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>
157 #include <gst/pbutils/descriptions.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
190 gint frame_samples_min, frame_samples_max;
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 /* pending serialized sink events, will be sent from finish_frame() */
246 GList *pending_events;
250 static GstElementClass *parent_class = NULL;
252 static void gst_audio_encoder_class_init (GstAudioEncoderClass * klass);
253 static void gst_audio_encoder_init (GstAudioEncoder * parse,
254 GstAudioEncoderClass * klass);
257 gst_audio_encoder_get_type (void)
259 static GType audio_encoder_type = 0;
261 if (!audio_encoder_type) {
262 static const GTypeInfo audio_encoder_info = {
263 sizeof (GstAudioEncoderClass),
264 (GBaseInitFunc) NULL,
265 (GBaseFinalizeFunc) NULL,
266 (GClassInitFunc) gst_audio_encoder_class_init,
269 sizeof (GstAudioEncoder),
271 (GInstanceInitFunc) gst_audio_encoder_init,
273 const GInterfaceInfo preset_interface_info = {
274 NULL, /* interface_init */
275 NULL, /* interface_finalize */
276 NULL /* interface_data */
279 audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
280 "GstAudioEncoder", &audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
282 g_type_add_interface_static (audio_encoder_type, GST_TYPE_PRESET,
283 &preset_interface_info);
285 return audio_encoder_type;
288 static void gst_audio_encoder_finalize (GObject * object);
289 static void gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full);
291 static void gst_audio_encoder_set_property (GObject * object,
292 guint prop_id, const GValue * value, GParamSpec * pspec);
293 static void gst_audio_encoder_get_property (GObject * object,
294 guint prop_id, GValue * value, GParamSpec * pspec);
296 static gboolean gst_audio_encoder_sink_activate_push (GstPad * pad,
299 static gboolean gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event);
300 static gboolean gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc,
302 static GstFlowReturn gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer);
303 static gboolean gst_audio_encoder_src_query (GstPad * pad, GstQuery * query);
304 static gboolean gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query);
305 static const GstQueryType *gst_audio_encoder_get_query_types (GstPad * pad);
306 static GstCaps *gst_audio_encoder_sink_getcaps (GstPad * pad, GstCaps * filter);
309 gst_audio_encoder_class_init (GstAudioEncoderClass * klass)
311 GObjectClass *gobject_class;
313 gobject_class = G_OBJECT_CLASS (klass);
314 parent_class = g_type_class_peek_parent (klass);
316 GST_DEBUG_CATEGORY_INIT (gst_audio_encoder_debug, "audioencoder", 0,
317 "audio encoder base class");
319 g_type_class_add_private (klass, sizeof (GstAudioEncoderPrivate));
321 gobject_class->set_property = gst_audio_encoder_set_property;
322 gobject_class->get_property = gst_audio_encoder_get_property;
324 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_encoder_finalize);
327 g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
328 g_param_spec_boolean ("perfect-timestamp", "Perfect Timestamps",
329 "Favour perfect timestamps over tracking upstream timestamps",
330 DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
331 g_object_class_install_property (gobject_class, PROP_GRANULE,
332 g_param_spec_boolean ("mark-granule", "Granule Marking",
333 "Apply granule semantics to buffer metadata (implies perfect-timestamp)",
334 DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
335 g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
336 g_param_spec_boolean ("hard-resync", "Hard Resync",
337 "Perform clipping and sample flushing upon discontinuity",
338 DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
339 g_object_class_install_property (gobject_class, PROP_TOLERANCE,
340 g_param_spec_int64 ("tolerance", "Tolerance",
341 "Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
342 0, G_MAXINT64, DEFAULT_TOLERANCE,
343 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
347 gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
349 GstPadTemplate *pad_template;
351 GST_DEBUG_OBJECT (enc, "gst_audio_encoder_init");
353 enc->priv = GST_AUDIO_ENCODER_GET_PRIVATE (enc);
355 /* only push mode supported */
357 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
358 g_return_if_fail (pad_template != NULL);
359 enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
360 gst_pad_set_event_function (enc->sinkpad,
361 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_event));
362 gst_pad_set_getcaps_function (enc->sinkpad,
363 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_getcaps));
364 gst_pad_set_query_function (enc->sinkpad,
365 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_query));
366 gst_pad_set_chain_function (enc->sinkpad,
367 GST_DEBUG_FUNCPTR (gst_audio_encoder_chain));
368 gst_pad_set_activatepush_function (enc->sinkpad,
369 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_activate_push));
370 gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
372 GST_DEBUG_OBJECT (enc, "sinkpad created");
374 /* and we don't mind upstream traveling stuff that much ... */
376 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
377 g_return_if_fail (pad_template != NULL);
378 enc->srcpad = gst_pad_new_from_template (pad_template, "src");
379 gst_pad_set_query_function (enc->srcpad,
380 GST_DEBUG_FUNCPTR (gst_audio_encoder_src_query));
381 gst_pad_set_query_type_function (enc->srcpad,
382 GST_DEBUG_FUNCPTR (gst_audio_encoder_get_query_types));
383 gst_pad_use_fixed_caps (enc->srcpad);
384 gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
385 GST_DEBUG_OBJECT (enc, "src created");
387 enc->priv->adapter = gst_adapter_new ();
389 g_static_rec_mutex_init (&enc->stream_lock);
391 /* property default */
392 enc->priv->granule = DEFAULT_GRANULE;
393 enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
394 enc->priv->hard_resync = DEFAULT_HARD_RESYNC;
395 enc->priv->tolerance = DEFAULT_TOLERANCE;
398 gst_audio_encoder_reset (enc, TRUE);
399 GST_DEBUG_OBJECT (enc, "init ok");
403 gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
405 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
407 GST_LOG_OBJECT (enc, "reset full %d", full);
410 enc->priv->active = FALSE;
411 enc->priv->samples_in = 0;
412 enc->priv->bytes_out = 0;
413 gst_audio_info_init (&enc->priv->ctx.info);
414 memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
417 gst_tag_list_free (enc->priv->tags);
418 enc->priv->tags = NULL;
420 g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
421 g_list_free (enc->priv->pending_events);
422 enc->priv->pending_events = NULL;
425 gst_segment_init (&enc->segment, GST_FORMAT_TIME);
427 gst_adapter_clear (enc->priv->adapter);
428 enc->priv->got_data = FALSE;
429 enc->priv->drained = TRUE;
430 enc->priv->offset = 0;
431 enc->priv->base_ts = GST_CLOCK_TIME_NONE;
432 enc->priv->base_gp = -1;
433 enc->priv->samples = 0;
434 enc->priv->discont = FALSE;
436 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
440 gst_audio_encoder_finalize (GObject * object)
442 GstAudioEncoder *enc = GST_AUDIO_ENCODER (object);
444 g_object_unref (enc->priv->adapter);
446 g_static_rec_mutex_free (&enc->stream_lock);
448 G_OBJECT_CLASS (parent_class)->finalize (object);
452 * gst_audio_encoder_finish_frame:
453 * @enc: a #GstAudioEncoder
454 * @buffer: encoded data
455 * @samples: number of samples (per channel) represented by encoded data
457 * Collects encoded data and pushes encoded data downstream.
458 * Source pad caps must be set when this is called.
460 * If @samples < 0, then best estimate is all samples provided to encoder
461 * (subclass) so far. @buf may be NULL, in which case next number of @samples
462 * are considered discarded, e.g. as a result of discontinuous transmission,
463 * and a discontinuity is marked.
465 * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
470 gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
473 GstAudioEncoderClass *klass;
474 GstAudioEncoderPrivate *priv;
475 GstAudioEncoderContext *ctx;
476 GstFlowReturn ret = GST_FLOW_OK;
478 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
480 ctx = &enc->priv->ctx;
482 /* subclass should know what it is producing by now */
483 g_return_val_if_fail (gst_pad_has_current_caps (enc->srcpad), GST_FLOW_ERROR);
484 /* subclass should not hand us no data */
485 g_return_val_if_fail (buf == NULL || gst_buffer_get_size (buf) > 0,
488 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
490 GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
491 buf ? gst_buffer_get_size (buf) : -1, samples);
493 /* mark subclass still alive and providing */
495 priv->got_data = TRUE;
497 if (priv->pending_events) {
498 GList *pending_events, *l;
500 pending_events = priv->pending_events;
501 priv->pending_events = NULL;
503 GST_DEBUG_OBJECT (enc, "Pushing pending events");
504 for (l = pending_events; l; l = l->next)
505 gst_pad_push_event (enc->srcpad, l->data);
506 g_list_free (pending_events);
509 /* send after pending events, which likely includes newsegment event */
510 if (G_UNLIKELY (enc->priv->tags)) {
516 /* add codec info to pending tags */
517 tags = enc->priv->tags;
518 /* no more pending */
519 enc->priv->tags = NULL;
521 caps = gst_pad_get_current_caps (enc->srcpad);
522 gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_CODEC, caps);
523 gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_AUDIO_CODEC,
526 GST_DEBUG_OBJECT (enc, "sending tags %" GST_PTR_FORMAT, tags);
527 gst_element_found_tags_for_pad (GST_ELEMENT (enc), enc->srcpad, tags);
530 /* remove corresponding samples from input */
532 samples = (enc->priv->offset / ctx->info.bpf);
534 if (G_LIKELY (samples)) {
535 /* track upstream ts if so configured */
536 if (!enc->priv->perfect_ts) {
537 guint64 ts, distance;
539 ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
540 g_assert (distance % ctx->info.bpf == 0);
541 distance /= ctx->info.bpf;
542 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
543 GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
544 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
545 GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
546 /* when draining adapter might be empty and no ts to offer */
547 if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
548 GstClockTimeDiff diff;
549 GstClockTime old_ts, next_ts;
551 /* passed into another buffer;
552 * mild check for discontinuity and only mark if so */
554 gst_util_uint64_scale (distance, GST_SECOND, ctx->info.rate);
555 old_ts = priv->base_ts +
556 gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->info.rate);
557 diff = GST_CLOCK_DIFF (next_ts, old_ts);
558 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
559 /* only mark discontinuity if beyond tolerance */
560 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
561 diff > enc->priv->tolerance)) {
562 GST_DEBUG_OBJECT (enc, "marked discont");
563 priv->discont = TRUE;
565 if (diff > GST_SECOND / ctx->info.rate / 2 ||
566 diff < -GST_SECOND / ctx->info.rate / 2) {
567 GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
568 " at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
569 /* re-sync to upstream ts */
571 priv->samples = distance;
573 GST_LOG_OBJECT (enc, "new upstream ts only introduces jitter");
577 /* advance sample view */
578 if (G_UNLIKELY (samples * ctx->info.bpf > priv->offset)) {
579 if (G_LIKELY (!priv->force)) {
580 /* no way we can let this pass */
581 g_assert_not_reached ();
586 if (samples * ctx->info.bpf >= gst_adapter_available (priv->adapter))
587 gst_adapter_clear (priv->adapter);
589 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
592 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
593 priv->offset -= samples * ctx->info.bpf;
594 /* avoid subsequent stray prev_ts */
595 if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
596 gst_adapter_clear (priv->adapter);
598 /* sample count advanced below after buffer handling */
602 if (G_LIKELY (buf)) {
605 size = gst_buffer_get_size (buf);
607 GST_LOG_OBJECT (enc, "taking %d bytes for output", size);
608 buf = gst_buffer_make_writable (buf);
611 if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
612 /* FIXME ? lookahead could lead to weird ts and duration ?
613 * (particularly if not in perfect mode) */
614 /* mind sample rounding and produce perfect output */
615 GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
616 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
618 GST_DEBUG_OBJECT (enc, "out samples %d", samples);
619 if (G_LIKELY (samples > 0)) {
620 priv->samples += samples;
621 GST_BUFFER_DURATION (buf) = priv->base_ts +
622 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
623 ctx->info.rate) - GST_BUFFER_TIMESTAMP (buf);
624 priv->last_duration = GST_BUFFER_DURATION (buf);
626 /* duration forecast in case of handling remainder;
627 * the last one is probably like the previous one ... */
628 GST_BUFFER_DURATION (buf) = priv->last_duration;
630 if (priv->base_gp >= 0) {
632 /* FIXME: in longer run, muxer should take care of this ... */
633 /* offset_end = granulepos for ogg muxer */
634 GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples -
635 enc->priv->ctx.lookahead;
636 /* offset = timestamp corresponding to granulepos for ogg muxer */
637 GST_BUFFER_OFFSET (buf) =
638 GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf),
641 GST_BUFFER_OFFSET (buf) = priv->bytes_out;
642 GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + size;
646 priv->bytes_out += size;
648 if (G_UNLIKELY (priv->discont)) {
649 GST_LOG_OBJECT (enc, "marking discont");
650 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
651 priv->discont = FALSE;
654 if (klass->pre_push) {
655 /* last chance for subclass to do some dirty stuff */
656 ret = klass->pre_push (enc, &buf);
657 if (ret != GST_FLOW_OK || !buf) {
658 GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
659 gst_flow_get_name (ret), buf);
661 gst_buffer_unref (buf);
666 GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
667 ", duration %" GST_TIME_FORMAT, size,
668 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
669 GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
671 ret = gst_pad_push (enc->srcpad, buf);
672 GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
674 /* merely advance samples, most work for that already done above */
675 priv->samples += samples;
679 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
686 GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
687 ("received more encoded samples %d than provided %d",
688 samples, priv->offset / ctx->info.bpf), (NULL));
690 gst_buffer_unref (buf);
691 ret = GST_FLOW_ERROR;
696 /* adapter tracking idea:
697 * - start of adapter corresponds with what has already been encoded
698 * (i.e. really returned by encoder subclass)
699 * - start + offset is what needs to be fed to subclass next */
701 gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
703 GstAudioEncoderClass *klass;
704 GstAudioEncoderPrivate *priv;
705 GstAudioEncoderContext *ctx;
708 GstFlowReturn ret = GST_FLOW_OK;
710 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
712 g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
715 ctx = &enc->priv->ctx;
717 while (ret == GST_FLOW_OK) {
720 av = gst_adapter_available (priv->adapter);
722 g_assert (priv->offset <= av);
726 ctx->frame_samples_min >
727 0 ? ctx->frame_samples_min * ctx->info.bpf : av;
728 GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d", av, need,
731 if ((need > av) || !av) {
732 if (G_UNLIKELY (force)) {
742 if (ctx->frame_samples_max > 0)
743 need = MIN (av, ctx->frame_samples_max * ctx->info.bpf);
745 if (ctx->frame_samples_min == ctx->frame_samples_max) {
746 /* if we have some extra metadata,
747 * provide for integer multiple of frames to allow for better granularity
749 if (ctx->frame_samples_min > 0 && need) {
750 if (ctx->frame_max > 1)
751 need = need * MIN ((av / need), ctx->frame_max);
752 else if (ctx->frame_max == 0)
753 need = need * (av / need);
760 data = gst_adapter_map (priv->adapter, priv->offset + need);
762 gst_buffer_new_wrapped_full ((gpointer) data, NULL, priv->offset,
766 GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
769 /* mark this already as consumed,
770 * which it should be when subclass gives us data in exchange for samples */
771 priv->offset += need;
772 priv->samples_in += need / ctx->info.bpf;
774 priv->got_data = FALSE;
775 ret = klass->handle_frame (enc, buf);
777 if (G_LIKELY (buf)) {
778 gst_buffer_unref (buf);
779 gst_adapter_unmap (priv->adapter, 0);
782 /* no data to feed, no leftover provided, then bail out */
783 if (G_UNLIKELY (!buf && !priv->got_data)) {
784 priv->drained = TRUE;
785 GST_LOG_OBJECT (enc, "no more data drained from subclass");
794 gst_audio_encoder_drain (GstAudioEncoder * enc)
796 if (enc->priv->drained)
799 return gst_audio_encoder_push_buffers (enc, TRUE);
803 gst_audio_encoder_set_base_gp (GstAudioEncoder * enc)
807 if (!enc->priv->granule)
810 /* use running time for granule */
811 /* incoming data is clipped, so a valid input should yield a valid output */
812 ts = gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME,
814 if (GST_CLOCK_TIME_IS_VALID (ts)) {
816 GST_CLOCK_TIME_TO_FRAMES (enc->priv->base_ts, enc->priv->ctx.info.rate);
817 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, enc->priv->base_gp);
819 /* should reasonably have a valid base,
820 * otherwise start at 0 if we did not already start there earlier */
821 if (enc->priv->base_gp < 0) {
822 enc->priv->base_gp = 0;
823 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
830 gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
832 GstAudioEncoder *enc;
833 GstAudioEncoderPrivate *priv;
834 GstAudioEncoderContext *ctx;
835 GstFlowReturn ret = GST_FLOW_OK;
839 enc = GST_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
842 ctx = &enc->priv->ctx;
844 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
846 /* should know what is coming by now */
850 size = gst_buffer_get_size (buffer);
853 "received buffer of size %d with ts %" GST_TIME_FORMAT
854 ", duration %" GST_TIME_FORMAT, size,
855 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
856 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
858 /* input shoud be whole number of sample frames */
859 if (size % ctx->info.bpf)
862 #ifndef GST_DISABLE_GST_DEBUG
864 GstClockTime duration;
865 GstClockTimeDiff diff;
867 /* verify buffer duration */
868 duration = gst_util_uint64_scale (size, GST_SECOND,
869 ctx->info.rate * ctx->info.bpf);
870 diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
871 if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
872 (diff > GST_SECOND / ctx->info.rate / 2 ||
873 diff < -GST_SECOND / ctx->info.rate / 2)) {
874 GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
875 GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
876 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
877 GST_TIME_ARGS (duration));
882 discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
883 if (G_UNLIKELY (discont)) {
884 GST_LOG_OBJECT (buffer, "marked discont");
885 enc->priv->discont = discont;
888 /* clip to segment */
889 /* NOTE: slightly painful linking -laudio only for this one ... */
890 buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->info.rate,
892 if (G_UNLIKELY (!buffer)) {
893 GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
897 size = gst_buffer_get_size (buffer);
900 "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
901 ", duration %" GST_TIME_FORMAT, size,
902 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
903 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
905 if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
906 priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
907 GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
908 GST_TIME_ARGS (priv->base_ts));
909 gst_audio_encoder_set_base_gp (enc);
912 /* check for continuity;
913 * checked elsewhere in non-perfect case */
914 if (enc->priv->perfect_ts) {
915 GstClockTimeDiff diff = 0;
916 GstClockTime next_ts = 0;
918 if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
919 GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
922 samples = priv->samples +
923 gst_adapter_available (priv->adapter) / ctx->info.bpf;
924 next_ts = priv->base_ts +
925 gst_util_uint64_scale (samples, GST_SECOND, ctx->info.rate);
926 GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
927 " samples past base_ts %" GST_TIME_FORMAT
928 ", expected ts %" GST_TIME_FORMAT, samples,
929 GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
930 diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
931 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
932 /* if within tolerance,
933 * discard buffer ts and carry on producing perfect stream,
934 * otherwise clip or resync to ts */
935 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
936 diff > enc->priv->tolerance)) {
937 GST_DEBUG_OBJECT (enc, "marked discont");
942 /* do some fancy tweaking in hard resync case */
943 if (discont && enc->priv->hard_resync) {
947 GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
948 GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
951 GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->info.rate) * ctx->info.bpf;
952 if (diff_bytes >= size) {
953 gst_buffer_unref (buffer);
956 buffer = gst_buffer_make_writable (buffer);
957 gst_buffer_resize (buffer, diff_bytes, size - diff_bytes);
959 GST_BUFFER_TIMESTAMP (buffer) += diff;
960 /* care even less about duration after this */
962 /* drain stuff prior to resync */
963 gst_audio_encoder_drain (enc);
967 priv->base_ts += diff;
968 gst_audio_encoder_set_base_gp (enc);
969 priv->discont |= discont;
972 gst_adapter_push (enc->priv->adapter, buffer);
973 /* new stuff, so we can push subclass again */
974 enc->priv->drained = FALSE;
976 ret = gst_audio_encoder_push_buffers (enc, FALSE);
979 GST_LOG_OBJECT (enc, "chain leaving");
981 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
988 GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
989 ("encoder not initialized"));
990 gst_buffer_unref (buffer);
991 ret = GST_FLOW_NOT_NEGOTIATED;
996 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
997 ("buffer size %d not a multiple of %d", gst_buffer_get_size (buffer),
999 gst_buffer_unref (buffer);
1000 ret = GST_FLOW_ERROR;
1006 audio_info_is_equal (GstAudioInfo * from, GstAudioInfo * to)
1010 if (from->finfo == NULL || to->finfo == NULL)
1012 if (GST_AUDIO_INFO_FORMAT (from) != GST_AUDIO_INFO_FORMAT (to))
1014 if (GST_AUDIO_INFO_RATE (from) != GST_AUDIO_INFO_RATE (to))
1016 if (GST_AUDIO_INFO_CHANNELS (from) != GST_AUDIO_INFO_CHANNELS (to))
1018 if (GST_AUDIO_INFO_CHANNELS (from) > 64)
1020 return memcmp (from->position, to->position,
1021 GST_AUDIO_INFO_CHANNELS (from) * sizeof (to->position[0]));
1025 gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
1027 GstAudioEncoderClass *klass;
1028 GstAudioEncoderContext *ctx;
1030 gboolean res = TRUE, changed = FALSE;
1033 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1035 /* subclass must do something here ... */
1036 g_return_val_if_fail (klass->set_format != NULL, FALSE);
1038 ctx = &enc->priv->ctx;
1040 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1042 GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
1044 if (!gst_caps_is_fixed (caps))
1047 /* adjust ts tracking to new sample rate */
1048 old_rate = GST_AUDIO_INFO_RATE (&ctx->info);
1049 if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
1050 enc->priv->base_ts +=
1051 GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
1052 enc->priv->samples = 0;
1055 if (!gst_audio_info_from_caps (&state, caps))
1058 changed = !audio_info_is_equal (&state, &ctx->info);
1061 GstClockTime old_min_latency;
1062 GstClockTime old_max_latency;
1064 /* drain any pending old data stuff */
1065 gst_audio_encoder_drain (enc);
1067 /* context defaults */
1068 enc->priv->ctx.frame_samples_min = 0;
1069 enc->priv->ctx.frame_samples_max = 0;
1070 enc->priv->ctx.frame_max = 0;
1071 enc->priv->ctx.lookahead = 0;
1073 /* element might report latency */
1074 GST_OBJECT_LOCK (enc);
1075 old_min_latency = ctx->min_latency;
1076 old_max_latency = ctx->max_latency;
1077 GST_OBJECT_UNLOCK (enc);
1079 if (klass->set_format)
1080 res = klass->set_format (enc, &state);
1082 /* notify if new latency */
1083 GST_OBJECT_LOCK (enc);
1084 if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
1085 (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
1086 GST_OBJECT_UNLOCK (enc);
1087 /* post latency message on the bus */
1088 gst_element_post_message (GST_ELEMENT (enc),
1089 gst_message_new_latency (GST_OBJECT (enc)));
1090 GST_OBJECT_LOCK (enc);
1092 GST_OBJECT_UNLOCK (enc);
1094 GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
1099 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1106 GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
1113 * gst_audio_encoder_proxy_getcaps:
1114 * @enc: a #GstAudioEncoder
1115 * @caps: initial caps
1117 * Returns caps that express @caps (or sink template caps if @caps == NULL)
1118 * restricted to channel/rate combinations supported by downstream elements
1121 * Returns: a #GstCaps owned by caller
1126 gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps)
1128 const GstCaps *templ_caps;
1129 GstCaps *allowed = NULL;
1130 GstCaps *fcaps, *filter_caps;
1133 /* we want to be able to communicate to upstream elements like audioconvert
1134 * and audioresample any rate/channel restrictions downstream (e.g. muxer
1135 * only accepting certain sample rates) */
1136 templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad);
1137 allowed = gst_pad_get_allowed_caps (enc->srcpad);
1138 if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
1139 fcaps = gst_caps_copy (templ_caps);
1143 GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
1144 GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
1146 filter_caps = gst_caps_new_empty ();
1148 for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
1151 q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
1153 /* pick rate + channel fields from allowed caps */
1154 for (j = 0; j < gst_caps_get_size (allowed); j++) {
1155 const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
1159 s = gst_structure_id_empty_new (q_name);
1160 if ((val = gst_structure_get_value (allowed_s, "rate")))
1161 gst_structure_set_value (s, "rate", val);
1162 if ((val = gst_structure_get_value (allowed_s, "channels")))
1163 gst_structure_set_value (s, "channels", val);
1164 /* following might also make sense for some encoded formats,
1166 if ((val = gst_structure_get_value (allowed_s, "width")))
1167 gst_structure_set_value (s, "width", val);
1168 if ((val = gst_structure_get_value (allowed_s, "depth")))
1169 gst_structure_set_value (s, "depth", val);
1170 if ((val = gst_structure_get_value (allowed_s, "endianness")))
1171 gst_structure_set_value (s, "endianness", val);
1172 if ((val = gst_structure_get_value (allowed_s, "signed")))
1173 gst_structure_set_value (s, "signed", val);
1174 if ((val = gst_structure_get_value (allowed_s, "channel-positions")))
1175 gst_structure_set_value (s, "channel-positions", val);
1177 gst_caps_merge_structure (filter_caps, s);
1181 fcaps = gst_caps_intersect (filter_caps, templ_caps);
1182 gst_caps_unref (filter_caps);
1185 gst_caps_replace (&allowed, NULL);
1187 GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
1193 gst_audio_encoder_sink_getcaps (GstPad * pad, GstCaps * filter)
1195 GstAudioEncoder *enc;
1196 GstAudioEncoderClass *klass;
1199 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1200 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1201 g_assert (pad == enc->sinkpad);
1204 caps = klass->getcaps (enc, filter);
1206 caps = gst_audio_encoder_proxy_getcaps (enc, NULL);
1207 gst_object_unref (enc);
1209 GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
1215 gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
1217 GstAudioEncoderClass *klass;
1218 gboolean handled = FALSE;
1220 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1222 switch (GST_EVENT_TYPE (event)) {
1223 case GST_EVENT_SEGMENT:
1227 gst_event_copy_segment (event, &seg);
1229 if (seg.format == GST_FORMAT_TIME) {
1230 GST_DEBUG_OBJECT (enc, "received TIME SEGMENT %" GST_PTR_FORMAT, &seg);
1232 GST_DEBUG_OBJECT (enc, "received SEGMENT %" GST_PTR_FORMAT, &seg);
1233 GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
1237 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1238 /* finish current segment */
1239 gst_audio_encoder_drain (enc);
1240 /* reset partially for new segment */
1241 gst_audio_encoder_reset (enc, FALSE);
1242 /* and follow along with segment */
1244 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1248 case GST_EVENT_FLUSH_START:
1251 case GST_EVENT_FLUSH_STOP:
1252 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1253 /* discard any pending stuff */
1254 /* TODO route through drain ?? */
1255 if (!enc->priv->drained && klass->flush)
1257 /* and get (re)set for the sequel */
1258 gst_audio_encoder_reset (enc, FALSE);
1260 g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
1261 g_list_free (enc->priv->pending_events);
1262 enc->priv->pending_events = NULL;
1263 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1268 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1269 gst_audio_encoder_drain (enc);
1270 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1277 gst_event_parse_tag (event, &tags);
1278 tags = gst_tag_list_copy (tags);
1279 gst_event_unref (event);
1281 /* FIXME: make generic based on GST_TAG_FLAG_ENCODED */
1282 gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
1283 gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
1284 gst_tag_list_remove_tag (tags, GST_TAG_VIDEO_CODEC);
1285 gst_tag_list_remove_tag (tags, GST_TAG_SUBTITLE_CODEC);
1286 gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
1287 gst_tag_list_remove_tag (tags, GST_TAG_BITRATE);
1288 gst_tag_list_remove_tag (tags, GST_TAG_NOMINAL_BITRATE);
1289 gst_tag_list_remove_tag (tags, GST_TAG_MAXIMUM_BITRATE);
1290 gst_tag_list_remove_tag (tags, GST_TAG_MINIMUM_BITRATE);
1291 gst_tag_list_remove_tag (tags, GST_TAG_ENCODER);
1292 gst_tag_list_remove_tag (tags, GST_TAG_ENCODER_VERSION);
1293 event = gst_event_new_tag (tags);
1295 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1296 enc->priv->pending_events =
1297 g_list_append (enc->priv->pending_events, event);
1298 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1303 case GST_EVENT_CAPS:
1307 gst_event_parse_caps (event, &caps);
1308 gst_audio_encoder_sink_setcaps (enc, caps);
1309 gst_event_unref (event);
1322 gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
1324 GstAudioEncoder *enc;
1325 GstAudioEncoderClass *klass;
1326 gboolean handled = FALSE;
1327 gboolean ret = TRUE;
1329 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1330 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1332 GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
1333 GST_EVENT_TYPE_NAME (event));
1336 handled = klass->event (enc, event);
1339 handled = gst_audio_encoder_sink_eventfunc (enc, event);
1342 /* Forward non-serialized events and EOS/FLUSH_STOP immediately.
1343 * For EOS this is required because no buffer or serialized event
1344 * will come after EOS and nothing could trigger another
1345 * _finish_frame() call.
1347 * For FLUSH_STOP this is required because it is expected
1348 * to be forwarded immediately and no buffers are queued anyway.
1350 if (!GST_EVENT_IS_SERIALIZED (event)
1351 || GST_EVENT_TYPE (event) == GST_EVENT_EOS
1352 || GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
1353 ret = gst_pad_event_default (pad, event);
1355 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1356 enc->priv->pending_events =
1357 g_list_append (enc->priv->pending_events, event);
1358 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1363 GST_DEBUG_OBJECT (enc, "event handled");
1365 gst_object_unref (enc);
1370 gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
1372 gboolean res = TRUE;
1373 GstAudioEncoder *enc;
1375 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1377 switch (GST_QUERY_TYPE (query)) {
1378 case GST_QUERY_FORMATS:
1380 gst_query_set_formats (query, 3,
1381 GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
1385 case GST_QUERY_CONVERT:
1387 GstFormat src_fmt, dest_fmt;
1388 gint64 src_val, dest_val;
1390 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1391 if (!(res = gst_audio_info_convert (&enc->priv->ctx.info,
1392 src_fmt, src_val, dest_fmt, &dest_val)))
1394 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1398 res = gst_pad_query_default (pad, query);
1403 gst_object_unref (enc);
1407 static const GstQueryType *
1408 gst_audio_encoder_get_query_types (GstPad * pad)
1410 static const GstQueryType gst_audio_encoder_src_query_types[] = {
1418 return gst_audio_encoder_src_query_types;
1422 * gst_audio_encoded_audio_convert:
1423 * @fmt: audio format of the encoded audio
1424 * @bytes: number of encoded bytes
1425 * @samples: number of encoded samples
1426 * @src_format: source format
1427 * @src_value: source value
1428 * @dest_format: destination format
1429 * @dest_value: destination format
1431 * Helper function to convert @src_value in @src_format to @dest_value in
1432 * @dest_format for encoded audio data. Conversion is possible between
1433 * BYTE and TIME format by using estimated bitrate based on
1434 * @samples and @bytes (and @fmt).
1438 /* FIXME: make gst_audio_encoded_audio_convert() public? */
1440 gst_audio_encoded_audio_convert (GstAudioInfo * fmt,
1441 gint64 bytes, gint64 samples, GstFormat src_format,
1442 gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
1444 gboolean res = FALSE;
1446 g_return_val_if_fail (dest_format != NULL, FALSE);
1447 g_return_val_if_fail (dest_value != NULL, FALSE);
1449 if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
1452 *dest_value = src_value;
1456 if (samples == 0 || bytes == 0 || fmt->rate == 0) {
1457 GST_DEBUG ("not enough metadata yet to convert");
1463 switch (src_format) {
1464 case GST_FORMAT_BYTES:
1465 switch (*dest_format) {
1466 case GST_FORMAT_TIME:
1467 *dest_value = gst_util_uint64_scale (src_value,
1468 GST_SECOND * samples, bytes);
1475 case GST_FORMAT_TIME:
1476 switch (*dest_format) {
1477 case GST_FORMAT_BYTES:
1478 *dest_value = gst_util_uint64_scale (src_value, bytes,
1479 samples * GST_SECOND);
1494 /* FIXME ? are any of these queries (other than latency) an encoder's business
1495 * also, the conversion stuff might seem to make sense, but seems to not mind
1496 * segment stuff etc at all
1497 * Supposedly that's backward compatibility ... */
1499 gst_audio_encoder_src_query (GstPad * pad, GstQuery * query)
1501 GstAudioEncoder *enc;
1503 gboolean res = FALSE;
1505 enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
1506 peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
1508 GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
1510 switch (GST_QUERY_TYPE (query)) {
1511 case GST_QUERY_POSITION:
1513 GstFormat fmt, req_fmt;
1516 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1517 GST_LOG_OBJECT (enc, "returning peer response");
1522 GST_LOG_OBJECT (enc, "no peer");
1526 gst_query_parse_position (query, &req_fmt, NULL);
1527 fmt = GST_FORMAT_TIME;
1528 if (!(res = gst_pad_query_position (peerpad, fmt, &pos)))
1531 if ((res = gst_pad_query_convert (peerpad, fmt, pos, req_fmt, &val))) {
1532 gst_query_set_position (query, req_fmt, val);
1536 case GST_QUERY_DURATION:
1538 GstFormat fmt, req_fmt;
1541 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1542 GST_LOG_OBJECT (enc, "returning peer response");
1547 GST_LOG_OBJECT (enc, "no peer");
1551 gst_query_parse_duration (query, &req_fmt, NULL);
1552 fmt = GST_FORMAT_TIME;
1553 if (!(res = gst_pad_query_duration (peerpad, fmt, &dur)))
1556 if ((res = gst_pad_query_convert (peerpad, fmt, dur, req_fmt, &val))) {
1557 gst_query_set_duration (query, req_fmt, val);
1561 case GST_QUERY_FORMATS:
1563 gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
1567 case GST_QUERY_CONVERT:
1569 GstFormat src_fmt, dest_fmt;
1570 gint64 src_val, dest_val;
1572 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1573 if (!(res = gst_audio_encoded_audio_convert (&enc->priv->ctx.info,
1574 enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
1575 &dest_fmt, &dest_val)))
1577 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1580 case GST_QUERY_LATENCY:
1582 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1584 GstClockTime min_latency, max_latency;
1586 gst_query_parse_latency (query, &live, &min_latency, &max_latency);
1587 GST_DEBUG_OBJECT (enc, "Peer latency: live %d, min %"
1588 GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
1589 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
1591 GST_OBJECT_LOCK (enc);
1592 /* add our latency */
1593 if (min_latency != -1)
1594 min_latency += enc->priv->ctx.min_latency;
1595 if (max_latency != -1)
1596 max_latency += enc->priv->ctx.max_latency;
1597 GST_OBJECT_UNLOCK (enc);
1599 gst_query_set_latency (query, live, min_latency, max_latency);
1604 res = gst_pad_query_default (pad, query);
1608 gst_object_unref (peerpad);
1613 gst_audio_encoder_set_property (GObject * object, guint prop_id,
1614 const GValue * value, GParamSpec * pspec)
1616 GstAudioEncoder *enc;
1618 enc = GST_AUDIO_ENCODER (object);
1621 case PROP_PERFECT_TS:
1622 if (enc->priv->granule && !g_value_get_boolean (value))
1623 GST_WARNING_OBJECT (enc, "perfect-timestamp can not be set FALSE "
1624 "while granule handling is enabled");
1626 enc->priv->perfect_ts = g_value_get_boolean (value);
1628 case PROP_HARD_RESYNC:
1629 enc->priv->hard_resync = g_value_get_boolean (value);
1631 case PROP_TOLERANCE:
1632 enc->priv->tolerance = g_value_get_int64 (value);
1635 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1641 gst_audio_encoder_get_property (GObject * object, guint prop_id,
1642 GValue * value, GParamSpec * pspec)
1644 GstAudioEncoder *enc;
1646 enc = GST_AUDIO_ENCODER (object);
1649 case PROP_PERFECT_TS:
1650 g_value_set_boolean (value, enc->priv->perfect_ts);
1653 g_value_set_boolean (value, enc->priv->granule);
1655 case PROP_HARD_RESYNC:
1656 g_value_set_boolean (value, enc->priv->hard_resync);
1658 case PROP_TOLERANCE:
1659 g_value_set_int64 (value, enc->priv->tolerance);
1662 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1668 gst_audio_encoder_activate (GstAudioEncoder * enc, gboolean active)
1670 GstAudioEncoderClass *klass;
1671 gboolean result = FALSE;
1673 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1675 g_return_val_if_fail (!enc->priv->granule || enc->priv->perfect_ts, FALSE);
1677 GST_DEBUG_OBJECT (enc, "activate %d", active);
1681 if (enc->priv->tags)
1682 gst_tag_list_free (enc->priv->tags);
1683 enc->priv->tags = gst_tag_list_new ();
1685 if (!enc->priv->active && klass->start)
1686 result = klass->start (enc);
1688 /* We must make sure streaming has finished before resetting things
1689 * and calling the ::stop vfunc */
1690 GST_PAD_STREAM_LOCK (enc->sinkpad);
1691 GST_PAD_STREAM_UNLOCK (enc->sinkpad);
1693 if (enc->priv->active && klass->stop)
1694 result = klass->stop (enc);
1697 gst_audio_encoder_reset (enc, TRUE);
1699 GST_DEBUG_OBJECT (enc, "activate return: %d", result);
1705 gst_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
1707 gboolean result = TRUE;
1708 GstAudioEncoder *enc;
1710 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1712 GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
1714 result = gst_audio_encoder_activate (enc, active);
1717 enc->priv->active = active;
1719 GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
1721 gst_object_unref (enc);
1726 * gst_audio_encoder_get_audio_info:
1727 * @enc: a #GstAudioEncoder
1729 * Returns: a #GstAudioInfo describing the input audio format
1734 gst_audio_encoder_get_audio_info (GstAudioEncoder * enc)
1736 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), NULL);
1738 return &enc->priv->ctx.info;
1742 * gst_audio_encoder_set_frame_samples_min:
1743 * @enc: a #GstAudioEncoder
1744 * @num: number of samples per frame
1746 * Sets number of samples (per channel) subclass needs to be handed,
1747 * at least or will be handed all available if 0.
1749 * If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max()
1750 * must be called with the same number.
1755 gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num)
1757 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1759 enc->priv->ctx.frame_samples_min = num;
1763 * gst_audio_encoder_get_frame_samples_min:
1764 * @enc: a #GstAudioEncoder
1766 * Returns: currently minimum requested samples per frame
1771 gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc)
1773 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1775 return enc->priv->ctx.frame_samples_min;
1779 * gst_audio_encoder_set_frame_samples_max:
1780 * @enc: a #GstAudioEncoder
1781 * @num: number of samples per frame
1783 * Sets number of samples (per channel) subclass needs to be handed,
1784 * at most or will be handed all available if 0.
1786 * If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min()
1787 * must be called with the same number.
1792 gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num)
1794 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1796 enc->priv->ctx.frame_samples_max = num;
1800 * gst_audio_encoder_get_frame_samples_min:
1801 * @enc: a #GstAudioEncoder
1803 * Returns: currently maximum requested samples per frame
1808 gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc)
1810 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1812 return enc->priv->ctx.frame_samples_max;
1816 * gst_audio_encoder_set_frame_max:
1817 * @enc: a #GstAudioEncoder
1818 * @num: number of frames
1820 * Sets max number of frames accepted at once (assumed minimally 1).
1821 * Requires @frame_samples_min and @frame_samples_max to be the equal.
1826 gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num)
1828 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1830 enc->priv->ctx.frame_max = num;
1834 * gst_audio_encoder_get_frame_max:
1835 * @enc: a #GstAudioEncoder
1837 * Returns: currently configured maximum handled frames
1842 gst_audio_encoder_get_frame_max (GstAudioEncoder * enc)
1844 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1846 return enc->priv->ctx.frame_max;
1850 * gst_audio_encoder_set_lookahead:
1851 * @enc: a #GstAudioEncoder
1854 * Sets encoder lookahead (in units of input rate samples)
1859 gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num)
1861 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1863 enc->priv->ctx.lookahead = num;
1867 * gst_audio_encoder_get_lookahead:
1868 * @enc: a #GstAudioEncoder
1870 * Returns: currently configured encoder lookahead
1873 gst_audio_encoder_get_lookahead (GstAudioEncoder * enc)
1875 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1877 return enc->priv->ctx.lookahead;
1881 * gst_audio_encoder_set_latency:
1882 * @enc: a #GstAudioEncoder
1883 * @min: minimum latency
1884 * @max: maximum latency
1886 * Sets encoder latency.
1891 gst_audio_encoder_set_latency (GstAudioEncoder * enc,
1892 GstClockTime min, GstClockTime max)
1894 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1896 GST_OBJECT_LOCK (enc);
1897 enc->priv->ctx.min_latency = min;
1898 enc->priv->ctx.max_latency = max;
1899 GST_OBJECT_UNLOCK (enc);
1903 * gst_audio_encoder_get_latency:
1904 * @enc: a #GstAudioEncoder
1905 * @min: (out) (allow-none): a pointer to storage to hold minimum latency
1906 * @max: (out) (allow-none): a pointer to storage to hold maximum latency
1908 * Sets the variables pointed to by @min and @max to the currently configured
1914 gst_audio_encoder_get_latency (GstAudioEncoder * enc,
1915 GstClockTime * min, GstClockTime * max)
1917 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1919 GST_OBJECT_LOCK (enc);
1921 *min = enc->priv->ctx.min_latency;
1923 *max = enc->priv->ctx.max_latency;
1924 GST_OBJECT_UNLOCK (enc);
1928 * gst_audio_encoder_set_mark_granule:
1929 * @enc: a #GstAudioEncoder
1930 * @enabled: new state
1932 * Enable or disable encoder granule handling.
1939 gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc, gboolean enabled)
1941 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1943 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1945 GST_OBJECT_LOCK (enc);
1946 enc->priv->granule = enabled;
1947 GST_OBJECT_UNLOCK (enc);
1951 * gst_audio_encoder_get_mark_granule:
1952 * @enc: a #GstAudioEncoder
1954 * Queries if the encoder will handle granule marking.
1956 * Returns: TRUE if granule marking is enabled.
1963 gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc)
1967 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1969 GST_OBJECT_LOCK (enc);
1970 result = enc->priv->granule;
1971 GST_OBJECT_UNLOCK (enc);
1977 * gst_audio_encoder_set_perfect_timestamp:
1978 * @enc: a #GstAudioEncoder
1979 * @enabled: new state
1981 * Enable or disable encoder perfect output timestamp preference.
1988 gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
1991 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1993 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1995 GST_OBJECT_LOCK (enc);
1996 enc->priv->perfect_ts = enabled;
1997 GST_OBJECT_UNLOCK (enc);
2001 * gst_audio_encoder_get_perfect_timestamp:
2002 * @enc: a #GstAudioEncoder
2004 * Queries encoder perfect timestamp behaviour.
2006 * Returns: TRUE if pefect timestamp setting enabled.
2013 gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc)
2017 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
2019 GST_OBJECT_LOCK (enc);
2020 result = enc->priv->perfect_ts;
2021 GST_OBJECT_UNLOCK (enc);
2027 * gst_audio_encoder_set_hard_sync:
2028 * @enc: a #GstAudioEncoder
2029 * @enabled: new state
2031 * Sets encoder hard resync handling.
2038 gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc, gboolean enabled)
2040 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
2042 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
2044 GST_OBJECT_LOCK (enc);
2045 enc->priv->hard_resync = enabled;
2046 GST_OBJECT_UNLOCK (enc);
2050 * gst_audio_encoder_get_hard_sync:
2051 * @enc: a #GstAudioEncoder
2053 * Queries encoder's hard resync setting.
2055 * Returns: TRUE if hard resync is enabled.
2062 gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc)
2066 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
2068 GST_OBJECT_LOCK (enc);
2069 result = enc->priv->hard_resync;
2070 GST_OBJECT_UNLOCK (enc);
2076 * gst_audio_encoder_set_tolerance:
2077 * @enc: a #GstAudioEncoder
2078 * @tolerance: new tolerance
2080 * Configures encoder audio jitter tolerance threshold.
2087 gst_audio_encoder_set_tolerance (GstAudioEncoder * enc, gint64 tolerance)
2089 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
2091 GST_OBJECT_LOCK (enc);
2092 enc->priv->tolerance = tolerance;
2093 GST_OBJECT_UNLOCK (enc);
2097 * gst_audio_encoder_get_tolerance:
2098 * @enc: a #GstAudioEncoder
2100 * Queries current audio jitter tolerance threshold.
2102 * Returns: encoder audio jitter tolerance threshold.
2109 gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)
2113 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
2115 GST_OBJECT_LOCK (enc);
2116 result = enc->priv->tolerance;
2117 GST_OBJECT_UNLOCK (enc);
2123 * gst_audio_encoder_merge_tags:
2124 * @enc: a #GstAudioEncoder
2125 * @tags: a #GstTagList to merge
2126 * @mode: the #GstTagMergeMode to use
2128 * Adds tags to so-called pending tags, which will be processed
2129 * before pushing out data downstream.
2131 * Note that this is provided for convenience, and the subclass is
2132 * not required to use this and can still do tag handling on its own,
2133 * although it should be aware that baseclass already takes care
2134 * of the usual CODEC/AUDIO_CODEC tags.
2141 gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
2142 const GstTagList * tags, GstTagMergeMode mode)
2146 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
2147 g_return_if_fail (tags == NULL || GST_IS_TAG_LIST (tags));
2149 GST_OBJECT_LOCK (enc);
2151 GST_DEBUG_OBJECT (enc, "merging tags %" GST_PTR_FORMAT, tags);
2152 otags = enc->priv->tags;
2153 enc->priv->tags = gst_tag_list_merge (enc->priv->tags, tags, mode);
2155 gst_tag_list_free (otags);
2156 GST_OBJECT_UNLOCK (enc);