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 #define GST_USE_UNSTABLE_API
155 #include "gstaudioencoder.h"
156 #include <gst/base/gstadapter.h>
157 #include <gst/audio/audio.h>
158 #include <gst/pbutils/descriptions.h>
164 GST_DEBUG_CATEGORY_STATIC (gst_audio_encoder_debug);
165 #define GST_CAT_DEFAULT gst_audio_encoder_debug
167 #define GST_AUDIO_ENCODER_GET_PRIVATE(obj) \
168 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_ENCODER, \
169 GstAudioEncoderPrivate))
180 #define DEFAULT_PERFECT_TS FALSE
181 #define DEFAULT_GRANULE FALSE
182 #define DEFAULT_HARD_RESYNC FALSE
183 #define DEFAULT_TOLERANCE 40000000
185 typedef struct _GstAudioEncoderContext
191 gint frame_samples_min, frame_samples_max;
194 /* MT-protected (with LOCK) */
195 GstClockTime min_latency;
196 GstClockTime max_latency;
197 } GstAudioEncoderContext;
199 struct _GstAudioEncoderPrivate
201 /* activation status */
204 /* input base/first ts as basis for output ts;
205 * kept nearly constant for perfect_ts,
206 * otherwise resyncs to upstream ts */
207 GstClockTime base_ts;
208 /* corresponding base granulepos */
210 /* input samples processed and sent downstream so far (w.r.t. base_ts) */
213 /* currently collected sample data */
215 /* offset in adapter up to which already supplied to encoder */
217 /* mark outgoing discont */
219 /* to guess duration of drained data */
220 GstClockTime last_duration;
222 /* subclass provided data in processing round */
224 /* subclass gave all it could already */
226 /* subclass currently being forcibly drained */
229 /* output bps estimatation */
230 /* global in samples seen */
232 /* global bytes sent out */
235 /* context storage */
236 GstAudioEncoderContext ctx;
241 gboolean hard_resync;
246 /* pending serialized sink events, will be sent from finish_frame() */
247 GList *pending_events;
250 static void gst_audio_encoder_finalize (GObject * object);
251 static void gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full);
253 static void gst_audio_encoder_set_property (GObject * object,
254 guint prop_id, const GValue * value, GParamSpec * pspec);
255 static void gst_audio_encoder_get_property (GObject * object,
256 guint prop_id, GValue * value, GParamSpec * pspec);
258 static gboolean gst_audio_encoder_sink_activate_push (GstPad * pad,
261 static gboolean gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event);
262 static gboolean gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps);
263 static GstFlowReturn gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer);
264 static gboolean gst_audio_encoder_src_query (GstPad * pad, GstQuery * query);
265 static gboolean gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query);
266 static const GstQueryType *gst_audio_encoder_get_query_types (GstPad * pad);
267 static GstCaps *gst_audio_encoder_sink_getcaps (GstPad * pad);
270 do_init (GType gtype)
272 const GInterfaceInfo preset_interface_info = {
273 NULL, /* interface_init */
274 NULL, /* interface_finalize */
275 NULL /* interface_data */
278 g_type_add_interface_static (gtype, GST_TYPE_PRESET, &preset_interface_info);
281 GST_BOILERPLATE_FULL (GstAudioEncoder, gst_audio_encoder, GstElement,
282 GST_TYPE_ELEMENT, do_init);
285 gst_audio_encoder_class_init (GstAudioEncoderClass * klass)
287 GObjectClass *gobject_class;
289 gobject_class = G_OBJECT_CLASS (klass);
291 GST_DEBUG_CATEGORY_INIT (gst_audio_encoder_debug, "audioencoder", 0,
292 "audio encoder base class");
294 g_type_class_add_private (klass, sizeof (GstAudioEncoderPrivate));
296 gobject_class->set_property = gst_audio_encoder_set_property;
297 gobject_class->get_property = gst_audio_encoder_get_property;
299 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_encoder_finalize);
302 g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
303 g_param_spec_boolean ("perfect-timestamp", "Perfect Timestamps",
304 "Favour perfect timestamps over tracking upstream timestamps",
305 DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
306 g_object_class_install_property (gobject_class, PROP_GRANULE,
307 g_param_spec_boolean ("mark-granule", "Granule Marking",
308 "Apply granule semantics to buffer metadata (implies perfect-timestamp)",
309 DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
310 g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
311 g_param_spec_boolean ("hard-resync", "Hard Resync",
312 "Perform clipping and sample flushing upon discontinuity",
313 DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
314 g_object_class_install_property (gobject_class, PROP_TOLERANCE,
315 g_param_spec_int64 ("tolerance", "Tolerance",
316 "Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
317 0, G_MAXINT64, DEFAULT_TOLERANCE,
318 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
322 gst_audio_encoder_base_init (gpointer g_class)
327 gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
329 GstPadTemplate *pad_template;
331 GST_DEBUG_OBJECT (enc, "gst_audio_encoder_init");
333 enc->priv = GST_AUDIO_ENCODER_GET_PRIVATE (enc);
335 /* only push mode supported */
337 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
338 g_return_if_fail (pad_template != NULL);
339 enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
340 gst_pad_set_event_function (enc->sinkpad,
341 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_event));
342 gst_pad_set_setcaps_function (enc->sinkpad,
343 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_setcaps));
344 gst_pad_set_getcaps_function (enc->sinkpad,
345 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_getcaps));
346 gst_pad_set_query_function (enc->sinkpad,
347 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_query));
348 gst_pad_set_chain_function (enc->sinkpad,
349 GST_DEBUG_FUNCPTR (gst_audio_encoder_chain));
350 gst_pad_set_activatepush_function (enc->sinkpad,
351 GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_activate_push));
352 gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
354 GST_DEBUG_OBJECT (enc, "sinkpad created");
356 /* and we don't mind upstream traveling stuff that much ... */
358 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
359 g_return_if_fail (pad_template != NULL);
360 enc->srcpad = gst_pad_new_from_template (pad_template, "src");
361 gst_pad_set_query_function (enc->srcpad,
362 GST_DEBUG_FUNCPTR (gst_audio_encoder_src_query));
363 gst_pad_set_query_type_function (enc->srcpad,
364 GST_DEBUG_FUNCPTR (gst_audio_encoder_get_query_types));
365 gst_pad_use_fixed_caps (enc->srcpad);
366 gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
367 GST_DEBUG_OBJECT (enc, "src created");
369 enc->priv->adapter = gst_adapter_new ();
371 g_static_rec_mutex_init (&enc->stream_lock);
373 /* property default */
374 enc->priv->granule = DEFAULT_GRANULE;
375 enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
376 enc->priv->hard_resync = DEFAULT_HARD_RESYNC;
377 enc->priv->tolerance = DEFAULT_TOLERANCE;
380 gst_audio_encoder_reset (enc, TRUE);
381 GST_DEBUG_OBJECT (enc, "init ok");
385 gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
387 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
389 GST_LOG_OBJECT (enc, "reset full %d", full);
392 enc->priv->active = FALSE;
393 enc->priv->samples_in = 0;
394 enc->priv->bytes_out = 0;
395 gst_audio_info_clear (&enc->priv->ctx.info);
396 memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
399 gst_tag_list_free (enc->priv->tags);
400 enc->priv->tags = NULL;
402 g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
403 g_list_free (enc->priv->pending_events);
404 enc->priv->pending_events = NULL;
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_AUDIO_ENCODER_STREAM_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_static_rec_mutex_free (&enc->stream_lock);
430 G_OBJECT_CLASS (parent_class)->finalize (object);
434 * gst_audio_encoder_finish_frame:
435 * @enc: a #GstAudioEncoder
436 * @buffer: encoded data
437 * @samples: number of samples (per channel) represented by encoded data
439 * Collects encoded data and/or pushes encoded data downstream.
440 * Source pad caps must be set when this is called. Depending on the nature
441 * of the (framing of) the format, subclass can decide whether to push
442 * encoded data directly or to collect various "frames" in a single buffer.
443 * Note that the latter behaviour is recommended whenever the format is allowed,
444 * as it incurs no additional latency and avoids otherwise generating a
445 * a multitude of (small) output buffers. If not explicitly pushed,
446 * any available encoded data is pushed at the end of each processing cycle,
447 * i.e. which encodes as much data as available input data allows.
449 * If @samples < 0, then best estimate is all samples provided to encoder
450 * (subclass) so far. @buf may be NULL, in which case next number of @samples
451 * are considered discarded, e.g. as a result of discontinuous transmission,
452 * and a discontinuity is marked (note that @buf == NULL => push == TRUE).
454 * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
459 gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
462 GstAudioEncoderClass *klass;
463 GstAudioEncoderPrivate *priv;
464 GstAudioEncoderContext *ctx;
465 GstFlowReturn ret = GST_FLOW_OK;
467 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
469 ctx = &enc->priv->ctx;
471 /* subclass should know what it is producing by now */
472 g_return_val_if_fail (GST_PAD_CAPS (enc->srcpad) != NULL, GST_FLOW_ERROR);
473 /* subclass should not hand us no data */
474 g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
477 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
479 if (G_UNLIKELY (enc->priv->tags)) {
482 /* add codec info to pending tags */
483 tags = enc->priv->tags;
484 /* no more pending */
485 enc->priv->tags = NULL;
486 gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_CODEC,
487 GST_PAD_CAPS (enc->srcpad));
488 gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_AUDIO_CODEC,
489 GST_PAD_CAPS (enc->srcpad));
490 GST_DEBUG_OBJECT (enc, "sending tags %" GST_PTR_FORMAT, tags);
491 gst_element_found_tags_for_pad (GST_ELEMENT (enc), enc->srcpad, tags);
494 GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
495 buf ? GST_BUFFER_SIZE (buf) : -1, samples);
497 /* mark subclass still alive and providing */
498 priv->got_data = TRUE;
500 if (priv->pending_events) {
501 GList *pending_events, *l;
503 pending_events = priv->pending_events;
504 priv->pending_events = NULL;
506 GST_DEBUG_OBJECT (enc, "Pushing pending events");
507 for (l = priv->pending_events; l; l = l->next)
508 gst_pad_push_event (enc->srcpad, l->data);
509 g_list_free (pending_events);
512 /* remove corresponding samples from input */
514 samples = (enc->priv->offset / ctx->info.bpf);
516 if (G_LIKELY (samples)) {
517 /* track upstream ts if so configured */
518 if (!enc->priv->perfect_ts) {
519 guint64 ts, distance;
521 ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
522 g_assert (distance % ctx->info.bpf == 0);
523 distance /= ctx->info.bpf;
524 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
525 GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
526 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
527 GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
528 /* when draining adapter might be empty and no ts to offer */
529 if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
530 GstClockTimeDiff diff;
531 GstClockTime old_ts, next_ts;
533 /* passed into another buffer;
534 * mild check for discontinuity and only mark if so */
536 gst_util_uint64_scale (distance, GST_SECOND, ctx->info.rate);
537 old_ts = priv->base_ts +
538 gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->info.rate);
539 diff = GST_CLOCK_DIFF (next_ts, old_ts);
540 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
541 /* only mark discontinuity if beyond tolerance */
542 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
543 diff > enc->priv->tolerance)) {
544 GST_DEBUG_OBJECT (enc, "marked discont");
545 priv->discont = TRUE;
547 if (diff > GST_SECOND / ctx->info.rate / 2 ||
548 diff < -GST_SECOND / ctx->info.rate / 2) {
549 GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
550 " at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
551 /* re-sync to upstream ts */
553 priv->samples = distance;
555 GST_LOG_OBJECT (enc, "new upstream ts only introduces jitter");
559 /* advance sample view */
560 if (G_UNLIKELY (samples * ctx->info.bpf > priv->offset)) {
561 if (G_LIKELY (!priv->force)) {
562 /* no way we can let this pass */
563 g_assert_not_reached ();
568 if (samples * ctx->info.bpf >= gst_adapter_available (priv->adapter))
569 gst_adapter_clear (priv->adapter);
571 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
574 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
575 priv->offset -= samples * ctx->info.bpf;
576 /* avoid subsequent stray prev_ts */
577 if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
578 gst_adapter_clear (priv->adapter);
580 /* sample count advanced below after buffer handling */
584 if (G_LIKELY (buf)) {
585 GST_LOG_OBJECT (enc, "taking %d bytes for output", GST_BUFFER_SIZE (buf));
586 buf = gst_buffer_make_metadata_writable (buf);
589 gst_buffer_set_caps (buf, GST_PAD_CAPS (enc->srcpad));
590 if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
591 /* FIXME ? lookahead could lead to weird ts and duration ?
592 * (particularly if not in perfect mode) */
593 /* mind sample rounding and produce perfect output */
594 GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
595 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
597 GST_DEBUG_OBJECT (enc, "out samples %d", samples);
598 if (G_LIKELY (samples > 0)) {
599 priv->samples += samples;
600 GST_BUFFER_DURATION (buf) = priv->base_ts +
601 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
602 ctx->info.rate) - GST_BUFFER_TIMESTAMP (buf);
603 priv->last_duration = GST_BUFFER_DURATION (buf);
605 /* duration forecast in case of handling remainder;
606 * the last one is probably like the previous one ... */
607 GST_BUFFER_DURATION (buf) = priv->last_duration;
609 if (priv->base_gp >= 0) {
611 /* FIXME: in longer run, muxer should take care of this ... */
612 /* offset_end = granulepos for ogg muxer */
613 GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples -
614 enc->priv->ctx.lookahead;
615 /* offset = timestamp corresponding to granulepos for ogg muxer */
616 GST_BUFFER_OFFSET (buf) =
617 GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf),
620 GST_BUFFER_OFFSET (buf) = priv->bytes_out;
621 GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf);
625 priv->bytes_out += GST_BUFFER_SIZE (buf);
627 if (G_UNLIKELY (priv->discont)) {
628 GST_LOG_OBJECT (enc, "marking discont");
629 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
630 priv->discont = FALSE;
633 if (klass->pre_push) {
634 /* last chance for subclass to do some dirty stuff */
635 ret = klass->pre_push (enc, &buf);
636 if (ret != GST_FLOW_OK || !buf) {
637 GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
638 gst_flow_get_name (ret), buf);
640 gst_buffer_unref (buf);
645 GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
646 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
647 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
648 GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
650 ret = gst_pad_push (enc->srcpad, buf);
651 GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
653 /* merely advance samples, most work for that already done above */
654 priv->samples += samples;
658 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
665 GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
666 ("received more encoded samples %d than provided %d",
667 samples, priv->offset / ctx->info.bpf), (NULL));
669 gst_buffer_unref (buf);
670 ret = GST_FLOW_ERROR;
675 /* adapter tracking idea:
676 * - start of adapter corresponds with what has already been encoded
677 * (i.e. really returned by encoder subclass)
678 * - start + offset is what needs to be fed to subclass next */
680 gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
682 GstAudioEncoderClass *klass;
683 GstAudioEncoderPrivate *priv;
684 GstAudioEncoderContext *ctx;
687 GstFlowReturn ret = GST_FLOW_OK;
689 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
691 g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
694 ctx = &enc->priv->ctx;
696 while (ret == GST_FLOW_OK) {
699 av = gst_adapter_available (priv->adapter);
701 g_assert (priv->offset <= av);
705 ctx->frame_samples_min >
706 0 ? ctx->frame_samples_min * ctx->info.bpf : av;
707 GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d", av, need,
710 if ((need > av) || !av) {
711 if (G_UNLIKELY (force)) {
721 if (ctx->frame_samples_max > 0)
722 need = MIN (av, ctx->frame_samples_max * ctx->info.bpf);
724 if (ctx->frame_samples_min == ctx->frame_samples_max) {
725 /* if we have some extra metadata,
726 * provide for integer multiple of frames to allow for better granularity
728 if (ctx->frame_samples_min > 0 && need) {
729 if (ctx->frame_max > 1)
730 need = need * MIN ((av / need), ctx->frame_max);
731 else if (ctx->frame_max == 0)
732 need = need * (av / need);
737 buf = gst_buffer_new ();
738 GST_BUFFER_DATA (buf) = (guint8 *)
739 gst_adapter_peek (priv->adapter, priv->offset + need) + priv->offset;
740 GST_BUFFER_SIZE (buf) = need;
743 GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
746 /* mark this already as consumed,
747 * which it should be when subclass gives us data in exchange for samples */
748 priv->offset += need;
749 priv->samples_in += need / ctx->info.bpf;
751 priv->got_data = FALSE;
752 ret = klass->handle_frame (enc, buf);
755 gst_buffer_unref (buf);
757 /* no data to feed, no leftover provided, then bail out */
758 if (G_UNLIKELY (!buf && !priv->got_data)) {
759 priv->drained = TRUE;
760 GST_LOG_OBJECT (enc, "no more data drained from subclass");
769 gst_audio_encoder_drain (GstAudioEncoder * enc)
771 if (enc->priv->drained)
774 return gst_audio_encoder_push_buffers (enc, TRUE);
778 gst_audio_encoder_set_base_gp (GstAudioEncoder * enc)
782 if (!enc->priv->granule)
785 /* use running time for granule */
786 /* incoming data is clipped, so a valid input should yield a valid output */
787 ts = gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME,
789 if (GST_CLOCK_TIME_IS_VALID (ts)) {
791 GST_CLOCK_TIME_TO_FRAMES (enc->priv->base_ts, enc->priv->ctx.info.rate);
792 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, enc->priv->base_gp);
794 /* should reasonably have a valid base,
795 * otherwise start at 0 if we did not already start there earlier */
796 if (enc->priv->base_gp < 0) {
797 enc->priv->base_gp = 0;
798 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
805 gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
807 GstAudioEncoder *enc;
808 GstAudioEncoderPrivate *priv;
809 GstAudioEncoderContext *ctx;
810 GstFlowReturn ret = GST_FLOW_OK;
813 enc = GST_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
816 ctx = &enc->priv->ctx;
818 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
820 /* should know what is coming by now */
825 "received buffer of size %d with ts %" GST_TIME_FORMAT
826 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
827 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
828 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
830 /* input shoud be whole number of sample frames */
831 if (GST_BUFFER_SIZE (buffer) % ctx->info.bpf)
834 #ifndef GST_DISABLE_GST_DEBUG
836 GstClockTime duration;
837 GstClockTimeDiff diff;
839 /* verify buffer duration */
840 duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND,
841 ctx->info.rate * ctx->info.bpf);
842 diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
843 if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
844 (diff > GST_SECOND / ctx->info.rate / 2 ||
845 diff < -GST_SECOND / ctx->info.rate / 2)) {
846 GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
847 GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
848 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
849 GST_TIME_ARGS (duration));
854 discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
855 if (G_UNLIKELY (discont)) {
856 GST_LOG_OBJECT (buffer, "marked discont");
857 enc->priv->discont = discont;
860 /* clip to segment */
861 /* NOTE: slightly painful linking -laudio only for this one ... */
862 buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->info.rate,
864 if (G_UNLIKELY (!buffer)) {
865 GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
870 "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
871 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
872 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
873 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
875 if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
876 priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
877 GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
878 GST_TIME_ARGS (priv->base_ts));
879 gst_audio_encoder_set_base_gp (enc);
882 /* check for continuity;
883 * checked elsewhere in non-perfect case */
884 if (enc->priv->perfect_ts) {
885 GstClockTimeDiff diff = 0;
886 GstClockTime next_ts = 0;
888 if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
889 GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
892 samples = priv->samples +
893 gst_adapter_available (priv->adapter) / ctx->info.bpf;
894 next_ts = priv->base_ts +
895 gst_util_uint64_scale (samples, GST_SECOND, ctx->info.rate);
896 GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
897 " samples past base_ts %" GST_TIME_FORMAT
898 ", expected ts %" GST_TIME_FORMAT, samples,
899 GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
900 diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
901 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
902 /* if within tolerance,
903 * discard buffer ts and carry on producing perfect stream,
904 * otherwise clip or resync to ts */
905 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
906 diff > enc->priv->tolerance)) {
907 GST_DEBUG_OBJECT (enc, "marked discont");
912 /* do some fancy tweaking in hard resync case */
913 if (discont && enc->priv->hard_resync) {
917 GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
918 GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
921 GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->info.rate) * ctx->info.bpf;
922 if (diff_bytes >= GST_BUFFER_SIZE (buffer)) {
923 gst_buffer_unref (buffer);
926 buffer = gst_buffer_make_metadata_writable (buffer);
927 GST_BUFFER_DATA (buffer) += diff_bytes;
928 GST_BUFFER_SIZE (buffer) -= diff_bytes;
930 GST_BUFFER_TIMESTAMP (buffer) += diff;
931 /* care even less about duration after this */
933 /* drain stuff prior to resync */
934 gst_audio_encoder_drain (enc);
938 priv->base_ts += diff;
939 gst_audio_encoder_set_base_gp (enc);
940 priv->discont |= discont;
943 gst_adapter_push (enc->priv->adapter, buffer);
944 /* new stuff, so we can push subclass again */
945 enc->priv->drained = FALSE;
947 ret = gst_audio_encoder_push_buffers (enc, FALSE);
950 GST_LOG_OBJECT (enc, "chain leaving");
952 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
959 GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
960 ("encoder not initialized"));
961 gst_buffer_unref (buffer);
962 ret = GST_FLOW_NOT_NEGOTIATED;
967 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
968 ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
970 gst_buffer_unref (buffer);
971 ret = GST_FLOW_ERROR;
977 audio_info_is_equal (GstAudioInfo * from, GstAudioInfo * to)
981 if (from->finfo == NULL || to->finfo == NULL)
983 if (GST_AUDIO_INFO_FORMAT (from) != GST_AUDIO_INFO_FORMAT (to))
985 if (GST_AUDIO_INFO_RATE (from) != GST_AUDIO_INFO_RATE (to))
987 if (GST_AUDIO_INFO_CHANNELS (from) != GST_AUDIO_INFO_CHANNELS (to))
989 if (GST_AUDIO_INFO_CHANNELS (from) > 64)
991 return memcmp (from->position, to->position,
992 GST_AUDIO_INFO_CHANNELS (from) * sizeof (to->position[0]));
996 gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
998 GstAudioEncoder *enc;
999 GstAudioEncoderClass *klass;
1000 GstAudioEncoderContext *ctx;
1001 GstAudioInfo *state, *old_state;
1002 gboolean res = TRUE, changed = FALSE;
1005 enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
1006 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1008 /* subclass must do something here ... */
1009 g_return_val_if_fail (klass->set_format != NULL, FALSE);
1011 ctx = &enc->priv->ctx;
1014 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1016 GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
1018 if (!gst_caps_is_fixed (caps))
1021 /* adjust ts tracking to new sample rate */
1022 old_rate = GST_AUDIO_INFO_RATE (state);
1023 if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
1024 enc->priv->base_ts +=
1025 GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
1026 enc->priv->samples = 0;
1029 old_state = gst_audio_info_copy (state);
1030 if (!gst_audio_info_from_caps (state, caps))
1033 changed = !audio_info_is_equal (state, old_state);
1034 gst_audio_info_free (old_state);
1037 GstClockTime old_min_latency;
1038 GstClockTime old_max_latency;
1040 /* drain any pending old data stuff */
1041 gst_audio_encoder_drain (enc);
1043 /* context defaults */
1044 enc->priv->ctx.frame_samples_min = 0;
1045 enc->priv->ctx.frame_samples_max = 0;
1046 enc->priv->ctx.frame_max = 0;
1047 enc->priv->ctx.lookahead = 0;
1049 /* element might report latency */
1050 GST_OBJECT_LOCK (enc);
1051 old_min_latency = ctx->min_latency;
1052 old_max_latency = ctx->max_latency;
1053 GST_OBJECT_UNLOCK (enc);
1055 if (klass->set_format)
1056 res = klass->set_format (enc, state);
1058 /* notify if new latency */
1059 GST_OBJECT_LOCK (enc);
1060 if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
1061 (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
1062 GST_OBJECT_UNLOCK (enc);
1063 /* post latency message on the bus */
1064 gst_element_post_message (GST_ELEMENT (enc),
1065 gst_message_new_latency (GST_OBJECT (enc)));
1066 GST_OBJECT_LOCK (enc);
1068 GST_OBJECT_UNLOCK (enc);
1070 GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
1075 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1082 GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
1089 * gst_audio_encoder_proxy_getcaps:
1090 * @enc: a #GstAudioEncoder
1091 * @caps: initial caps
1093 * Returns caps that express @caps (or sink template caps if @caps == NULL)
1094 * restricted to channel/rate combinations supported by downstream elements
1097 * Returns: a #GstCaps owned by caller
1102 gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps)
1104 const GstCaps *templ_caps;
1105 GstCaps *allowed = NULL;
1106 GstCaps *fcaps, *filter_caps;
1109 /* we want to be able to communicate to upstream elements like audioconvert
1110 * and audioresample any rate/channel restrictions downstream (e.g. muxer
1111 * only accepting certain sample rates) */
1112 templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad);
1113 allowed = gst_pad_get_allowed_caps (enc->srcpad);
1114 if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
1115 fcaps = gst_caps_copy (templ_caps);
1119 GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
1120 GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
1122 filter_caps = gst_caps_new_empty ();
1124 for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
1127 q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
1129 /* pick rate + channel fields from allowed caps */
1130 for (j = 0; j < gst_caps_get_size (allowed); j++) {
1131 const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
1135 s = gst_structure_id_empty_new (q_name);
1136 if ((val = gst_structure_get_value (allowed_s, "rate")))
1137 gst_structure_set_value (s, "rate", val);
1138 if ((val = gst_structure_get_value (allowed_s, "channels")))
1139 gst_structure_set_value (s, "channels", val);
1140 /* following might also make sense for some encoded formats,
1142 if ((val = gst_structure_get_value (allowed_s, "width")))
1143 gst_structure_set_value (s, "width", val);
1144 if ((val = gst_structure_get_value (allowed_s, "depth")))
1145 gst_structure_set_value (s, "depth", val);
1146 if ((val = gst_structure_get_value (allowed_s, "endianness")))
1147 gst_structure_set_value (s, "endianness", val);
1148 if ((val = gst_structure_get_value (allowed_s, "signed")))
1149 gst_structure_set_value (s, "signed", val);
1150 if ((val = gst_structure_get_value (allowed_s, "channel-positions")))
1151 gst_structure_set_value (s, "channel-positions", val);
1153 gst_caps_merge_structure (filter_caps, s);
1157 fcaps = gst_caps_intersect (filter_caps, templ_caps);
1158 gst_caps_unref (filter_caps);
1161 gst_caps_replace (&allowed, NULL);
1163 GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
1169 gst_audio_encoder_sink_getcaps (GstPad * pad)
1171 GstAudioEncoder *enc;
1172 GstAudioEncoderClass *klass;
1175 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1176 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1177 g_assert (pad == enc->sinkpad);
1180 caps = klass->getcaps (enc);
1182 caps = gst_audio_encoder_proxy_getcaps (enc, NULL);
1183 gst_object_unref (enc);
1185 GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
1191 gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
1193 GstAudioEncoderClass *klass;
1194 gboolean handled = FALSE;
1196 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1198 switch (GST_EVENT_TYPE (event)) {
1199 case GST_EVENT_NEWSEGMENT:
1202 gdouble rate, arate;
1203 gint64 start, stop, time;
1206 gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
1207 &start, &stop, &time);
1209 if (format == GST_FORMAT_TIME) {
1210 GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
1211 " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
1212 ", rate %g, applied_rate %g",
1213 GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
1216 GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
1217 " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
1218 ", rate %g, applied_rate %g", start, stop, time, rate, arate);
1219 GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
1223 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1224 /* finish current segment */
1225 gst_audio_encoder_drain (enc);
1226 /* reset partially for new segment */
1227 gst_audio_encoder_reset (enc, FALSE);
1228 /* and follow along with segment */
1229 gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
1230 format, start, stop, time);
1231 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1235 case GST_EVENT_FLUSH_START:
1238 case GST_EVENT_FLUSH_STOP:
1239 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1240 /* discard any pending stuff */
1241 /* TODO route through drain ?? */
1242 if (!enc->priv->drained && klass->flush)
1244 /* and get (re)set for the sequel */
1245 gst_audio_encoder_reset (enc, FALSE);
1247 g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
1248 g_list_free (enc->priv->pending_events);
1249 enc->priv->pending_events = NULL;
1250 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1255 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1256 gst_audio_encoder_drain (enc);
1257 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1264 gst_event_parse_tag (event, &tags);
1265 tags = gst_tag_list_copy (tags);
1266 gst_event_unref (event);
1267 gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
1268 gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
1269 event = gst_event_new_tag (tags);
1271 GST_OBJECT_LOCK (enc);
1272 enc->priv->pending_events =
1273 g_list_append (enc->priv->pending_events, event);
1274 GST_OBJECT_UNLOCK (enc);
1287 gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
1289 GstAudioEncoder *enc;
1290 GstAudioEncoderClass *klass;
1291 gboolean handled = FALSE;
1292 gboolean ret = TRUE;
1294 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1295 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1297 GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
1298 GST_EVENT_TYPE_NAME (event));
1301 handled = klass->event (enc, event);
1304 handled = gst_audio_encoder_sink_eventfunc (enc, event);
1307 /* Forward non-serialized events and EOS/FLUSH_STOP immediately.
1308 * For EOS this is required because no buffer or serialized event
1309 * will come after EOS and nothing could trigger another
1310 * _finish_frame() call.
1312 * For FLUSH_STOP this is required because it is expected
1313 * to be forwarded immediately and no buffers are queued anyway.
1315 if (!GST_EVENT_IS_SERIALIZED (event)
1316 || GST_EVENT_TYPE (event) == GST_EVENT_EOS
1317 || GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
1318 ret = gst_pad_event_default (pad, event);
1320 GST_AUDIO_ENCODER_STREAM_LOCK (enc);
1321 enc->priv->pending_events =
1322 g_list_append (enc->priv->pending_events, event);
1323 GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
1328 GST_DEBUG_OBJECT (enc, "event handled");
1330 gst_object_unref (enc);
1335 gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
1337 gboolean res = TRUE;
1338 GstAudioEncoder *enc;
1340 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1342 switch (GST_QUERY_TYPE (query)) {
1343 case GST_QUERY_FORMATS:
1345 gst_query_set_formats (query, 3,
1346 GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
1350 case GST_QUERY_CONVERT:
1352 GstFormat src_fmt, dest_fmt;
1353 gint64 src_val, dest_val;
1355 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1356 if (!(res = gst_audio_info_convert (&enc->priv->ctx.info,
1357 src_fmt, src_val, dest_fmt, &dest_val)))
1359 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1363 res = gst_pad_query_default (pad, query);
1368 gst_object_unref (enc);
1372 static const GstQueryType *
1373 gst_audio_encoder_get_query_types (GstPad * pad)
1375 static const GstQueryType gst_audio_encoder_src_query_types[] = {
1383 return gst_audio_encoder_src_query_types;
1387 * gst_audio_encoded_audio_convert:
1388 * @fmt: audio format of the encoded audio
1389 * @bytes: number of encoded bytes
1390 * @samples: number of encoded samples
1391 * @src_format: source format
1392 * @src_value: source value
1393 * @dest_format: destination format
1394 * @dest_value: destination format
1396 * Helper function to convert @src_value in @src_format to @dest_value in
1397 * @dest_format for encoded audio data. Conversion is possible between
1398 * BYTE and TIME format by using estimated bitrate based on
1399 * @samples and @bytes (and @fmt).
1403 /* FIXME: make gst_audio_encoded_audio_convert() public? */
1405 gst_audio_encoded_audio_convert (GstAudioInfo * fmt,
1406 gint64 bytes, gint64 samples, GstFormat src_format,
1407 gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
1409 gboolean res = FALSE;
1411 g_return_val_if_fail (dest_format != NULL, FALSE);
1412 g_return_val_if_fail (dest_value != NULL, FALSE);
1414 if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
1417 *dest_value = src_value;
1421 if (samples == 0 || bytes == 0 || fmt->rate == 0) {
1422 GST_DEBUG ("not enough metadata yet to convert");
1428 switch (src_format) {
1429 case GST_FORMAT_BYTES:
1430 switch (*dest_format) {
1431 case GST_FORMAT_TIME:
1432 *dest_value = gst_util_uint64_scale (src_value,
1433 GST_SECOND * samples, bytes);
1440 case GST_FORMAT_TIME:
1441 switch (*dest_format) {
1442 case GST_FORMAT_BYTES:
1443 *dest_value = gst_util_uint64_scale (src_value, bytes,
1444 samples * GST_SECOND);
1459 /* FIXME ? are any of these queries (other than latency) an encoder's business
1460 * also, the conversion stuff might seem to make sense, but seems to not mind
1461 * segment stuff etc at all
1462 * Supposedly that's backward compatibility ... */
1464 gst_audio_encoder_src_query (GstPad * pad, GstQuery * query)
1466 GstAudioEncoder *enc;
1468 gboolean res = FALSE;
1470 enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
1471 peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
1473 GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
1475 switch (GST_QUERY_TYPE (query)) {
1476 case GST_QUERY_POSITION:
1478 GstFormat fmt, req_fmt;
1481 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1482 GST_LOG_OBJECT (enc, "returning peer response");
1487 GST_LOG_OBJECT (enc, "no peer");
1491 gst_query_parse_position (query, &req_fmt, NULL);
1492 fmt = GST_FORMAT_TIME;
1493 if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
1496 if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) {
1497 gst_query_set_position (query, req_fmt, val);
1501 case GST_QUERY_DURATION:
1503 GstFormat fmt, req_fmt;
1506 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1507 GST_LOG_OBJECT (enc, "returning peer response");
1512 GST_LOG_OBJECT (enc, "no peer");
1516 gst_query_parse_duration (query, &req_fmt, NULL);
1517 fmt = GST_FORMAT_TIME;
1518 if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
1521 if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) {
1522 gst_query_set_duration (query, req_fmt, val);
1526 case GST_QUERY_FORMATS:
1528 gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
1532 case GST_QUERY_CONVERT:
1534 GstFormat src_fmt, dest_fmt;
1535 gint64 src_val, dest_val;
1537 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1538 if (!(res = gst_audio_encoded_audio_convert (&enc->priv->ctx.info,
1539 enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
1540 &dest_fmt, &dest_val)))
1542 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1545 case GST_QUERY_LATENCY:
1547 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1549 GstClockTime min_latency, max_latency;
1551 gst_query_parse_latency (query, &live, &min_latency, &max_latency);
1552 GST_DEBUG_OBJECT (enc, "Peer latency: live %d, min %"
1553 GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
1554 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
1556 GST_OBJECT_LOCK (enc);
1557 /* add our latency */
1558 if (min_latency != -1)
1559 min_latency += enc->priv->ctx.min_latency;
1560 if (max_latency != -1)
1561 max_latency += enc->priv->ctx.max_latency;
1562 GST_OBJECT_UNLOCK (enc);
1564 gst_query_set_latency (query, live, min_latency, max_latency);
1569 res = gst_pad_query_default (pad, query);
1573 gst_object_unref (peerpad);
1578 gst_audio_encoder_set_property (GObject * object, guint prop_id,
1579 const GValue * value, GParamSpec * pspec)
1581 GstAudioEncoder *enc;
1583 enc = GST_AUDIO_ENCODER (object);
1586 case PROP_PERFECT_TS:
1587 if (enc->priv->granule && !g_value_get_boolean (value))
1588 GST_WARNING_OBJECT (enc, "perfect-timestamp can not be set FALSE "
1589 "while granule handling is enabled");
1591 enc->priv->perfect_ts = g_value_get_boolean (value);
1593 case PROP_HARD_RESYNC:
1594 enc->priv->hard_resync = g_value_get_boolean (value);
1596 case PROP_TOLERANCE:
1597 enc->priv->tolerance = g_value_get_int64 (value);
1600 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1606 gst_audio_encoder_get_property (GObject * object, guint prop_id,
1607 GValue * value, GParamSpec * pspec)
1609 GstAudioEncoder *enc;
1611 enc = GST_AUDIO_ENCODER (object);
1614 case PROP_PERFECT_TS:
1615 g_value_set_boolean (value, enc->priv->perfect_ts);
1618 g_value_set_boolean (value, enc->priv->granule);
1620 case PROP_HARD_RESYNC:
1621 g_value_set_boolean (value, enc->priv->hard_resync);
1623 case PROP_TOLERANCE:
1624 g_value_set_int64 (value, enc->priv->tolerance);
1627 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1633 gst_audio_encoder_activate (GstAudioEncoder * enc, gboolean active)
1635 GstAudioEncoderClass *klass;
1636 gboolean result = FALSE;
1638 klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
1640 g_return_val_if_fail (!enc->priv->granule || enc->priv->perfect_ts, FALSE);
1642 GST_DEBUG_OBJECT (enc, "activate %d", active);
1646 if (enc->priv->tags)
1647 gst_tag_list_free (enc->priv->tags);
1648 enc->priv->tags = gst_tag_list_new ();
1650 if (!enc->priv->active && klass->start)
1651 result = klass->start (enc);
1653 /* We must make sure streaming has finished before resetting things
1654 * and calling the ::stop vfunc */
1655 GST_PAD_STREAM_LOCK (enc->sinkpad);
1656 GST_PAD_STREAM_UNLOCK (enc->sinkpad);
1658 if (enc->priv->active && klass->stop)
1659 result = klass->stop (enc);
1662 gst_audio_encoder_reset (enc, TRUE);
1664 GST_DEBUG_OBJECT (enc, "activate return: %d", result);
1670 gst_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
1672 gboolean result = TRUE;
1673 GstAudioEncoder *enc;
1675 enc = GST_AUDIO_ENCODER (gst_pad_get_parent (pad));
1677 GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
1679 result = gst_audio_encoder_activate (enc, active);
1682 enc->priv->active = active;
1684 GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
1686 gst_object_unref (enc);
1691 * gst_audio_encoder_get_audio_info:
1692 * @enc: a #GstAudioEncoder
1694 * Returns: a #GstAudioInfo describing the input audio format
1699 gst_audio_encoder_get_audio_info (GstAudioEncoder * enc)
1701 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), NULL);
1703 return &enc->priv->ctx.info;
1707 * gst_audio_encoder_set_frame_samples_min:
1708 * @enc: a #GstAudioEncoder
1709 * @num: number of samples per frame
1711 * Sets number of samples (per channel) subclass needs to be handed,
1712 * at least or will be handed all available if 0.
1717 gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num)
1719 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1721 enc->priv->ctx.frame_samples_min = num;
1725 * gst_audio_encoder_get_frame_samples_min:
1726 * @enc: a #GstAudioEncoder
1728 * Returns: currently minimum requested samples per frame
1733 gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc)
1735 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1737 return enc->priv->ctx.frame_samples_min;
1741 * gst_audio_encoder_set_frame_samples_max:
1742 * @enc: a #GstAudioEncoder
1743 * @num: number of samples per frame
1745 * Sets number of samples (per channel) subclass needs to be handed,
1746 * at most or will be handed all available if 0.
1751 gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num)
1753 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1755 enc->priv->ctx.frame_samples_max = num;
1759 * gst_audio_encoder_get_frame_samples_min:
1760 * @enc: a #GstAudioEncoder
1762 * Returns: currently maximum requested samples per frame
1767 gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc)
1769 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1771 return enc->priv->ctx.frame_samples_max;
1775 * gst_audio_encoder_set_frame_max:
1776 * @enc: a #GstAudioEncoder
1777 * @num: number of frames
1779 * Sets max number of frames accepted at once (assumed minimally 1).
1780 * Requires @frame_samples_min and @frame_samples_max to be the equal.
1785 gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num)
1787 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1789 enc->priv->ctx.frame_max = num;
1793 * gst_audio_encoder_get_frame_max:
1794 * @enc: a #GstAudioEncoder
1796 * Returns: currently configured maximum handled frames
1801 gst_audio_encoder_get_frame_max (GstAudioEncoder * enc)
1803 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1805 return enc->priv->ctx.frame_max;
1809 * gst_audio_encoder_set_lookahead:
1810 * @enc: a #GstAudioEncoder
1813 * Sets encoder lookahead (in units of input rate samples)
1818 gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num)
1820 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1822 enc->priv->ctx.lookahead = num;
1826 * gst_audio_encoder_get_lookahead:
1827 * @enc: a #GstAudioEncoder
1829 * Returns: currently configured encoder lookahead
1832 gst_audio_encoder_get_lookahead (GstAudioEncoder * enc)
1834 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
1836 return enc->priv->ctx.lookahead;
1840 * gst_audio_encoder_set_latency:
1841 * @enc: a #GstAudioEncoder
1842 * @min: minimum latency
1843 * @max: maximum latency
1845 * Sets encoder latency.
1850 gst_audio_encoder_set_latency (GstAudioEncoder * enc,
1851 GstClockTime min, GstClockTime max)
1853 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1855 GST_OBJECT_LOCK (enc);
1856 enc->priv->ctx.min_latency = min;
1857 enc->priv->ctx.max_latency = max;
1858 GST_OBJECT_UNLOCK (enc);
1862 * gst_audio_encoder_get_latency:
1863 * @enc: a #GstAudioEncoder
1864 * @min: (out) (allow-none): a pointer to storage to hold minimum latency
1865 * @max: (out) (allow-none): a pointer to storage to hold maximum latency
1867 * Sets the variables pointed to by @min and @max to the currently configured
1873 gst_audio_encoder_get_latency (GstAudioEncoder * enc,
1874 GstClockTime * min, GstClockTime * max)
1876 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1878 GST_OBJECT_LOCK (enc);
1880 *min = enc->priv->ctx.min_latency;
1882 *max = enc->priv->ctx.max_latency;
1883 GST_OBJECT_UNLOCK (enc);
1887 * gst_audio_encoder_set_mark_granule:
1888 * @enc: a #GstAudioEncoder
1889 * @enabled: new state
1891 * Enable or disable encoder granule handling.
1898 gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc, gboolean enabled)
1900 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1902 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1904 GST_OBJECT_LOCK (enc);
1905 enc->priv->granule = enabled;
1906 GST_OBJECT_UNLOCK (enc);
1910 * gst_audio_encoder_get_mark_granule:
1911 * @enc: a #GstAudioEncoder
1913 * Queries if the encoder will handle granule marking.
1915 * Returns: TRUE if granule marking is enabled.
1922 gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc)
1926 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1928 GST_OBJECT_LOCK (enc);
1929 result = enc->priv->granule;
1930 GST_OBJECT_UNLOCK (enc);
1936 * gst_audio_encoder_set_perfect_timestamp:
1937 * @enc: a #GstAudioEncoder
1938 * @enabled: new state
1940 * Enable or disable encoder perfect output timestamp preference.
1947 gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
1950 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
1952 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1954 GST_OBJECT_LOCK (enc);
1955 enc->priv->perfect_ts = enabled;
1956 GST_OBJECT_UNLOCK (enc);
1960 * gst_audio_encoder_get_perfect_timestamp:
1961 * @enc: a #GstAudioEncoder
1963 * Queries encoder perfect timestamp behaviour.
1965 * Returns: TRUE if pefect timestamp setting enabled.
1972 gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc)
1976 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
1978 GST_OBJECT_LOCK (enc);
1979 result = enc->priv->perfect_ts;
1980 GST_OBJECT_UNLOCK (enc);
1986 * gst_audio_encoder_set_hard_sync:
1987 * @enc: a #GstAudioEncoder
1988 * @enabled: new state
1990 * Sets encoder hard resync handling.
1997 gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc, gboolean enabled)
1999 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
2001 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
2003 GST_OBJECT_LOCK (enc);
2004 enc->priv->hard_resync = enabled;
2005 GST_OBJECT_UNLOCK (enc);
2009 * gst_audio_encoder_get_hard_sync:
2010 * @enc: a #GstAudioEncoder
2012 * Queries encoder's hard resync setting.
2014 * Returns: TRUE if hard resync is enabled.
2021 gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc)
2025 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
2027 GST_OBJECT_LOCK (enc);
2028 result = enc->priv->hard_resync;
2029 GST_OBJECT_UNLOCK (enc);
2035 * gst_audio_encoder_set_tolerance:
2036 * @enc: a #GstAudioEncoder
2037 * @tolerance: new tolerance
2039 * Configures encoder audio jitter tolerance threshold.
2046 gst_audio_encoder_set_tolerance (GstAudioEncoder * enc, gint64 tolerance)
2048 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
2050 GST_OBJECT_LOCK (enc);
2051 enc->priv->tolerance = tolerance;
2052 GST_OBJECT_UNLOCK (enc);
2056 * gst_audio_encoder_get_tolerance:
2057 * @enc: a #GstAudioEncoder
2059 * Queries current audio jitter tolerance threshold.
2061 * Returns: encoder audio jitter tolerance threshold.
2068 gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)
2072 g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
2074 GST_OBJECT_LOCK (enc);
2075 result = enc->priv->tolerance;
2076 GST_OBJECT_UNLOCK (enc);
2082 * gst_audio_encoder_merge_tags:
2083 * @enc: a #GstAudioEncoder
2084 * @tags: a #GstTagList to merge
2085 * @mode: the #GstTagMergeMode to use
2087 * Adds tags to so-called pending tags, which will be processed
2088 * before pushing out data downstream.
2090 * Note that this is provided for convenience, and the subclass is
2091 * not required to use this and can still do tag handling on its own,
2092 * although it should be aware that baseclass already takes care
2093 * of the usual CODEC/AUDIO_CODEC tags.
2100 gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
2101 const GstTagList * tags, GstTagMergeMode mode)
2105 g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
2106 g_return_if_fail (tags == NULL || GST_IS_TAG_LIST (tags));
2108 GST_OBJECT_LOCK (enc);
2110 GST_DEBUG_OBJECT (enc, "merging tags %" GST_PTR_FORMAT, tags);
2111 otags = enc->priv->tags;
2112 enc->priv->tags = gst_tag_list_merge (enc->priv->tags, tags, mode);
2114 gst_tag_list_free (otags);
2115 GST_OBJECT_UNLOCK (enc);