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-timestamp.
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-timestamp case, an upstream variation
112 * exceeding 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-timestamp case, this one
115 * optionally (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 #include "gstaudioencoder.h"
155 #include <gst/base/gstadapter.h>
156 #include <gst/audio/audio.h>
162 GST_DEBUG_CATEGORY_STATIC (gst_audio_encoder_debug);
163 #define GST_CAT_DEFAULT gst_audio_encoder_debug
165 #define GST_AUDIO_ENCODER_GET_PRIVATE(obj) \
166 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_ENCODER, \
167 GstAudioEncoderPrivate))
178 #define DEFAULT_PERFECT_TS FALSE
179 #define DEFAULT_GRANULE FALSE
180 #define DEFAULT_HARD_RESYNC FALSE
181 #define DEFAULT_TOLERANCE 40000000
183 typedef struct _GstAudioEncoderContext
192 /* MT-protected (with LOCK) */
193 GstClockTime min_latency;
194 GstClockTime max_latency;
195 } GstAudioEncoderContext;
197 struct _GstAudioEncoderPrivate
199 /* activation status */
202 /* input base/first ts as basis for output ts;
203 * kept nearly constant for perfect_ts,
204 * otherwise resyncs to upstream ts */
205 GstClockTime base_ts;
206 /* corresponding base granulepos */
208 /* input samples processed and sent downstream so far (w.r.t. base_ts) */
211 /* currently collected sample data */
213 /* offset in adapter up to which already supplied to encoder */
215 /* mark outgoing discont */
217 /* to guess duration of drained data */
218 GstClockTime last_duration;
220 /* subclass provided data in processing round */
222 /* subclass gave all it could already */
224 /* subclass currently being forcibly drained */
227 /* output bps estimatation */
228 /* global in samples seen */
230 /* global bytes sent out */
233 /* context storage */
234 GstAudioEncoderContext ctx;
239 gboolean hard_resync;
244 static GstElementClass *parent_class = NULL;
246 static void gst_audio_encoder_class_init (GstAudioEncoderClass * klass);
247 static void gst_audio_encoder_init (GstAudioEncoder * parse,
248 GstAudioEncoderClass * klass);
251 gst_audio_encoder_get_type (void)
253 static GType audio_encoder_type = 0;
255 if (!audio_encoder_type) {
256 static const GTypeInfo audio_encoder_info = {
257 sizeof (GstAudioEncoderClass),
258 (GBaseInitFunc) NULL,
259 (GBaseFinalizeFunc) NULL,
260 (GClassInitFunc) gst_audio_encoder_class_init,
263 sizeof (GstAudioEncoder),
265 (GInstanceInitFunc) gst_audio_encoder_init,
267 const GInterfaceInfo preset_interface_info = {
268 NULL, /* interface_init */
269 NULL, /* interface_finalize */
270 NULL /* interface_data */
273 audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
274 "GstAudioEncoder", &audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
276 g_type_add_interface_static (audio_encoder_type, GST_TYPE_PRESET,
277 &preset_interface_info);
279 return audio_encoder_type;
282 static void gst_audio_encoder_finalize (GObject * object);
283 static void gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full);
285 static void gst_audio_encoder_set_property (GObject * object,
286 guint prop_id, const GValue * value, GParamSpec * pspec);
287 static void gst_audio_encoder_get_property (GObject * object,
288 guint prop_id, GValue * value, GParamSpec * pspec);
290 static gboolean gst_audio_encoder_sink_activate_push (GstPad * pad,
293 static gboolean gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event);
294 static gboolean gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc,
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, GstCaps * filter);
303 gst_audio_encoder_class_init (GstAudioEncoderClass * klass)
305 GObjectClass *gobject_class;
307 gobject_class = G_OBJECT_CLASS (klass);
308 parent_class = g_type_class_peek_parent (klass);
310 GST_DEBUG_CATEGORY_INIT (gst_audio_encoder_debug, "audioencoder", 0,
311 "audio encoder base class");
313 g_type_class_add_private (klass, sizeof (GstAudioEncoderPrivate));
315 gobject_class->set_property = gst_audio_encoder_set_property;
316 gobject_class->get_property = gst_audio_encoder_get_property;
318 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_encoder_finalize);
321 g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
322 g_param_spec_boolean ("perfect-timestamp", "Perfect Timestamps",
323 "Favour perfect timestamps over tracking upstream timestamps",
324 DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
325 g_object_class_install_property (gobject_class, PROP_GRANULE,
326 g_param_spec_boolean ("mark-granule", "Granule Marking",
327 "Apply granule semantics to buffer metadata (implies perfect-timestamp)",
328 DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
329 g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
330 g_param_spec_boolean ("hard-resync", "Hard Resync",
331 "Perform clipping and sample flushing upon discontinuity",
332 DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
333 g_object_class_install_property (gobject_class, PROP_TOLERANCE,
334 g_param_spec_int64 ("tolerance", "Tolerance",
335 "Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
336 0, G_MAXINT64, DEFAULT_TOLERANCE,
337 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
341 gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
343 GstPadTemplate *pad_template;
345 GST_DEBUG_OBJECT (enc, "gst_audio_encoder_init");
347 enc->priv = GST_AUDIO_ENCODER_GET_PRIVATE (enc);
349 /* only push mode supported */
351 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
352 g_return_if_fail (pad_template != NULL);
353 enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
354 gst_pad_set_event_function (enc->sinkpad,
355 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_event));
356 gst_pad_set_getcaps_function (enc->sinkpad,
357 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_getcaps));
358 gst_pad_set_query_function (enc->sinkpad,
359 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_query));
360 gst_pad_set_chain_function (enc->sinkpad,
361 GST_DEBUG_FUNCPTR (gst_audio_encoder_chain));
362 gst_pad_set_activatepush_function (enc->sinkpad,
363 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_activate_push));
364 gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
366 GST_DEBUG_OBJECT (enc, "sinkpad created");
368 /* and we don't mind upstream traveling stuff that much ... */
370 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
371 g_return_if_fail (pad_template != NULL);
372 enc->srcpad = gst_pad_new_from_template (pad_template, "src");
373 gst_pad_set_query_function (enc->srcpad,
374 GST_DEBUG_FUNCPTR (gst_audio_encoder_src_query));
375 gst_pad_set_query_type_function (enc->srcpad,
376 GST_DEBUG_FUNCPTR (gst_audio_encoder_get_query_types));
377 gst_pad_use_fixed_caps (enc->srcpad);
378 gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
379 GST_DEBUG_OBJECT (enc, "src created");
381 enc->priv->adapter = gst_adapter_new ();
383 /* property default */
384 enc->priv->granule = DEFAULT_GRANULE;
385 enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
386 enc->priv->hard_resync = DEFAULT_HARD_RESYNC;
387 enc->priv->tolerance = DEFAULT_TOLERANCE;
390 gst_audio_encoder_reset (enc, TRUE);
391 GST_DEBUG_OBJECT (enc, "init ok");
395 gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
397 GST_OBJECT_LOCK (enc);
400 enc->priv->active = FALSE;
401 enc->priv->samples_in = 0;
402 enc->priv->bytes_out = 0;
403 gst_audio_info_init (&enc->priv->ctx.info);
404 memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
407 gst_segment_init (&enc->segment, GST_FORMAT_TIME);
409 gst_adapter_clear (enc->priv->adapter);
410 enc->priv->got_data = FALSE;
411 enc->priv->drained = TRUE;
412 enc->priv->offset = 0;
413 enc->priv->base_ts = GST_CLOCK_TIME_NONE;
414 enc->priv->base_gp = -1;
415 enc->priv->samples = 0;
416 enc->priv->discont = FALSE;
418 GST_OBJECT_UNLOCK (enc);
422 gst_audio_encoder_finalize (GObject * object)
424 GstAudioEncoder *enc = GST_AUDIO_ENCODER (object);
426 g_object_unref (enc->priv->adapter);
428 G_OBJECT_CLASS (parent_class)->finalize (object);
432 * gst_audio_encoder_finish_frame:
433 * @enc: a #GstAudioEncoder
434 * @buffer: encoded data
435 * @samples: number of samples (per channel) represented by encoded data
437 * Collects encoded data and/or pushes encoded data downstream.
438 * Source pad caps must be set when this is called. Depending on the nature
439 * of the (framing of) the format, subclass can decide whether to push
440 * encoded data directly or to collect various "frames" in a single buffer.
441 * Note that the latter behaviour is recommended whenever the format is allowed,
442 * as it incurs no additional latency and avoids otherwise generating a
443 * a multitude of (small) output buffers. If not explicitly pushed,
444 * any available encoded data is pushed at the end of each processing cycle,
445 * i.e. which encodes as much data as available input data allows.
447 * If @samples < 0, then best estimate is all samples provided to encoder
448 * (subclass) so far. @buf may be NULL, in which case next number of @samples
449 * are considered discarded, e.g. as a result of discontinuous transmission,
450 * and a discontinuity is marked (note that @buf == NULL => push == TRUE).
452 * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
457 gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
460 GstAudioEncoderClass *klass;
461 GstAudioEncoderPrivate *priv;
462 GstAudioEncoderContext *ctx;
463 GstFlowReturn ret = GST_FLOW_OK;
465 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
467 ctx = &enc->priv->ctx;
469 /* subclass should know what it is producing by now */
470 g_return_val_if_fail (gst_pad_has_current_caps (enc->srcpad), GST_FLOW_ERROR);
471 /* subclass should not hand us no data */
472 g_return_val_if_fail (buf == NULL || gst_buffer_get_size (buf) > 0,
475 GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
476 buf ? gst_buffer_get_size (buf) : -1, samples);
478 /* mark subclass still alive and providing */
479 priv->got_data = TRUE;
481 /* remove corresponding samples from input */
483 samples = (enc->priv->offset / ctx->info.bpf);
485 if (G_LIKELY (samples)) {
486 /* track upstream ts if so configured */
487 if (!enc->priv->perfect_ts) {
488 guint64 ts, distance;
490 ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
491 g_assert (distance % ctx->info.bpf == 0);
492 distance /= ctx->info.bpf;
493 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
494 GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
495 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
496 GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
497 /* when draining adapter might be empty and no ts to offer */
498 if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
499 GstClockTimeDiff diff;
500 GstClockTime old_ts, next_ts;
502 /* passed into another buffer;
503 * mild check for discontinuity and only mark if so */
505 gst_util_uint64_scale (distance, GST_SECOND, ctx->info.rate);
506 old_ts = priv->base_ts +
507 gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->info.rate);
508 diff = GST_CLOCK_DIFF (next_ts, old_ts);
509 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
510 /* only mark discontinuity if beyond tolerance */
511 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
512 diff > enc->priv->tolerance)) {
513 GST_DEBUG_OBJECT (enc, "marked discont");
514 priv->discont = TRUE;
516 if (diff > GST_SECOND / ctx->info.rate / 2 ||
517 diff < -GST_SECOND / ctx->info.rate / 2) {
518 GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
519 " at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
520 /* re-sync to upstream ts */
522 priv->samples = distance;
524 GST_LOG_OBJECT (enc, "new upstream ts only introduces jitter");
528 /* advance sample view */
529 if (G_UNLIKELY (samples * ctx->info.bpf > priv->offset)) {
530 if (G_LIKELY (!priv->force)) {
531 /* no way we can let this pass */
532 g_assert_not_reached ();
537 if (samples * ctx->info.bpf >= gst_adapter_available (priv->adapter))
538 gst_adapter_clear (priv->adapter);
540 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
543 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
544 priv->offset -= samples * ctx->info.bpf;
545 /* avoid subsequent stray prev_ts */
546 if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
547 gst_adapter_clear (priv->adapter);
549 /* sample count advanced below after buffer handling */
553 if (G_LIKELY (buf)) {
556 size = gst_buffer_get_size (buf);
558 GST_LOG_OBJECT (enc, "taking %d bytes for output", size);
559 buf = gst_buffer_make_writable (buf);
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 + size;
597 priv->bytes_out += size;
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, size,
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);
701 data = gst_adapter_map (priv->adapter, priv->offset + need);
703 gst_buffer_new_wrapped_full ((gpointer) data, NULL, priv->offset,
707 GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
710 /* mark this already as consumed,
711 * which it should be when subclass gives us data in exchange for samples */
712 priv->offset += need;
713 priv->samples_in += need / ctx->info.bpf;
715 priv->got_data = FALSE;
716 ret = klass->handle_frame (enc, buf);
718 if (G_LIKELY (buf)) {
719 gst_buffer_unref (buf);
720 gst_adapter_unmap (priv->adapter, 0);
723 /* no data to feed, no leftover provided, then bail out */
724 if (G_UNLIKELY (!buf && !priv->got_data)) {
725 priv->drained = TRUE;
726 GST_LOG_OBJECT (enc, "no more data drained from subclass");
735 gst_audio_encoder_drain (GstAudioEncoder * enc)
737 if (enc->priv->drained)
740 return gst_audio_encoder_push_buffers (enc, TRUE);
744 gst_audio_encoder_set_base_gp (GstAudioEncoder * enc)
748 if (!enc->priv->granule)
751 /* use running time for granule */
752 /* incoming data is clipped, so a valid input should yield a valid output */
753 ts = gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME,
755 if (GST_CLOCK_TIME_IS_VALID (ts)) {
757 GST_CLOCK_TIME_TO_FRAMES (enc->priv->base_ts, enc->priv->ctx.info.rate);
758 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, enc->priv->base_gp);
760 /* should reasonably have a valid base,
761 * otherwise start at 0 if we did not already start there earlier */
762 if (enc->priv->base_gp < 0) {
763 enc->priv->base_gp = 0;
764 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
771 gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
773 GstAudioEncoder *enc;
774 GstAudioEncoderPrivate *priv;
775 GstAudioEncoderContext *ctx;
776 GstFlowReturn ret = GST_FLOW_OK;
780 enc = GST_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
783 ctx = &enc->priv->ctx;
785 /* should know what is coming by now */
789 size = gst_buffer_get_size (buffer);
792 "received buffer of size %d with ts %" GST_TIME_FORMAT
793 ", duration %" GST_TIME_FORMAT, size,
794 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
795 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
797 /* input shoud be whole number of sample frames */
798 if (size % ctx->info.bpf)
801 #ifndef GST_DISABLE_GST_DEBUG
803 GstClockTime duration;
804 GstClockTimeDiff diff;
806 /* verify buffer duration */
807 duration = gst_util_uint64_scale (size, GST_SECOND,
808 ctx->info.rate * ctx->info.bpf);
809 diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
810 if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
811 (diff > GST_SECOND / ctx->info.rate / 2 ||
812 diff < -GST_SECOND / ctx->info.rate / 2)) {
813 GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
814 GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
815 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
816 GST_TIME_ARGS (duration));
821 discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
822 if (G_UNLIKELY (discont)) {
823 GST_LOG_OBJECT (buffer, "marked discont");
824 enc->priv->discont = discont;
827 /* clip to segment */
828 /* NOTE: slightly painful linking -laudio only for this one ... */
829 buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->info.rate,
831 if (G_UNLIKELY (!buffer)) {
832 GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
836 size = gst_buffer_get_size (buffer);
839 "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
840 ", duration %" GST_TIME_FORMAT, size,
841 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
842 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
844 if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
845 priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
846 GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
847 GST_TIME_ARGS (priv->base_ts));
848 gst_audio_encoder_set_base_gp (enc);
851 /* check for continuity;
852 * checked elsewhere in non-perfect case */
853 if (enc->priv->perfect_ts) {
854 GstClockTimeDiff diff = 0;
855 GstClockTime next_ts = 0;
857 if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
858 GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
861 samples = priv->samples +
862 gst_adapter_available (priv->adapter) / ctx->info.bpf;
863 next_ts = priv->base_ts +
864 gst_util_uint64_scale (samples, GST_SECOND, ctx->info.rate);
865 GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
866 " samples past base_ts %" GST_TIME_FORMAT
867 ", expected ts %" GST_TIME_FORMAT, samples,
868 GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
869 diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
870 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
871 /* if within tolerance,
872 * discard buffer ts and carry on producing perfect stream,
873 * otherwise clip or resync to ts */
874 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
875 diff > enc->priv->tolerance)) {
876 GST_DEBUG_OBJECT (enc, "marked discont");
881 /* do some fancy tweaking in hard resync case */
882 if (discont && enc->priv->hard_resync) {
886 GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
887 GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
890 GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->info.rate) * ctx->info.bpf;
891 if (diff_bytes >= size) {
892 gst_buffer_unref (buffer);
895 buffer = gst_buffer_make_writable (buffer);
896 gst_buffer_resize (buffer, diff_bytes, size - diff_bytes);
898 GST_BUFFER_TIMESTAMP (buffer) += diff;
899 /* care even less about duration after this */
901 /* drain stuff prior to resync */
902 gst_audio_encoder_drain (enc);
906 priv->base_ts += diff;
907 gst_audio_encoder_set_base_gp (enc);
908 priv->discont |= discont;
911 gst_adapter_push (enc->priv->adapter, buffer);
912 /* new stuff, so we can push subclass again */
913 enc->priv->drained = FALSE;
915 ret = gst_audio_encoder_push_buffers (enc, FALSE);
918 GST_LOG_OBJECT (enc, "chain leaving");
924 GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
925 ("encoder not initialized"));
926 gst_buffer_unref (buffer);
927 return GST_FLOW_NOT_NEGOTIATED;
931 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
932 ("buffer size %d not a multiple of %d", gst_buffer_get_size (buffer),
934 gst_buffer_unref (buffer);
935 return GST_FLOW_ERROR;
940 audio_info_is_equal (GstAudioInfo * from, GstAudioInfo * to)
944 if (from->finfo == NULL || to->finfo == NULL)
946 if (GST_AUDIO_INFO_FORMAT (from) != GST_AUDIO_INFO_FORMAT (to))
948 if (GST_AUDIO_INFO_RATE (from) != GST_AUDIO_INFO_RATE (to))
950 if (GST_AUDIO_INFO_CHANNELS (from) != GST_AUDIO_INFO_CHANNELS (to))
952 if (GST_AUDIO_INFO_CHANNELS (from) > 64)
954 return memcmp (from->position, to->position,
955 GST_AUDIO_INFO_CHANNELS (from) * sizeof (to->position[0]));
959 gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
961 GstAudioEncoderClass *klass;
962 GstAudioEncoderContext *ctx;
964 gboolean res = TRUE, changed = FALSE;
967 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
969 /* subclass must do something here ... */
970 g_return_val_if_fail (klass->set_format != NULL, FALSE);
972 ctx = &enc->priv->ctx;
974 GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
976 if (!gst_caps_is_fixed (caps))
979 /* adjust ts tracking to new sample rate */
980 old_rate = GST_AUDIO_INFO_RATE (&ctx->info);
981 if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
982 enc->priv->base_ts +=
983 GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
984 enc->priv->samples = 0;
987 if (!gst_audio_info_from_caps (&state, caps))
990 changed = !audio_info_is_equal (&state, &ctx->info);
993 GstClockTime old_min_latency;
994 GstClockTime old_max_latency;
996 /* drain any pending old data stuff */
997 gst_audio_encoder_drain (enc);
999 /* context defaults */
1000 enc->priv->ctx.frame_samples = 0;
1001 enc->priv->ctx.frame_max = 0;
1002 enc->priv->ctx.lookahead = 0;
1004 /* element might report latency */
1005 GST_OBJECT_LOCK (enc);
1006 old_min_latency = ctx->min_latency;
1007 old_max_latency = ctx->max_latency;
1008 GST_OBJECT_UNLOCK (enc);
1010 if (klass->set_format)
1011 res = klass->set_format (enc, &state);
1013 /* notify if new latency */
1014 GST_OBJECT_LOCK (enc);
1015 if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
1016 (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
1017 GST_OBJECT_UNLOCK (enc);
1018 /* post latency message on the bus */
1019 gst_element_post_message (GST_ELEMENT (enc),
1020 gst_message_new_latency (GST_OBJECT (enc)));
1021 GST_OBJECT_LOCK (enc);
1023 GST_OBJECT_UNLOCK (enc);
1025 GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
1033 GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
1040 * gst_audio_encoder_proxy_getcaps:
1041 * @enc: a #GstAudioEncoder
1042 * @caps: initial caps
1044 * Returns caps that express @caps (or sink template caps if @caps == NULL)
1045 * restricted to channel/rate combinations supported by downstream elements
1048 * Returns: a #GstCaps owned by caller
1053 gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps)
1055 const GstCaps *templ_caps;
1056 GstCaps *allowed = NULL;
1057 GstCaps *fcaps, *filter_caps;
1060 /* we want to be able to communicate to upstream elements like audioconvert
1061 * and audioresample any rate/channel restrictions downstream (e.g. muxer
1062 * only accepting certain sample rates) */
1063 templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad);
1064 allowed = gst_pad_get_allowed_caps (enc->srcpad);
1065 if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
1066 fcaps = gst_caps_copy (templ_caps);
1070 GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
1071 GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
1073 filter_caps = gst_caps_new_empty ();
1075 for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
1078 q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
1080 /* pick rate + channel fields from allowed caps */
1081 for (j = 0; j < gst_caps_get_size (allowed); j++) {
1082 const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
1086 s = gst_structure_id_empty_new (q_name);
1087 if ((val = gst_structure_get_value (allowed_s, "rate")))
1088 gst_structure_set_value (s, "rate", val);
1089 if ((val = gst_structure_get_value (allowed_s, "channels")))
1090 gst_structure_set_value (s, "channels", val);
1091 /* following might also make sense for some encoded formats,
1093 if ((val = gst_structure_get_value (allowed_s, "width")))
1094 gst_structure_set_value (s, "width", val);
1095 if ((val = gst_structure_get_value (allowed_s, "depth")))
1096 gst_structure_set_value (s, "depth", val);
1097 if ((val = gst_structure_get_value (allowed_s, "endianness")))
1098 gst_structure_set_value (s, "endianness", val);
1099 if ((val = gst_structure_get_value (allowed_s, "signed")))
1100 gst_structure_set_value (s, "signed", val);
1101 if ((val = gst_structure_get_value (allowed_s, "channel-positions")))
1102 gst_structure_set_value (s, "channel-positions", val);
1104 gst_caps_merge_structure (filter_caps, s);
1108 fcaps = gst_caps_intersect (filter_caps, templ_caps);
1109 gst_caps_unref (filter_caps);
1112 gst_caps_replace (&allowed, NULL);
1114 GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
1120 gst_audio_encoder_sink_getcaps (GstPad * pad, GstCaps * filter)
1122 GstAudioEncoder *enc;
1123 GstAudioEncoderClass *klass;
1126 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1127 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1128 g_assert (pad == enc->sinkpad);
1131 caps = klass->getcaps (enc, filter);
1133 caps = gst_audio_encoder_proxy_getcaps (enc, NULL);
1134 gst_object_unref (enc);
1136 GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
1142 gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
1144 GstAudioEncoderClass *klass;
1145 gboolean handled = FALSE;
1147 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1149 switch (GST_EVENT_TYPE (event)) {
1150 case GST_EVENT_SEGMENT:
1154 gst_event_copy_segment (event, &seg);
1156 if (seg.format == GST_FORMAT_TIME) {
1157 GST_DEBUG_OBJECT (enc, "received TIME SEGMENT %" GST_PTR_FORMAT, &seg);
1159 GST_DEBUG_OBJECT (enc, "received SEGMENT %" GST_PTR_FORMAT, &seg);
1160 GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
1164 /* finish current segment */
1165 gst_audio_encoder_drain (enc);
1166 /* reset partially for new segment */
1167 gst_audio_encoder_reset (enc, FALSE);
1168 /* and follow along with segment */
1173 case GST_EVENT_FLUSH_START:
1176 case GST_EVENT_FLUSH_STOP:
1177 /* discard any pending stuff */
1178 /* TODO route through drain ?? */
1179 if (!enc->priv->drained && klass->flush)
1181 /* and get (re)set for the sequel */
1182 gst_audio_encoder_reset (enc, FALSE);
1186 gst_audio_encoder_drain (enc);
1189 case GST_EVENT_CAPS:
1193 gst_event_parse_caps (event, &caps);
1194 gst_audio_encoder_sink_setcaps (enc, caps);
1195 gst_event_unref (event);
1208 gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
1210 GstAudioEncoder *enc;
1211 GstAudioEncoderClass *klass;
1212 gboolean handled = FALSE;
1213 gboolean ret = TRUE;
1215 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1216 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1218 GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
1219 GST_EVENT_TYPE_NAME (event));
1222 handled = klass->event (enc, event);
1225 handled = gst_audio_encoder_sink_eventfunc (enc, event);
1228 ret = gst_pad_event_default (pad, event);
1230 GST_DEBUG_OBJECT (enc, "event handled");
1232 gst_object_unref (enc);
1237 gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
1239 gboolean res = TRUE;
1240 GstAudioEncoder *enc;
1242 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1244 switch (GST_QUERY_TYPE (query)) {
1245 case GST_QUERY_FORMATS:
1247 gst_query_set_formats (query, 3,
1248 GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
1252 case GST_QUERY_CONVERT:
1254 GstFormat src_fmt, dest_fmt;
1255 gint64 src_val, dest_val;
1257 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1258 if (!(res = gst_audio_info_convert (&enc->priv->ctx.info,
1259 src_fmt, src_val, dest_fmt, &dest_val)))
1261 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1265 res = gst_pad_query_default (pad, query);
1270 gst_object_unref (enc);
1274 static const GstQueryType *
1275 gst_audio_encoder_get_query_types (GstPad * pad)
1277 static const GstQueryType gst_audio_encoder_src_query_types[] = {
1285 return gst_audio_encoder_src_query_types;
1289 * gst_audio_encoded_audio_convert:
1290 * @fmt: audio format of the encoded audio
1291 * @bytes: number of encoded bytes
1292 * @samples: number of encoded samples
1293 * @src_format: source format
1294 * @src_value: source value
1295 * @dest_format: destination format
1296 * @dest_value: destination format
1298 * Helper function to convert @src_value in @src_format to @dest_value in
1299 * @dest_format for encoded audio data. Conversion is possible between
1300 * BYTE and TIME format by using estimated bitrate based on
1301 * @samples and @bytes (and @fmt).
1305 /* FIXME: make gst_audio_encoded_audio_convert() public? */
1307 gst_audio_encoded_audio_convert (GstAudioInfo * fmt,
1308 gint64 bytes, gint64 samples, GstFormat src_format,
1309 gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
1311 gboolean res = FALSE;
1313 g_return_val_if_fail (dest_format != NULL, FALSE);
1314 g_return_val_if_fail (dest_value != NULL, FALSE);
1316 if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
1319 *dest_value = src_value;
1323 if (samples == 0 || bytes == 0 || fmt->rate == 0) {
1324 GST_DEBUG ("not enough metadata yet to convert");
1330 switch (src_format) {
1331 case GST_FORMAT_BYTES:
1332 switch (*dest_format) {
1333 case GST_FORMAT_TIME:
1334 *dest_value = gst_util_uint64_scale (src_value,
1335 GST_SECOND * samples, bytes);
1342 case GST_FORMAT_TIME:
1343 switch (*dest_format) {
1344 case GST_FORMAT_BYTES:
1345 *dest_value = gst_util_uint64_scale (src_value, bytes,
1346 samples * GST_SECOND);
1361 /* FIXME ? are any of these queries (other than latency) an encoder's business
1362 * also, the conversion stuff might seem to make sense, but seems to not mind
1363 * segment stuff etc at all
1364 * Supposedly that's backward compatibility ... */
1366 gst_audio_encoder_src_query (GstPad * pad, GstQuery * query)
1368 GstAudioEncoder *enc;
1370 gboolean res = FALSE;
1372 enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
1373 peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
1375 GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
1377 switch (GST_QUERY_TYPE (query)) {
1378 case GST_QUERY_POSITION:
1380 GstFormat fmt, req_fmt;
1383 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1384 GST_LOG_OBJECT (enc, "returning peer response");
1389 GST_LOG_OBJECT (enc, "no peer");
1393 gst_query_parse_position (query, &req_fmt, NULL);
1394 fmt = GST_FORMAT_TIME;
1395 if (!(res = gst_pad_query_position (peerpad, fmt, &pos)))
1398 if ((res = gst_pad_query_convert (peerpad, fmt, pos, req_fmt, &val))) {
1399 gst_query_set_position (query, req_fmt, val);
1403 case GST_QUERY_DURATION:
1405 GstFormat fmt, req_fmt;
1408 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1409 GST_LOG_OBJECT (enc, "returning peer response");
1414 GST_LOG_OBJECT (enc, "no peer");
1418 gst_query_parse_duration (query, &req_fmt, NULL);
1419 fmt = GST_FORMAT_TIME;
1420 if (!(res = gst_pad_query_duration (peerpad, fmt, &dur)))
1423 if ((res = gst_pad_query_convert (peerpad, fmt, dur, req_fmt, &val))) {
1424 gst_query_set_duration (query, req_fmt, val);
1428 case GST_QUERY_FORMATS:
1430 gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
1434 case GST_QUERY_CONVERT:
1436 GstFormat src_fmt, dest_fmt;
1437 gint64 src_val, dest_val;
1439 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1440 if (!(res = gst_audio_encoded_audio_convert (&enc->priv->ctx.info,
1441 enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
1442 &dest_fmt, &dest_val)))
1444 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1447 case GST_QUERY_LATENCY:
1449 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1451 GstClockTime min_latency, max_latency;
1453 gst_query_parse_latency (query, &live, &min_latency, &max_latency);
1454 GST_DEBUG_OBJECT (enc, "Peer latency: live %d, min %"
1455 GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
1456 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
1458 GST_OBJECT_LOCK (enc);
1459 /* add our latency */
1460 if (min_latency != -1)
1461 min_latency += enc->priv->ctx.min_latency;
1462 if (max_latency != -1)
1463 max_latency += enc->priv->ctx.max_latency;
1464 GST_OBJECT_UNLOCK (enc);
1466 gst_query_set_latency (query, live, min_latency, max_latency);
1471 res = gst_pad_query_default (pad, query);
1475 gst_object_unref (peerpad);
1480 gst_audio_encoder_set_property (GObject * object, guint prop_id,
1481 const GValue * value, GParamSpec * pspec)
1483 GstAudioEncoder *enc;
1485 enc = GST_AUDIO_ENCODER (object);
1488 case PROP_PERFECT_TS:
1489 if (enc->priv->granule && !g_value_get_boolean (value))
1490 GST_WARNING_OBJECT (enc, "perfect-timestamp can not be set FALSE "
1491 "while granule handling is enabled");
1493 enc->priv->perfect_ts = g_value_get_boolean (value);
1495 case PROP_HARD_RESYNC:
1496 enc->priv->hard_resync = g_value_get_boolean (value);
1498 case PROP_TOLERANCE:
1499 enc->priv->tolerance = g_value_get_int64 (value);
1502 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1508 gst_audio_encoder_get_property (GObject * object, guint prop_id,
1509 GValue * value, GParamSpec * pspec)
1511 GstAudioEncoder *enc;
1513 enc = GST_AUDIO_ENCODER (object);
1516 case PROP_PERFECT_TS:
1517 g_value_set_boolean (value, enc->priv->perfect_ts);
1520 g_value_set_boolean (value, enc->priv->granule);
1522 case PROP_HARD_RESYNC:
1523 g_value_set_boolean (value, enc->priv->hard_resync);
1525 case PROP_TOLERANCE:
1526 g_value_set_int64 (value, enc->priv->tolerance);
1529 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1535 gst_audio_encoder_activate (GstAudioEncoder * enc, gboolean active)
1537 GstAudioEncoderClass *klass;
1538 gboolean result = FALSE;
1540 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1542 g_return_val_if_fail (!enc->priv->granule || enc->priv->perfect_ts, FALSE);
1544 GST_DEBUG_OBJECT (enc, "activate %d", active);
1547 if (!enc->priv->active && klass->start)
1548 result = klass->start (enc);
1550 /* We must make sure streaming has finished before resetting things
1551 * and calling the ::stop vfunc */
1552 GST_PAD_STREAM_LOCK (enc->sinkpad);
1553 GST_PAD_STREAM_UNLOCK (enc->sinkpad);
1555 if (enc->priv->active && klass->stop)
1556 result = klass->stop (enc);
1559 gst_audio_encoder_reset (enc, TRUE);
1561 GST_DEBUG_OBJECT (enc, "activate return: %d", result);
1567 gst_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
1569 gboolean result = TRUE;
1570 GstAudioEncoder *enc;
1572 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1574 GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
1576 result = gst_audio_encoder_activate (enc, active);
1579 enc->priv->active = active;
1581 GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
1583 gst_object_unref (enc);
1588 * gst_audio_encoder_get_audio_info:
1589 * @enc: a #GstAudioEncoder
1591 * Returns: a #GstAudioInfo describing the input audio format
1596 gst_audio_encoder_get_audio_info (GstAudioEncoder * enc)
1598 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), NULL);
1600 return &enc->priv->ctx.info;
1604 * gst_audio_encoder_set_frame_samples:
1605 * @enc: a #GstAudioEncoder
1606 * @num: number of samples per frame
1608 * Sets number of samples (per channel) subclass needs to be handed,
1609 * or will be handed all available if 0.
1614 gst_audio_encoder_set_frame_samples (GstAudioEncoder * enc, gint num)
1616 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1618 enc->priv->ctx.frame_samples = num;
1622 * gst_audio_encoder_get_frame_samples:
1623 * @enc: a #GstAudioEncoder
1625 * Returns: currently requested samples per frame
1630 gst_audio_encoder_get_frame_samples (GstAudioEncoder * enc)
1632 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1634 return enc->priv->ctx.frame_samples;
1638 * gst_audio_encoder_set_frame_max:
1639 * @enc: a #GstAudioEncoder
1640 * @num: number of frames
1642 * Sets max number of frames accepted at once (assumed minimally 1)
1647 gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num)
1649 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1651 enc->priv->ctx.frame_max = num;
1655 * gst_audio_encoder_get_frame_max:
1656 * @enc: a #GstAudioEncoder
1658 * Returns: currently configured maximum handled frames
1663 gst_audio_encoder_get_frame_max (GstAudioEncoder * enc)
1665 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1667 return enc->priv->ctx.frame_max;
1671 * gst_audio_encoder_set_lookahead:
1672 * @enc: a #GstAudioEncoder
1675 * Sets encoder lookahead (in units of input rate samples)
1680 gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num)
1682 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1684 enc->priv->ctx.lookahead = num;
1688 * gst_audio_encoder_get_lookahead:
1689 * @enc: a #GstAudioEncoder
1691 * Returns: currently configured encoder lookahead
1694 gst_audio_encoder_get_lookahead (GstAudioEncoder * enc)
1696 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1698 return enc->priv->ctx.lookahead;
1702 * gst_audio_encoder_set_latency:
1703 * @enc: a #GstAudioEncoder
1704 * @min: minimum latency
1705 * @max: maximum latency
1707 * Sets encoder latency.
1712 gst_audio_encoder_set_latency (GstAudioEncoder * enc,
1713 GstClockTime min, GstClockTime max)
1715 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1717 GST_OBJECT_LOCK (enc);
1718 enc->priv->ctx.min_latency = min;
1719 enc->priv->ctx.max_latency = max;
1720 GST_OBJECT_UNLOCK (enc);
1724 * gst_audio_encoder_get_latency:
1725 * @enc: a #GstAudioEncoder
1726 * @min: (out) (allow-none): a pointer to storage to hold minimum latency
1727 * @max: (out) (allow-none): a pointer to storage to hold maximum latency
1729 * Sets the variables pointed to by @min and @max to the currently configured
1735 gst_audio_encoder_get_latency (GstAudioEncoder * enc,
1736 GstClockTime * min, GstClockTime * max)
1738 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1740 GST_OBJECT_LOCK (enc);
1742 *min = enc->priv->ctx.min_latency;
1744 *max = enc->priv->ctx.max_latency;
1745 GST_OBJECT_UNLOCK (enc);
1749 * gst_audio_encoder_set_mark_granule:
1750 * @enc: a #GstAudioEncoder
1751 * @enabled: new state
1753 * Enable or disable encoder granule handling.
1760 gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc, gboolean enabled)
1762 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1764 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1766 GST_OBJECT_LOCK (enc);
1767 enc->priv->granule = enabled;
1768 GST_OBJECT_UNLOCK (enc);
1772 * gst_audio_encoder_get_mark_granule:
1773 * @enc: a #GstAudioEncoder
1775 * Queries if the encoder will handle granule marking.
1777 * Returns: TRUE if granule marking is enabled.
1784 gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc)
1788 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1790 GST_OBJECT_LOCK (enc);
1791 result = enc->priv->granule;
1792 GST_OBJECT_UNLOCK (enc);
1798 * gst_audio_encoder_set_perfect_timestamp:
1799 * @enc: a #GstAudioEncoder
1800 * @enabled: new state
1802 * Enable or disable encoder perfect output timestamp preference.
1809 gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
1812 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1814 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1816 GST_OBJECT_LOCK (enc);
1817 enc->priv->perfect_ts = enabled;
1818 GST_OBJECT_UNLOCK (enc);
1822 * gst_audio_encoder_get_perfect_timestamp:
1823 * @enc: a #GstAudioEncoder
1825 * Queries encoder perfect timestamp behaviour.
1827 * Returns: TRUE if pefect timestamp setting enabled.
1834 gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc)
1838 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1840 GST_OBJECT_LOCK (enc);
1841 result = enc->priv->perfect_ts;
1842 GST_OBJECT_UNLOCK (enc);
1848 * gst_audio_encoder_set_hard_sync:
1849 * @enc: a #GstAudioEncoder
1850 * @enabled: new state
1852 * Sets encoder hard resync handling.
1859 gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc, gboolean enabled)
1861 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1863 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1865 GST_OBJECT_LOCK (enc);
1866 enc->priv->hard_resync = enabled;
1867 GST_OBJECT_UNLOCK (enc);
1871 * gst_audio_encoder_get_hard_sync:
1872 * @enc: a #GstAudioEncoder
1874 * Queries encoder's hard resync setting.
1876 * Returns: TRUE if hard resync is enabled.
1883 gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc)
1887 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1889 GST_OBJECT_LOCK (enc);
1890 result = enc->priv->hard_resync;
1891 GST_OBJECT_UNLOCK (enc);
1897 * gst_audio_encoder_set_tolerance:
1898 * @enc: a #GstAudioEncoder
1899 * @tolerance: new tolerance
1901 * Configures encoder audio jitter tolerance threshold.
1908 gst_audio_encoder_set_tolerance (GstAudioEncoder * enc, gint64 tolerance)
1910 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1912 GST_OBJECT_LOCK (enc);
1913 enc->priv->tolerance = tolerance;
1914 GST_OBJECT_UNLOCK (enc);
1918 * gst_audio_encoder_get_tolerance:
1919 * @enc: a #GstAudioEncoder
1921 * Queries current audio jitter tolerance threshold.
1923 * Returns: encoder audio jitter tolerance threshold.
1930 gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)
1934 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1936 GST_OBJECT_LOCK (enc);
1937 result = enc->priv->tolerance;
1938 GST_OBJECT_UNLOCK (enc);