arrange for a perfect stream of
104 * output timestamps, depending on #GstAudioEncoder:perfect-ts.
105 * However, in the latter case, the input may not be so perfect or ideal, which
106 * is handled as follows. An input timestamp is compared with the expected
107 * timestamp as dictated by input sample stream and if the deviation is less
108 * than #GstAudioEncoder:tolerance, the deviation is discarded.
109 * Otherwise, it is considered a discontuinity and subsequent output timestamp
110 * is resynced to the new position after performing configured discontinuity
111 * processing. In the non-perfect-ts case, an upstream variation exceeding
112 * tolerance only leads to marking DISCONT on subsequent outgoing
113 * (while timestamps are adjusted to upstream regardless of variation).
114 * While DISCONT is also marked in the perfect-ts case, this one optionally
115 * (see #GstAudioEncoder:hard-resync)
116 * performs some additional steps, such as clipping of (early) input samples
117 * or draining all currently remaining input data, depending on the direction
118 * of the discontuinity.
120 * If perfect timestamps are arranged, it is also possible to request baseclass
121 * (usually set by subclass) to provide additional buffer metadata (in OFFSET
122 * and OFFSET_END) fields according to granule defined semantics currently
123 * needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count
124 * including buffer) and OFFSET_END to corresponding timestamp (as determined
125 * by same sample count and sample rate).
127 * Things that subclass need to take care of:
129 * <listitem><para>Provide pad templates</para></listitem>
131 * Set source pad caps when appropriate
134 * Inform base class of buffer processing needs using context's
135 * frame_samples and frame_bytes.
138 * Set user-configurable properties to sane defaults for format and
139 * implementing codec at hand, e.g. those controlling timestamp behaviour
140 * and discontinuity processing.
143 * Accept data in @handle_frame and provide encoded results to
144 * @gst_audio_encoder_finish_frame.
154 #define GST_USE_UNSTABLE_API
155 #include "gstaudioencoder.h"
156 #include <gst/base/gstadapter.h>
157 #include <gst/audio/audio.h>
163 GST_DEBUG_CATEGORY_STATIC (gst_audio_encoder_debug);
164 #define GST_CAT_DEFAULT gst_audio_encoder_debug
166 #define GST_AUDIO_ENCODER_GET_PRIVATE(obj) \
167 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_ENCODER, \
168 GstAudioEncoderPrivate))
179 #define DEFAULT_PERFECT_TS FALSE
180 #define DEFAULT_GRANULE FALSE
181 #define DEFAULT_HARD_RESYNC FALSE
182 #define DEFAULT_TOLERANCE 40000000
184 typedef struct _GstAudioEncoderContext
193 /* MT-protected (with LOCK) */
194 GstClockTime min_latency;
195 GstClockTime max_latency;
196 } GstAudioEncoderContext;
198 struct _GstAudioEncoderPrivate
200 /* activation status */
203 /* input base/first ts as basis for output ts;
204 * kept nearly constant for perfect_ts,
205 * otherwise resyncs to upstream ts */
206 GstClockTime base_ts;
207 /* corresponding base granulepos */
209 /* input samples processed and sent downstream so far (w.r.t. base_ts) */
212 /* currently collected sample data */
214 /* offset in adapter up to which already supplied to encoder */
216 /* mark outgoing discont */
218 /* to guess duration of drained data */
219 GstClockTime last_duration;
221 /* subclass provided data in processing round */
223 /* subclass gave all it could already */
225 /* subclass currently being forcibly drained */
228 /* output bps estimatation */
229 /* global in samples seen */
231 /* global bytes sent out */
234 /* context storage */
235 GstAudioEncoderContext ctx;
240 gboolean hard_resync;
245 static GstElementClass *parent_class = NULL;
247 static void gst_audio_encoder_class_init (GstAudioEncoderClass * klass);
248 static void gst_audio_encoder_init (GstAudioEncoder * parse,
249 GstAudioEncoderClass * klass);
252 gst_audio_encoder_get_type (void)
254 static GType audio_encoder_type = 0;
256 if (!audio_encoder_type) {
257 static const GTypeInfo audio_encoder_info = {
258 sizeof (GstAudioEncoderClass),
259 (GBaseInitFunc) NULL,
260 (GBaseFinalizeFunc) NULL,
261 (GClassInitFunc) gst_audio_encoder_class_init,
264 sizeof (GstAudioEncoder),
266 (GInstanceInitFunc) gst_audio_encoder_init,
268 const GInterfaceInfo preset_interface_info = {
269 NULL, /* interface_init */
270 NULL, /* interface_finalize */
271 NULL /* interface_data */
274 audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
275 "GstAudioEncoder", &audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
277 g_type_add_interface_static (audio_encoder_type, GST_TYPE_PRESET,
278 &preset_interface_info);
280 return audio_encoder_type;
283 static void gst_audio_encoder_finalize (GObject * object);
284 static void gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full);
286 static void gst_audio_encoder_set_property (GObject * object,
287 guint prop_id, const GValue * value, GParamSpec * pspec);
288 static void gst_audio_encoder_get_property (GObject * object,
289 guint prop_id, GValue * value, GParamSpec * pspec);
291 static gboolean gst_audio_encoder_sink_activate_push (GstPad * pad,
294 static gboolean gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event);
295 static gboolean gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps);
296 static GstFlowReturn gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer);
297 static gboolean gst_audio_encoder_src_query (GstPad * pad, GstQuery * query);
298 static gboolean gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query);
299 static const GstQueryType *gst_audio_encoder_get_query_types (GstPad * pad);
300 static GstCaps *gst_audio_encoder_sink_getcaps (GstPad * pad);
304 gst_audio_encoder_class_init (GstAudioEncoderClass * klass)
306 GObjectClass *gobject_class;
308 gobject_class = G_OBJECT_CLASS (klass);
309 parent_class = g_type_class_peek_parent (klass);
311 GST_DEBUG_CATEGORY_INIT (gst_audio_encoder_debug, "audioencoder", 0,
312 "audio encoder base class");
314 g_type_class_add_private (klass, sizeof (GstAudioEncoderPrivate));
316 gobject_class->set_property = gst_audio_encoder_set_property;
317 gobject_class->get_property = gst_audio_encoder_get_property;
319 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_encoder_finalize);
322 g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
323 g_param_spec_boolean ("perfect-timestamp", "Perfect Timestamps",
324 "Favour perfect timestamps over tracking upstream timestamps",
325 DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
326 g_object_class_install_property (gobject_class, PROP_GRANULE,
327 g_param_spec_boolean ("mark-granule", "Granule Marking",
328 "Apply granule semantics to buffer metadata (implies perfect-ts)",
329 DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
330 g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
331 g_param_spec_boolean ("hard-resync", "Hard Resync",
332 "Perform clipping and sample flushing upon discontinuity",
333 DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
334 g_object_class_install_property (gobject_class, PROP_TOLERANCE,
335 g_param_spec_int64 ("tolerance", "Tolerance",
336 "Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
337 0, G_MAXINT64, DEFAULT_TOLERANCE,
338 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
342 gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
344 GstPadTemplate *pad_template;
346 GST_DEBUG_OBJECT (enc, "gst_audio_encoder_init");
348 enc->priv = GST_AUDIO_ENCODER_GET_PRIVATE (enc);
350 /* only push mode supported */
352 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
353 g_return_if_fail (pad_template != NULL);
354 enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
355 gst_pad_set_event_function (enc->sinkpad,
356 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_event));
357 gst_pad_set_setcaps_function (enc->sinkpad,
358 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_setcaps));
359 gst_pad_set_getcaps_function (enc->sinkpad,
360 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_getcaps));
361 gst_pad_set_query_function (enc->sinkpad,
362 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_query));
363 gst_pad_set_chain_function (enc->sinkpad,
364 GST_DEBUG_FUNCPTR (gst_audio_encoder_chain));
365 gst_pad_set_activatepush_function (enc->sinkpad,
366 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_activate_push));
367 gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
369 GST_DEBUG_OBJECT (enc, "sinkpad created");
371 /* and we don't mind upstream traveling stuff that much ... */
373 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
374 g_return_if_fail (pad_template != NULL);
375 enc->srcpad = gst_pad_new_from_template (pad_template, "src");
376 gst_pad_set_query_function (enc->srcpad,
377 GST_DEBUG_FUNCPTR (gst_audio_encoder_src_query));
378 gst_pad_set_query_type_function (enc->srcpad,
379 GST_DEBUG_FUNCPTR (gst_audio_encoder_get_query_types));
380 gst_pad_use_fixed_caps (enc->srcpad);
381 gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
382 GST_DEBUG_OBJECT (enc, "src created");
384 enc->priv->adapter = gst_adapter_new ();
386 /* property default */
387 enc->priv->granule = DEFAULT_GRANULE;
388 enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
389 enc->priv->hard_resync = DEFAULT_HARD_RESYNC;
390 enc->priv->tolerance = DEFAULT_TOLERANCE;
393 gst_audio_encoder_reset (enc, TRUE);
394 GST_DEBUG_OBJECT (enc, "init ok");
398 gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
400 GST_OBJECT_LOCK (enc);
403 enc->priv->active = FALSE;
404 enc->priv->samples_in = 0;
405 enc->priv->bytes_out = 0;
406 gst_audio_info_clear (&enc->priv->ctx.info);
407 memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
410 gst_segment_init (&enc->segment, GST_FORMAT_TIME);
412 gst_adapter_clear (enc->priv->adapter);
413 enc->priv->got_data = FALSE;
414 enc->priv->drained = TRUE;
415 enc->priv->offset = 0;
416 enc->priv->base_ts = GST_CLOCK_TIME_NONE;
417 enc->priv->base_gp = -1;
418 enc->priv->samples = 0;
419 enc->priv->discont = FALSE;
421 GST_OBJECT_UNLOCK (enc);
425 gst_audio_encoder_finalize (GObject * object)
427 GstAudioEncoder *enc = GST_AUDIO_ENCODER (object);
429 g_object_unref (enc->priv->adapter);
431 G_OBJECT_CLASS (parent_class)->finalize (object);
435 * gst_audio_encoder_finish_frame:
436 * @enc: a #GstAudioEncoder
437 * @buffer: encoded data
438 * @samples: number of samples (per channel) represented by encoded data
440 * Collects encoded data and/or pushes encoded data downstream.
441 * Source pad caps must be set when this is called. Depending on the nature
442 * of the (framing of) the format, subclass can decide whether to push
443 * encoded data directly or to collect various "frames" in a single buffer.
444 * Note that the latter behaviour is recommended whenever the format is allowed,
445 * as it incurs no additional latency and avoids otherwise generating a
446 * a multitude of (small) output buffers. If not explicitly pushed,
447 * any available encoded data is pushed at the end of each processing cycle,
448 * i.e. which encodes as much data as available input data allows.
450 * If @samples < 0, then best estimate is all samples provided to encoder
451 * (subclass) so far. @buf may be NULL, in which case next number of @samples
452 * are considered discarded, e.g. as a result of discontinuous transmission,
453 * and a discontinuity is marked (note that @buf == NULL => push == TRUE).
455 * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
460 gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
463 GstAudioEncoderClass *klass;
464 GstAudioEncoderPrivate *priv;
465 GstAudioEncoderContext *ctx;
466 GstFlowReturn ret = GST_FLOW_OK;
468 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
470 ctx = &enc->priv->ctx;
472 /* subclass should know what it is producing by now */
473 g_return_val_if_fail (GST_PAD_CAPS (enc->srcpad) != NULL, GST_FLOW_ERROR);
474 /* subclass should not hand us no data */
475 g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
478 GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
479 buf ? GST_BUFFER_SIZE (buf) : -1, samples);
481 /* mark subclass still alive and providing */
482 priv->got_data = TRUE;
484 /* remove corresponding samples from input */
486 samples = (enc->priv->offset / ctx->info.bpf);
488 if (G_LIKELY (samples)) {
489 /* track upstream ts if so configured */
490 if (!enc->priv->perfect_ts) {
491 guint64 ts, distance;
493 ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
494 g_assert (distance % ctx->info.bpf == 0);
495 distance /= ctx->info.bpf;
496 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
497 GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
498 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
499 GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
500 /* when draining adapter might be empty and no ts to offer */
501 if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
502 GstClockTimeDiff diff;
503 GstClockTime old_ts, next_ts;
505 /* passed into another buffer;
506 * mild check for discontinuity and only mark if so */
508 gst_util_uint64_scale (distance, GST_SECOND, ctx->info.rate);
509 old_ts = priv->base_ts +
510 gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->info.rate);
511 diff = GST_CLOCK_DIFF (next_ts, old_ts);
512 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
513 /* only mark discontinuity if beyond tolerance */
514 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
515 diff > enc->priv->tolerance)) {
516 GST_DEBUG_OBJECT (enc, "marked discont");
517 priv->discont = TRUE;
519 if (diff > GST_SECOND / ctx->info.rate / 2 ||
520 diff < -GST_SECOND / ctx->info.rate / 2) {
521 GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
522 " at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
523 /* re-sync to upstream ts */
525 priv->samples = distance;
527 GST_LOG_OBJECT (enc, "new upstream ts only introduces jitter");
531 /* advance sample view */
532 if (G_UNLIKELY (samples * ctx->info.bpf > priv->offset)) {
533 if (G_LIKELY (!priv->force)) {
534 /* no way we can let this pass */
535 g_assert_not_reached ();
540 if (samples * ctx->info.bpf >= gst_adapter_available (priv->adapter))
541 gst_adapter_clear (priv->adapter);
543 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
546 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
547 priv->offset -= samples * ctx->info.bpf;
548 /* avoid subsequent stray prev_ts */
549 if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
550 gst_adapter_clear (priv->adapter);
552 /* sample count advanced below after buffer handling */
556 if (G_LIKELY (buf)) {
557 GST_LOG_OBJECT (enc, "taking %d bytes for output", GST_BUFFER_SIZE (buf));
558 buf = gst_buffer_make_metadata_writable (buf);
561 gst_buffer_set_caps (buf, GST_PAD_CAPS (enc->srcpad));
562 if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
563 /* FIXME ? lookahead could lead to weird ts and duration ?
564 * (particularly if not in perfect mode) */
565 /* mind sample rounding and produce perfect output */
566 GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
567 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
569 GST_DEBUG_OBJECT (enc, "out samples %d", samples);
570 if (G_LIKELY (samples > 0)) {
571 priv->samples += samples;
572 GST_BUFFER_DURATION (buf) = priv->base_ts +
573 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
574 ctx->info.rate) - GST_BUFFER_TIMESTAMP (buf);
575 priv->last_duration = GST_BUFFER_DURATION (buf);
577 /* duration forecast in case of handling remainder;
578 * the last one is probably like the previous one ... */
579 GST_BUFFER_DURATION (buf) = priv->last_duration;
581 if (priv->base_gp >= 0) {
583 /* FIXME: in longer run, muxer should take care of this ... */
584 /* offset_end = granulepos for ogg muxer */
585 GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples -
586 enc->priv->ctx.lookahead;
587 /* offset = timestamp corresponding to granulepos for ogg muxer */
588 GST_BUFFER_OFFSET (buf) =
589 GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf),
592 GST_BUFFER_OFFSET (buf) = priv->bytes_out;
593 GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf);
597 priv->bytes_out += GST_BUFFER_SIZE (buf);
599 if (G_UNLIKELY (priv->discont)) {
600 GST_LOG_OBJECT (enc, "marking discont");
601 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
602 priv->discont = FALSE;
605 if (klass->pre_push) {
606 /* last chance for subclass to do some dirty stuff */
607 ret = klass->pre_push (enc, &buf);
608 if (ret != GST_FLOW_OK || !buf) {
609 GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
610 gst_flow_get_name (ret), buf);
612 gst_buffer_unref (buf);
617 GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
618 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
619 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
620 GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
622 ret = gst_pad_push (enc->srcpad, buf);
623 GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
625 /* merely advance samples, most work for that already done above */
626 priv->samples += samples;
635 GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
636 ("received more encoded samples %d than provided %d",
637 samples, priv->offset / ctx->info.bpf), (NULL));
639 gst_buffer_unref (buf);
640 return GST_FLOW_ERROR;
644 /* adapter tracking idea:
645 * - start of adapter corresponds with what has already been encoded
646 * (i.e. really returned by encoder subclass)
647 * - start + offset is what needs to be fed to subclass next */
649 gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
651 GstAudioEncoderClass *klass;
652 GstAudioEncoderPrivate *priv;
653 GstAudioEncoderContext *ctx;
656 GstFlowReturn ret = GST_FLOW_OK;
658 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
660 g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
663 ctx = &enc->priv->ctx;
665 while (ret == GST_FLOW_OK) {
668 av = gst_adapter_available (priv->adapter);
670 g_assert (priv->offset <= av);
673 need = ctx->frame_samples > 0 ? ctx->frame_samples * ctx->info.bpf : av;
674 GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d",
677 if ((need > av) || !av) {
678 if (G_UNLIKELY (force)) {
688 /* if we have some extra metadata,
689 * provide for integer multiple of frames to allow for better granularity
691 if (ctx->frame_samples > 0 && need) {
692 if (ctx->frame_max > 1)
693 need = need * MIN ((av / need), ctx->frame_max);
694 else if (ctx->frame_max == 0)
695 need = need * (av / need);
699 buf = gst_buffer_new ();
700 GST_BUFFER_DATA (buf) = (guint8 *)
701 gst_adapter_peek (priv->adapter, priv->offset + need) + priv->offset;
702 GST_BUFFER_SIZE (buf) = need;
705 GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
708 /* mark this already as consumed,
709 * which it should be when subclass gives us data in exchange for samples */
710 priv->offset += need;
711 priv->samples_in += need / ctx->info.bpf;
713 priv->got_data = FALSE;
714 ret = klass->handle_frame (enc, buf);
717 gst_buffer_unref (buf);
719 /* no data to feed, no leftover provided, then bail out */
720 if (G_UNLIKELY (!buf && !priv->got_data)) {
721 priv->drained = TRUE;
722 GST_LOG_OBJECT (enc, "no more data drained from subclass");
731 gst_audio_encoder_drain (GstAudioEncoder * enc)
733 if (enc->priv->drained)
736 return gst_audio_encoder_push_buffers (enc, TRUE);
740 gst_audio_encoder_set_base_gp (GstAudioEncoder * enc)
744 if (!enc->priv->granule)
747 /* use running time for granule */
748 /* incoming data is clipped, so a valid input should yield a valid output */
749 ts = gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME,
751 if (GST_CLOCK_TIME_IS_VALID (ts)) {
753 GST_CLOCK_TIME_TO_FRAMES (enc->priv->base_ts, enc->priv->ctx.info.rate);
754 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, enc->priv->base_gp);
756 /* should reasonably have a valid base,
757 * otherwise start at 0 if we did not already start there earlier */
758 if (enc->priv->base_gp < 0) {
759 enc->priv->base_gp = 0;
760 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
767 gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
769 GstAudioEncoder *enc;
770 GstAudioEncoderPrivate *priv;
771 GstAudioEncoderContext *ctx;
772 GstFlowReturn ret = GST_FLOW_OK;
775 enc = GST_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
778 ctx = &enc->priv->ctx;
780 /* should know what is coming by now */
785 "received buffer of size %d with ts %" GST_TIME_FORMAT
786 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
787 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
788 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
790 /* input shoud be whole number of sample frames */
791 if (GST_BUFFER_SIZE (buffer) % ctx->info.bpf)
794 #ifndef GST_DISABLE_GST_DEBUG
796 GstClockTime duration;
797 GstClockTimeDiff diff;
799 /* verify buffer duration */
800 duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND,
801 ctx->info.rate * ctx->info.bpf);
802 diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
803 if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
804 (diff > GST_SECOND / ctx->info.rate / 2 ||
805 diff < -GST_SECOND / ctx->info.rate / 2)) {
806 GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
807 GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
808 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
809 GST_TIME_ARGS (duration));
814 discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
815 if (G_UNLIKELY (discont)) {
816 GST_LOG_OBJECT (buffer, "marked discont");
817 enc->priv->discont = discont;
820 /* clip to segment */
821 /* NOTE: slightly painful linking -laudio only for this one ... */
822 buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->info.rate,
824 if (G_UNLIKELY (!buffer)) {
825 GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
830 "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
831 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
832 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
833 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
835 if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
836 priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
837 GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
838 GST_TIME_ARGS (priv->base_ts));
839 gst_audio_encoder_set_base_gp (enc);
842 /* check for continuity;
843 * checked elsewhere in non-perfect case */
844 if (enc->priv->perfect_ts) {
845 GstClockTimeDiff diff = 0;
846 GstClockTime next_ts = 0;
848 if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
849 GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
852 samples = priv->samples +
853 gst_adapter_available (priv->adapter) / ctx->info.bpf;
854 next_ts = priv->base_ts +
855 gst_util_uint64_scale (samples, GST_SECOND, ctx->info.rate);
856 GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
857 " samples past base_ts %" GST_TIME_FORMAT
858 ", expected ts %" GST_TIME_FORMAT, samples,
859 GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
860 diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
861 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
862 /* if within tolerance,
863 * discard buffer ts and carry on producing perfect stream,
864 * otherwise clip or resync to ts */
865 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
866 diff > enc->priv->tolerance)) {
867 GST_DEBUG_OBJECT (enc, "marked discont");
872 /* do some fancy tweaking in hard resync case */
873 if (discont && enc->priv->hard_resync) {
877 GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
878 GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
881 GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->info.rate) * ctx->info.bpf;
882 if (diff_bytes >= GST_BUFFER_SIZE (buffer)) {
883 gst_buffer_unref (buffer);
886 buffer = gst_buffer_make_metadata_writable (buffer);
887 GST_BUFFER_DATA (buffer) += diff_bytes;
888 GST_BUFFER_SIZE (buffer) -= diff_bytes;
890 GST_BUFFER_TIMESTAMP (buffer) += diff;
891 /* care even less about duration after this */
893 /* drain stuff prior to resync */
894 gst_audio_encoder_drain (enc);
898 priv->base_ts += diff;
899 gst_audio_encoder_set_base_gp (enc);
900 priv->discont |= discont;
903 gst_adapter_push (enc->priv->adapter, buffer);
904 /* new stuff, so we can push subclass again */
905 enc->priv->drained = FALSE;
907 ret = gst_audio_encoder_push_buffers (enc, FALSE);
910 GST_LOG_OBJECT (enc, "chain leaving");
916 GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
917 ("encoder not initialized"));
918 gst_buffer_unref (buffer);
919 return GST_FLOW_NOT_NEGOTIATED;
923 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
924 ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
926 gst_buffer_unref (buffer);
927 return GST_FLOW_ERROR;
932 audio_info_is_equal (GstAudioInfo * from, GstAudioInfo * to)
936 if (GST_AUDIO_INFO_FORMAT (from) != GST_AUDIO_INFO_FORMAT (to))
938 if (GST_AUDIO_INFO_RATE (from) != GST_AUDIO_INFO_RATE (to))
940 if (GST_AUDIO_INFO_CHANNELS (from) != GST_AUDIO_INFO_CHANNELS (to))
942 if (GST_AUDIO_INFO_CHANNELS (from) > 64)
944 return memcmp (from->position, to->position,
945 GST_AUDIO_INFO_CHANNELS (from) * sizeof (to->position[0]));
949 gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
951 GstAudioEncoder *enc;
952 GstAudioEncoderClass *klass;
953 GstAudioEncoderContext *ctx;
954 GstAudioInfo *state, *old_state;
955 gboolean res = TRUE, changed = FALSE;
958 enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
959 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
961 /* subclass must do something here ... */
962 g_return_val_if_fail (klass->set_format != NULL, FALSE);
964 ctx = &enc->priv->ctx;
967 GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
969 if (!gst_caps_is_fixed (caps))
972 /* adjust ts tracking to new sample rate */
973 old_rate = GST_AUDIO_INFO_RATE (state);
974 if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
975 enc->priv->base_ts +=
976 GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
977 enc->priv->samples = 0;
980 old_state = gst_audio_info_copy (state);
981 if (!gst_audio_info_from_caps (state, caps))
984 changed = audio_info_is_equal (state, old_state);
985 gst_audio_info_free (old_state);
988 GstClockTime old_min_latency;
989 GstClockTime old_max_latency;
991 /* drain any pending old data stuff */
992 gst_audio_encoder_drain (enc);
994 /* context defaults */
995 enc->priv->ctx.frame_samples = 0;
996 enc->priv->ctx.frame_max = 0;
997 enc->priv->ctx.lookahead = 0;
999 /* element might report latency */
1000 GST_OBJECT_LOCK (enc);
1001 old_min_latency = ctx->min_latency;
1002 old_max_latency = ctx->max_latency;
1003 GST_OBJECT_UNLOCK (enc);
1005 if (klass->set_format)
1006 res = klass->set_format (enc, state);
1008 /* notify if new latency */
1009 GST_OBJECT_LOCK (enc);
1010 if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
1011 (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
1012 GST_OBJECT_UNLOCK (enc);
1013 /* post latency message on the bus */
1014 gst_element_post_message (GST_ELEMENT (enc),
1015 gst_message_new_latency (GST_OBJECT (enc)));
1016 GST_OBJECT_LOCK (enc);
1018 GST_OBJECT_UNLOCK (enc);
1020 GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
1028 GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
1035 * gst_audio_encoder_proxy_getcaps:
1036 * @enc: a #GstAudioEncoder
1037 * @caps: initial caps
1039 * Returns caps that express @caps (or sink template caps if @caps == NULL)
1040 * restricted to channel/rate combinations supported by downstream elements
1043 * Returns: a #GstCaps owned by caller
1048 gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps)
1050 const GstCaps *templ_caps;
1051 GstCaps *allowed = NULL;
1052 GstCaps *fcaps, *filter_caps;
1055 /* we want to be able to communicate to upstream elements like audioconvert
1056 * and audioresample any rate/channel restrictions downstream (e.g. muxer
1057 * only accepting certain sample rates) */
1058 templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad);
1059 allowed = gst_pad_get_allowed_caps (enc->srcpad);
1060 if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
1061 fcaps = gst_caps_copy (templ_caps);
1065 GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
1066 GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
1068 filter_caps = gst_caps_new_empty ();
1070 for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
1073 q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
1075 /* pick rate + channel fields from allowed caps */
1076 for (j = 0; j < gst_caps_get_size (allowed); j++) {
1077 const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
1081 s = gst_structure_id_empty_new (q_name);
1082 if ((val = gst_structure_get_value (allowed_s, "rate")))
1083 gst_structure_set_value (s, "rate", val);
1084 if ((val = gst_structure_get_value (allowed_s, "channels")))
1085 gst_structure_set_value (s, "channels", val);
1087 gst_caps_merge_structure (filter_caps, s);
1091 fcaps = gst_caps_intersect (filter_caps, templ_caps);
1092 gst_caps_unref (filter_caps);
1095 gst_caps_replace (&allowed, NULL);
1097 GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
1103 gst_audio_encoder_sink_getcaps (GstPad * pad)
1105 GstAudioEncoder *enc;
1106 GstAudioEncoderClass *klass;
1109 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1110 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1111 g_assert (pad == enc->sinkpad);
1114 caps = klass->getcaps (enc);
1116 caps = gst_audio_encoder_proxy_getcaps (enc, NULL);
1117 gst_object_unref (enc);
1119 GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
1125 gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
1127 GstAudioEncoderClass *klass;
1128 gboolean handled = FALSE;
1130 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1132 switch (GST_EVENT_TYPE (event)) {
1133 case GST_EVENT_NEWSEGMENT:
1136 gdouble rate, arate;
1137 gint64 start, stop, time;
1140 gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
1141 &start, &stop, &time);
1143 if (format == GST_FORMAT_TIME) {
1144 GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
1145 " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
1146 ", rate %g, applied_rate %g",
1147 GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
1150 GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
1151 " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
1152 ", rate %g, applied_rate %g", start, stop, time, rate, arate);
1153 GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
1157 /* finish current segment */
1158 gst_audio_encoder_drain (enc);
1159 /* reset partially for new segment */
1160 gst_audio_encoder_reset (enc, FALSE);
1161 /* and follow along with segment */
1162 gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
1163 format, start, stop, time);
1167 case GST_EVENT_FLUSH_START:
1170 case GST_EVENT_FLUSH_STOP:
1171 /* discard any pending stuff */
1172 /* TODO route through drain ?? */
1173 if (!enc->priv->drained && klass->flush)
1175 /* and get (re)set for the sequel */
1176 gst_audio_encoder_reset (enc, FALSE);
1180 gst_audio_encoder_drain (enc);
1191 gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
1193 GstAudioEncoder *enc;
1194 GstAudioEncoderClass *klass;
1195 gboolean handled = FALSE;
1196 gboolean ret = TRUE;
1198 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1199 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1201 GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
1202 GST_EVENT_TYPE_NAME (event));
1205 handled = klass->event (enc, event);
1208 handled = gst_audio_encoder_sink_eventfunc (enc, event);
1211 ret = gst_pad_event_default (pad, event);
1213 GST_DEBUG_OBJECT (enc, "event handled");
1215 gst_object_unref (enc);
1220 gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
1222 gboolean res = TRUE;
1223 GstAudioEncoder *enc;
1225 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1227 switch (GST_QUERY_TYPE (query)) {
1228 case GST_QUERY_FORMATS:
1230 gst_query_set_formats (query, 3,
1231 GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
1235 case GST_QUERY_CONVERT:
1237 GstFormat src_fmt, dest_fmt;
1238 gint64 src_val, dest_val;
1240 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1241 if (!(res = gst_audio_info_convert (&enc->priv->ctx.info,
1242 src_fmt, src_val, dest_fmt, &dest_val)))
1244 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1248 res = gst_pad_query_default (pad, query);
1253 gst_object_unref (enc);
1257 static const GstQueryType *
1258 gst_audio_encoder_get_query_types (GstPad * pad)
1260 static const GstQueryType gst_audio_encoder_src_query_types[] = {
1268 return gst_audio_encoder_src_query_types;
1272 * gst_audio_encoded_audio_convert:
1273 * @fmt: audio format of the encoded audio
1274 * @bytes: number of encoded bytes
1275 * @samples: number of encoded samples
1276 * @src_format: source format
1277 * @src_value: source value
1278 * @dest_format: destination format
1279 * @dest_value: destination format
1281 * Helper function to convert @src_value in @src_format to @dest_value in
1282 * @dest_format for encoded audio data. Conversion is possible between
1283 * BYTE and TIME format by using estimated bitrate based on
1284 * @samples and @bytes (and @fmt).
1288 /* FIXME: make gst_audio_encoded_audio_convert() public? */
1290 gst_audio_encoded_audio_convert (GstAudioInfo * fmt,
1291 gint64 bytes, gint64 samples, GstFormat src_format,
1292 gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
1294 gboolean res = FALSE;
1296 g_return_val_if_fail (dest_format != NULL, FALSE);
1297 g_return_val_if_fail (dest_value != NULL, FALSE);
1299 if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
1302 *dest_value = src_value;
1306 if (samples == 0 || bytes == 0 || fmt->rate == 0) {
1307 GST_DEBUG ("not enough metadata yet to convert");
1313 switch (src_format) {
1314 case GST_FORMAT_BYTES:
1315 switch (*dest_format) {
1316 case GST_FORMAT_TIME:
1317 *dest_value = gst_util_uint64_scale (src_value,
1318 GST_SECOND * samples, bytes);
1325 case GST_FORMAT_TIME:
1326 switch (*dest_format) {
1327 case GST_FORMAT_BYTES:
1328 *dest_value = gst_util_uint64_scale (src_value, bytes,
1329 samples * GST_SECOND);
1344 /* FIXME ? are any of these queries (other than latency) an encoder's business
1345 * also, the conversion stuff might seem to make sense, but seems to not mind
1346 * segment stuff etc at all
1347 * Supposedly that's backward compatibility ... */
1349 gst_audio_encoder_src_query (GstPad * pad, GstQuery * query)
1351 GstAudioEncoder *enc;
1353 gboolean res = FALSE;
1355 enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
1356 peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
1358 GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
1360 switch (GST_QUERY_TYPE (query)) {
1361 case GST_QUERY_POSITION:
1363 GstFormat fmt, req_fmt;
1366 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1367 GST_LOG_OBJECT (enc, "returning peer response");
1372 GST_LOG_OBJECT (enc, "no peer");
1376 gst_query_parse_position (query, &req_fmt, NULL);
1377 fmt = GST_FORMAT_TIME;
1378 if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
1381 if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) {
1382 gst_query_set_position (query, req_fmt, val);
1386 case GST_QUERY_DURATION:
1388 GstFormat fmt, req_fmt;
1391 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1392 GST_LOG_OBJECT (enc, "returning peer response");
1397 GST_LOG_OBJECT (enc, "no peer");
1401 gst_query_parse_duration (query, &req_fmt, NULL);
1402 fmt = GST_FORMAT_TIME;
1403 if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
1406 if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) {
1407 gst_query_set_duration (query, req_fmt, val);
1411 case GST_QUERY_FORMATS:
1413 gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
1417 case GST_QUERY_CONVERT:
1419 GstFormat src_fmt, dest_fmt;
1420 gint64 src_val, dest_val;
1422 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1423 if (!(res = gst_audio_encoded_audio_convert (&enc->priv->ctx.info,
1424 enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
1425 &dest_fmt, &dest_val)))
1427 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1430 case GST_QUERY_LATENCY:
1432 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1434 GstClockTime min_latency, max_latency;
1436 gst_query_parse_latency (query, &live, &min_latency, &max_latency);
1437 GST_DEBUG_OBJECT (enc, "Peer latency: live %d, min %"
1438 GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
1439 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
1441 GST_OBJECT_LOCK (enc);
1442 /* add our latency */
1443 if (min_latency != -1)
1444 min_latency += enc->priv->ctx.min_latency;
1445 if (max_latency != -1)
1446 max_latency += enc->priv->ctx.max_latency;
1447 GST_OBJECT_UNLOCK (enc);
1449 gst_query_set_latency (query, live, min_latency, max_latency);
1454 res = gst_pad_query_default (pad, query);
1458 gst_object_unref (peerpad);
1463 gst_audio_encoder_set_property (GObject * object, guint prop_id,
1464 const GValue * value, GParamSpec * pspec)
1466 GstAudioEncoder *enc;
1468 enc = GST_AUDIO_ENCODER (object);
1471 case PROP_PERFECT_TS:
1472 if (enc->priv->granule && !g_value_get_boolean (value))
1473 GST_WARNING_OBJECT (enc, "perfect-ts can not be set FALSE");
1475 enc->priv->perfect_ts = g_value_get_boolean (value);
1477 case PROP_HARD_RESYNC:
1478 enc->priv->hard_resync = g_value_get_boolean (value);
1480 case PROP_TOLERANCE:
1481 enc->priv->tolerance = g_value_get_int64 (value);
1484 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1490 gst_audio_encoder_get_property (GObject * object, guint prop_id,
1491 GValue * value, GParamSpec * pspec)
1493 GstAudioEncoder *enc;
1495 enc = GST_AUDIO_ENCODER (object);
1498 case PROP_PERFECT_TS:
1499 g_value_set_boolean (value, enc->priv->perfect_ts);
1502 g_value_set_boolean (value, enc->priv->granule);
1504 case PROP_HARD_RESYNC:
1505 g_value_set_boolean (value, enc->priv->hard_resync);
1507 case PROP_TOLERANCE:
1508 g_value_set_int64 (value, enc->priv->tolerance);
1511 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1517 gst_audio_encoder_activate (GstAudioEncoder * enc, gboolean active)
1519 GstAudioEncoderClass *klass;
1520 gboolean result = FALSE;
1522 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1524 g_return_val_if_fail (!enc->priv->granule || enc->priv->perfect_ts, FALSE);
1526 GST_DEBUG_OBJECT (enc, "activate %d", active);
1529 if (!enc->priv->active && klass->start)
1530 result = klass->start (enc);
1532 /* We must make sure streaming has finished before resetting things
1533 * and calling the ::stop vfunc */
1534 GST_PAD_STREAM_LOCK (enc->sinkpad);
1535 GST_PAD_STREAM_UNLOCK (enc->sinkpad);
1537 if (enc->priv->active && klass->stop)
1538 result = klass->stop (enc);
1541 gst_audio_encoder_reset (enc, TRUE);
1543 GST_DEBUG_OBJECT (enc, "activate return: %d", result);
1549 gst_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
1551 gboolean result = TRUE;
1552 GstAudioEncoder *enc;
1554 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1556 GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
1558 result = gst_audio_encoder_activate (enc, active);
1561 enc->priv->active = active;
1563 GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
1565 gst_object_unref (enc);
1570 * gst_audio_encoder_get_audio_info:
1571 * @enc: a #GstAudioEncoder
1573 * Returns: a #GstAudioInfo describing the input audio format
1578 gst_audio_encoder_get_audio_info (GstAudioEncoder * enc)
1580 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), NULL);
1582 return &enc->priv->ctx.info;
1586 * gst_audio_encoder_set_frame_samples:
1587 * @enc: a #GstAudioEncoder
1588 * @num: number of samples per frame
1590 * Sets number of samples (per channel) subclass needs to be handed,
1591 * or will be handed all available if 0.
1596 gst_audio_encoder_set_frame_samples (GstAudioEncoder * enc, gint num)
1598 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1600 enc->priv->ctx.frame_samples = num;
1604 * gst_audio_encoder_get_frame_samples:
1605 * @enc: a #GstAudioEncoder
1607 * Returns: currently requested samples per frame
1612 gst_audio_encoder_get_frame_samples (GstAudioEncoder * enc)
1614 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1616 return enc->priv->ctx.frame_samples;
1620 * gst_audio_encoder_set_frame_max:
1621 * @enc: a #GstAudioEncoder
1622 * @num: number of frames
1624 * Sets max number of frames accepted at once (assumed minimally 1)
1629 gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num)
1631 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1633 enc->priv->ctx.frame_max = num;
1637 * gst_audio_encoder_get_frame_max:
1638 * @enc: a #GstAudioEncoder
1640 * Returns: currently configured maximum handled frames
1645 gst_audio_encoder_get_frame_max (GstAudioEncoder * enc)
1647 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1649 return enc->priv->ctx.frame_max;
1653 * gst_audio_encoder_set_lookahead:
1654 * @enc: a #GstAudioEncoder
1657 * Sets encoder lookahead (in units of input rate samples)
1662 gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num)
1664 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1666 enc->priv->ctx.lookahead = num;
1670 * gst_audio_encoder_get_lookahead:
1671 * @enc: a #GstAudioEncoder
1673 * Returns: currently configured encoder lookahead
1676 gst_audio_encoder_get_lookahead (GstAudioEncoder * enc)
1678 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1680 return enc->priv->ctx.lookahead;
1684 * gst_audio_encoder_set_latency:
1685 * @enc: a #GstAudioEncoder
1686 * @min: minimum latency
1687 * @max: maximum latency
1689 * Sets encoder latency.
1694 gst_audio_encoder_set_latency (GstAudioEncoder * enc,
1695 GstClockTime min, GstClockTime max)
1697 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1699 GST_OBJECT_LOCK (enc);
1700 enc->priv->ctx.min_latency = min;
1701 enc->priv->ctx.max_latency = max;
1702 GST_OBJECT_UNLOCK (enc);
1706 * gst_audio_encoder_get_latency:
1707 * @enc: a #GstAudioEncoder
1708 * @min: a pointer to storage to hold minimum latency
1709 * @max: a pointer to storage to hold maximum latency
1711 * Returns currently configured latency.
1716 gst_audio_encoder_get_latency (GstAudioEncoder * enc,
1717 GstClockTime * min, GstClockTime * max)
1719 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1721 GST_OBJECT_LOCK (enc);
1723 *min = enc->priv->ctx.min_latency;
1725 *max = enc->priv->ctx.max_latency;
1726 GST_OBJECT_UNLOCK (enc);
1730 * gst_audio_encoder_set_mark_granule:
1731 * @enc: a #GstAudioEncoder
1732 * @enabled: new state
1734 * Enable or disable encoder granule handling.
1741 gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc, gboolean enabled)
1743 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1745 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1747 GST_OBJECT_LOCK (enc);
1748 enc->priv->granule = enabled;
1749 GST_OBJECT_UNLOCK (enc);
1753 * gst_audio_encoder_get_mark_granule:
1754 * @enc: a #GstAudioEncoder
1756 * Queries if the encoder will handle granule marking.
1758 * Returns: TRUE if granule marking is enabled.
1765 gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc)
1769 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1771 GST_OBJECT_LOCK (enc);
1772 result = enc->priv->granule;
1773 GST_OBJECT_UNLOCK (enc);
1779 * gst_audio_encoder_set_perfect_timestamp:
1780 * @enc: a #GstAudioEncoder
1781 * @enabled: new state
1783 * Enable or disable encoder perfect output timestamp preference.
1790 gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
1793 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1795 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1797 GST_OBJECT_LOCK (enc);
1798 enc->priv->perfect_ts = enabled;
1799 GST_OBJECT_UNLOCK (enc);
1803 * gst_audio_encoder_get_perfect_timestamp:
1804 * @enc: a #GstAudioEncoder
1806 * Queries encoder perfect timestamp behaviour.
1808 * Returns: TRUE if pefect timestamp setting enabled.
1815 gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc)
1819 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1821 GST_OBJECT_LOCK (enc);
1822 result = enc->priv->perfect_ts;
1823 GST_OBJECT_UNLOCK (enc);
1829 * gst_audio_encoder_set_hard_sync:
1830 * @enc: a #GstAudioEncoder
1831 * @enabled: new state
1833 * Sets encoder hard resync handling.
1840 gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc, gboolean enabled)
1842 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1844 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1846 GST_OBJECT_LOCK (enc);
1847 enc->priv->hard_resync = enabled;
1848 GST_OBJECT_UNLOCK (enc);
1852 * gst_audio_encoder_get_hard_sync:
1853 * @enc: a #GstAudioEncoder
1855 * Queries encoder's hard resync setting.
1857 * Returns: TRUE if hard resync is enabled.
1864 gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc)
1868 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1870 GST_OBJECT_LOCK (enc);
1871 result = enc->priv->hard_resync;
1872 GST_OBJECT_UNLOCK (enc);
1878 * gst_audio_encoder_set_tolerance:
1879 * @enc: a #GstAudioEncoder
1880 * @tolerance: new tolerance
1882 * Configures encoder audio jitter tolerance threshold.
1889 gst_audio_encoder_set_tolerance (GstAudioEncoder * enc, gint64 tolerance)
1891 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1893 GST_OBJECT_LOCK (enc);
1894 enc->priv->tolerance = tolerance;
1895 GST_OBJECT_UNLOCK (enc);
1899 * gst_audio_encoder_get_tolerance:
1900 * @enc: a #GstAudioEncoder
1902 * Queries current audio jitter tolerance threshold.
1904 * Returns: encoder audio jitter tolerance threshold.
1911 gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)
1915 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1917 GST_OBJECT_LOCK (enc);
1918 result = enc->priv->tolerance;
1919 GST_OBJECT_UNLOCK (enc);