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:gstbaseaudioencoder
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 * GstBaseAudioEncoder and subclass should cooperate as follows.
34 * <itemizedlist><title>Configuration</title>
36 * Initially, GstBaseAudioEncoder calls @start when the encoder element
37 * is activated, which allows subclass to perform any global setup.
40 * GstBaseAudioEncoder 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 * GstBaseAudioEncoder calls @stop at end of all processing.
52 * As of configuration stage, and throughout processing, GstBaseAudioEncoder
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_base_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 GstBaseAudioEncoderClass 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 * GstBaseAudioEncoder 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_base_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 #GstBaseAudioEncoder:perfect-ts.
105 * However, in the latter case, the input may not be so perfect or ideal, which
106 * is handled as follows. An input timestamp is compared with the expected
107 * timestamp as dictated by input sample stream and if the deviation is less
108 * than #GstBaseAudioEncoder:tolerance, the deviation is discarded.
109 * Otherwise, it is considered a discontuinity and subsequent output timestamp
110 * is resynced to the new position after performing configured discontinuity
111 * processing. In the non-perfect-ts case, an upstream variation exceeding
112 * tolerance only leads to marking DISCONT on subsequent outgoing
113 * (while timestamps are adjusted to upstream regardless of variation).
114 * While DISCONT is also marked in the perfect-ts case, this one optionally
115 * (see #GstBaseAudioEncoder: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_base_audio_encoder_finish_frame.
154 #define GST_USE_UNSTABLE_API
155 #include "gstbaseaudioencoder.h"
156 #include <gst/base/gstadapter.h>
157 #include <gst/audio/audio.h>
163 GST_DEBUG_CATEGORY_STATIC (gst_base_audio_encoder_debug);
164 #define GST_CAT_DEFAULT gst_base_audio_encoder_debug
166 #define GST_BASE_AUDIO_ENCODER_GET_PRIVATE(obj) \
167 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_ENCODER, \
168 GstBaseAudioEncoderPrivate))
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 _GstBaseAudioEncoderContext
193 /* MT-protected (with LOCK) */
194 GstClockTime min_latency;
195 GstClockTime max_latency;
196 } GstBaseAudioEncoderContext;
198 struct _GstBaseAudioEncoderPrivate
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 GstBaseAudioEncoderContext ctx;
240 gboolean hard_resync;
245 static GstElementClass *parent_class = NULL;
247 static void gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass *
249 static void gst_base_audio_encoder_init (GstBaseAudioEncoder * parse,
250 GstBaseAudioEncoderClass * klass);
253 gst_base_audio_encoder_get_type (void)
255 static GType base_audio_encoder_type = 0;
257 if (!base_audio_encoder_type) {
258 static const GTypeInfo base_audio_encoder_info = {
259 sizeof (GstBaseAudioEncoderClass),
260 (GBaseInitFunc) NULL,
261 (GBaseFinalizeFunc) NULL,
262 (GClassInitFunc) gst_base_audio_encoder_class_init,
265 sizeof (GstBaseAudioEncoder),
267 (GInstanceInitFunc) gst_base_audio_encoder_init,
269 const GInterfaceInfo preset_interface_info = {
270 NULL, /* interface_init */
271 NULL, /* interface_finalize */
272 NULL /* interface_data */
275 base_audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
276 "GstBaseAudioEncoder", &base_audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
278 g_type_add_interface_static (base_audio_encoder_type, GST_TYPE_PRESET,
279 &preset_interface_info);
281 return base_audio_encoder_type;
284 static void gst_base_audio_encoder_finalize (GObject * object);
285 static void gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc,
288 static void gst_base_audio_encoder_set_property (GObject * object,
289 guint prop_id, const GValue * value, GParamSpec * pspec);
290 static void gst_base_audio_encoder_get_property (GObject * object,
291 guint prop_id, GValue * value, GParamSpec * pspec);
293 static gboolean gst_base_audio_encoder_sink_activate_push (GstPad * pad,
296 static gboolean gst_base_audio_encoder_sink_event (GstPad * pad,
298 static gboolean gst_base_audio_encoder_sink_setcaps (GstPad * pad,
300 static GstFlowReturn gst_base_audio_encoder_chain (GstPad * pad,
302 static gboolean gst_base_audio_encoder_src_query (GstPad * pad,
304 static gboolean gst_base_audio_encoder_sink_query (GstPad * pad,
306 static const GstQueryType *gst_base_audio_encoder_get_query_types (GstPad *
308 static GstCaps *gst_base_audio_encoder_sink_getcaps (GstPad * pad);
312 gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass * klass)
314 GObjectClass *gobject_class;
316 gobject_class = G_OBJECT_CLASS (klass);
317 parent_class = g_type_class_peek_parent (klass);
319 GST_DEBUG_CATEGORY_INIT (gst_base_audio_encoder_debug, "baseaudioencoder", 0,
320 "baseaudioencoder element");
322 g_type_class_add_private (klass, sizeof (GstBaseAudioEncoderPrivate));
324 gobject_class->set_property = gst_base_audio_encoder_set_property;
325 gobject_class->get_property = gst_base_audio_encoder_get_property;
327 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_audio_encoder_finalize);
330 g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
331 g_param_spec_boolean ("perfect-timestamp", "Perfect Timestamps",
332 "Favour perfect timestamps over tracking upstream timestamps",
333 DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
334 g_object_class_install_property (gobject_class, PROP_GRANULE,
335 g_param_spec_boolean ("mark-granule", "Granule Marking",
336 "Apply granule semantics to buffer metadata (implies perfect-ts)",
337 DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
338 g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
339 g_param_spec_boolean ("hard-resync", "Hard Resync",
340 "Perform clipping and sample flushing upon discontinuity",
341 DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
342 g_object_class_install_property (gobject_class, PROP_TOLERANCE,
343 g_param_spec_int64 ("tolerance", "Tolerance",
344 "Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
345 0, G_MAXINT64, DEFAULT_TOLERANCE,
346 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
350 gst_base_audio_encoder_init (GstBaseAudioEncoder * enc,
351 GstBaseAudioEncoderClass * bclass)
353 GstPadTemplate *pad_template;
355 GST_DEBUG_OBJECT (enc, "gst_base_audio_encoder_init");
357 enc->priv = GST_BASE_AUDIO_ENCODER_GET_PRIVATE (enc);
359 /* only push mode supported */
361 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
362 g_return_if_fail (pad_template != NULL);
363 enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
364 gst_pad_set_event_function (enc->sinkpad,
365 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_event));
366 gst_pad_set_setcaps_function (enc->sinkpad,
367 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_setcaps));
368 gst_pad_set_getcaps_function (enc->sinkpad,
369 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_getcaps));
370 gst_pad_set_query_function (enc->sinkpad,
371 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_query));
372 gst_pad_set_chain_function (enc->sinkpad,
373 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_chain));
374 gst_pad_set_activatepush_function (enc->sinkpad,
375 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_activate_push));
376 gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
378 GST_DEBUG_OBJECT (enc, "sinkpad created");
380 /* and we don't mind upstream traveling stuff that much ... */
382 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
383 g_return_if_fail (pad_template != NULL);
384 enc->srcpad = gst_pad_new_from_template (pad_template, "src");
385 gst_pad_set_query_function (enc->srcpad,
386 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_src_query));
387 gst_pad_set_query_type_function (enc->srcpad,
388 GST_DEBUG_FUNCPTR (gst_base_audio_encoder_get_query_types));
389 gst_pad_use_fixed_caps (enc->srcpad);
390 gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
391 GST_DEBUG_OBJECT (enc, "src created");
393 enc->priv->adapter = gst_adapter_new ();
395 /* property default */
396 enc->priv->granule = DEFAULT_GRANULE;
397 enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
398 enc->priv->hard_resync = DEFAULT_HARD_RESYNC;
399 enc->priv->tolerance = DEFAULT_TOLERANCE;
402 gst_base_audio_encoder_reset (enc, TRUE);
403 GST_DEBUG_OBJECT (enc, "init ok");
407 gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
409 GST_OBJECT_LOCK (enc);
412 enc->priv->active = FALSE;
413 enc->priv->samples_in = 0;
414 enc->priv->bytes_out = 0;
415 gst_audio_info_clear (&enc->priv->ctx.info);
416 memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
419 gst_segment_init (&enc->segment, GST_FORMAT_TIME);
421 gst_adapter_clear (enc->priv->adapter);
422 enc->priv->got_data = FALSE;
423 enc->priv->drained = TRUE;
424 enc->priv->offset = 0;
425 enc->priv->base_ts = GST_CLOCK_TIME_NONE;
426 enc->priv->base_gp = -1;
427 enc->priv->samples = 0;
428 enc->priv->discont = FALSE;
430 GST_OBJECT_UNLOCK (enc);
434 gst_base_audio_encoder_finalize (GObject * object)
436 GstBaseAudioEncoder *enc = GST_BASE_AUDIO_ENCODER (object);
438 g_object_unref (enc->priv->adapter);
440 G_OBJECT_CLASS (parent_class)->finalize (object);
444 * gst_base_audio_encoder_finish_frame:
445 * @enc: a #GstBaseAudioEncoder
446 * @buffer: encoded data
447 * @samples: number of samples (per channel) represented by encoded data
449 * Collects encoded data and/or pushes encoded data downstream.
450 * Source pad caps must be set when this is called. Depending on the nature
451 * of the (framing of) the format, subclass can decide whether to push
452 * encoded data directly or to collect various "frames" in a single buffer.
453 * Note that the latter behaviour is recommended whenever the format is allowed,
454 * as it incurs no additional latency and avoids otherwise generating a
455 * a multitude of (small) output buffers. If not explicitly pushed,
456 * any available encoded data is pushed at the end of each processing cycle,
457 * i.e. which encodes as much data as available input data allows.
459 * If @samples < 0, then best estimate is all samples provided to encoder
460 * (subclass) so far. @buf may be NULL, in which case next number of @samples
461 * are considered discarded, e.g. as a result of discontinuous transmission,
462 * and a discontinuity is marked (note that @buf == NULL => push == TRUE).
464 * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
469 gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
472 GstBaseAudioEncoderClass *klass;
473 GstBaseAudioEncoderPrivate *priv;
474 GstBaseAudioEncoderContext *ctx;
475 GstFlowReturn ret = GST_FLOW_OK;
477 klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
479 ctx = &enc->priv->ctx;
481 /* subclass should know what it is producing by now */
482 g_return_val_if_fail (GST_PAD_CAPS (enc->srcpad) != NULL, GST_FLOW_ERROR);
483 /* subclass should not hand us no data */
484 g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
487 GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
488 buf ? GST_BUFFER_SIZE (buf) : -1, samples);
490 /* mark subclass still alive and providing */
491 priv->got_data = TRUE;
493 /* remove corresponding samples from input */
495 samples = (enc->priv->offset / ctx->info.bpf);
497 if (G_LIKELY (samples)) {
498 /* track upstream ts if so configured */
499 if (!enc->priv->perfect_ts) {
500 guint64 ts, distance;
502 ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
503 g_assert (distance % ctx->info.bpf == 0);
504 distance /= ctx->info.bpf;
505 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
506 GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
507 GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
508 GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
509 /* when draining adapter might be empty and no ts to offer */
510 if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
511 GstClockTimeDiff diff;
512 GstClockTime old_ts, next_ts;
514 /* passed into another buffer;
515 * mild check for discontinuity and only mark if so */
517 gst_util_uint64_scale (distance, GST_SECOND, ctx->info.rate);
518 old_ts = priv->base_ts +
519 gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->info.rate);
520 diff = GST_CLOCK_DIFF (next_ts, old_ts);
521 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
522 /* only mark discontinuity if beyond tolerance */
523 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
524 diff > enc->priv->tolerance)) {
525 GST_DEBUG_OBJECT (enc, "marked discont");
526 priv->discont = TRUE;
528 if (diff > GST_SECOND / ctx->info.rate / 2 ||
529 diff < -GST_SECOND / ctx->info.rate / 2) {
530 GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
531 " at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
532 /* re-sync to upstream ts */
534 priv->samples = distance;
536 GST_LOG_OBJECT (enc, "new upstream ts only introduces jitter");
540 /* advance sample view */
541 if (G_UNLIKELY (samples * ctx->info.bpf > priv->offset)) {
542 if (G_LIKELY (!priv->force)) {
543 /* no way we can let this pass */
544 g_assert_not_reached ();
549 if (samples * ctx->info.bpf >= gst_adapter_available (priv->adapter))
550 gst_adapter_clear (priv->adapter);
552 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
555 gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
556 priv->offset -= samples * ctx->info.bpf;
557 /* avoid subsequent stray prev_ts */
558 if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
559 gst_adapter_clear (priv->adapter);
561 /* sample count advanced below after buffer handling */
565 if (G_LIKELY (buf)) {
566 GST_LOG_OBJECT (enc, "taking %d bytes for output", GST_BUFFER_SIZE (buf));
567 buf = gst_buffer_make_metadata_writable (buf);
570 gst_buffer_set_caps (buf, GST_PAD_CAPS (enc->srcpad));
571 if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
572 /* FIXME ? lookahead could lead to weird ts and duration ?
573 * (particularly if not in perfect mode) */
574 /* mind sample rounding and produce perfect output */
575 GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
576 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
578 GST_DEBUG_OBJECT (enc, "out samples %d", samples);
579 if (G_LIKELY (samples > 0)) {
580 priv->samples += samples;
581 GST_BUFFER_DURATION (buf) = priv->base_ts +
582 gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
583 ctx->info.rate) - GST_BUFFER_TIMESTAMP (buf);
584 priv->last_duration = GST_BUFFER_DURATION (buf);
586 /* duration forecast in case of handling remainder;
587 * the last one is probably like the previous one ... */
588 GST_BUFFER_DURATION (buf) = priv->last_duration;
590 if (priv->base_gp >= 0) {
592 /* FIXME: in longer run, muxer should take care of this ... */
593 /* offset_end = granulepos for ogg muxer */
594 GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples -
595 enc->priv->ctx.lookahead;
596 /* offset = timestamp corresponding to granulepos for ogg muxer */
597 GST_BUFFER_OFFSET (buf) =
598 GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf),
601 GST_BUFFER_OFFSET (buf) = priv->bytes_out;
602 GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf);
606 priv->bytes_out += GST_BUFFER_SIZE (buf);
608 if (G_UNLIKELY (priv->discont)) {
609 GST_LOG_OBJECT (enc, "marking discont");
610 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
611 priv->discont = FALSE;
614 if (klass->pre_push) {
615 /* last chance for subclass to do some dirty stuff */
616 ret = klass->pre_push (enc, &buf);
617 if (ret != GST_FLOW_OK || !buf) {
618 GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
619 gst_flow_get_name (ret), buf);
621 gst_buffer_unref (buf);
626 GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
627 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
628 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
629 GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
631 ret = gst_pad_push (enc->srcpad, buf);
632 GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
634 /* merely advance samples, most work for that already done above */
635 priv->samples += samples;
644 GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
645 ("received more encoded samples %d than provided %d",
646 samples, priv->offset / ctx->info.bpf), (NULL));
648 gst_buffer_unref (buf);
649 return GST_FLOW_ERROR;
653 /* adapter tracking idea:
654 * - start of adapter corresponds with what has already been encoded
655 * (i.e. really returned by encoder subclass)
656 * - start + offset is what needs to be fed to subclass next */
658 gst_base_audio_encoder_push_buffers (GstBaseAudioEncoder * enc, gboolean force)
660 GstBaseAudioEncoderClass *klass;
661 GstBaseAudioEncoderPrivate *priv;
662 GstBaseAudioEncoderContext *ctx;
665 GstFlowReturn ret = GST_FLOW_OK;
667 klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
669 g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
672 ctx = &enc->priv->ctx;
674 while (ret == GST_FLOW_OK) {
677 av = gst_adapter_available (priv->adapter);
679 g_assert (priv->offset <= av);
682 need = ctx->frame_samples > 0 ? ctx->frame_samples * ctx->info.bpf : av;
683 GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d",
686 if ((need > av) || !av) {
687 if (G_UNLIKELY (force)) {
697 /* if we have some extra metadata,
698 * provide for integer multiple of frames to allow for better granularity
700 if (ctx->frame_samples > 0 && need) {
701 if (ctx->frame_max > 1)
702 need = need * MIN ((av / need), ctx->frame_max);
703 else if (ctx->frame_max == 0)
704 need = need * (av / need);
708 buf = gst_buffer_new ();
709 GST_BUFFER_DATA (buf) = (guint8 *)
710 gst_adapter_peek (priv->adapter, priv->offset + need) + priv->offset;
711 GST_BUFFER_SIZE (buf) = need;
714 GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
717 /* mark this already as consumed,
718 * which it should be when subclass gives us data in exchange for samples */
719 priv->offset += need;
720 priv->samples_in += need / ctx->info.bpf;
722 priv->got_data = FALSE;
723 ret = klass->handle_frame (enc, buf);
726 gst_buffer_unref (buf);
728 /* no data to feed, no leftover provided, then bail out */
729 if (G_UNLIKELY (!buf && !priv->got_data)) {
730 priv->drained = TRUE;
731 GST_LOG_OBJECT (enc, "no more data drained from subclass");
740 gst_base_audio_encoder_drain (GstBaseAudioEncoder * enc)
742 if (enc->priv->drained)
745 return gst_base_audio_encoder_push_buffers (enc, TRUE);
749 gst_base_audio_encoder_set_base_gp (GstBaseAudioEncoder * enc)
753 if (!enc->priv->granule)
756 /* use running time for granule */
757 /* incoming data is clipped, so a valid input should yield a valid output */
758 ts = gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME,
760 if (GST_CLOCK_TIME_IS_VALID (ts)) {
762 GST_CLOCK_TIME_TO_FRAMES (enc->priv->base_ts, enc->priv->ctx.info.rate);
763 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, enc->priv->base_gp);
765 /* should reasonably have a valid base,
766 * otherwise start at 0 if we did not already start there earlier */
767 if (enc->priv->base_gp < 0) {
768 enc->priv->base_gp = 0;
769 GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
776 gst_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
778 GstBaseAudioEncoder *enc;
779 GstBaseAudioEncoderPrivate *priv;
780 GstBaseAudioEncoderContext *ctx;
781 GstFlowReturn ret = GST_FLOW_OK;
784 enc = GST_BASE_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
787 ctx = &enc->priv->ctx;
789 /* should know what is coming by now */
794 "received buffer of size %d with ts %" GST_TIME_FORMAT
795 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
796 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
797 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
799 /* input shoud be whole number of sample frames */
800 if (GST_BUFFER_SIZE (buffer) % ctx->info.bpf)
803 #ifndef GST_DISABLE_GST_DEBUG
805 GstClockTime duration;
806 GstClockTimeDiff diff;
808 /* verify buffer duration */
809 duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND,
810 ctx->info.rate * ctx->info.bpf);
811 diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
812 if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
813 (diff > GST_SECOND / ctx->info.rate / 2 ||
814 diff < -GST_SECOND / ctx->info.rate / 2)) {
815 GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
816 GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
817 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
818 GST_TIME_ARGS (duration));
823 discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
824 if (G_UNLIKELY (discont)) {
825 GST_LOG_OBJECT (buffer, "marked discont");
826 enc->priv->discont = discont;
829 /* clip to segment */
830 /* NOTE: slightly painful linking -laudio only for this one ... */
831 buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->info.rate,
833 if (G_UNLIKELY (!buffer)) {
834 GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
839 "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
840 ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
841 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
842 GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
844 if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
845 priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
846 GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
847 GST_TIME_ARGS (priv->base_ts));
848 gst_base_audio_encoder_set_base_gp (enc);
851 /* check for continuity;
852 * checked elsewhere in non-perfect case */
853 if (enc->priv->perfect_ts) {
854 GstClockTimeDiff diff = 0;
855 GstClockTime next_ts = 0;
857 if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
858 GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
861 samples = priv->samples +
862 gst_adapter_available (priv->adapter) / ctx->info.bpf;
863 next_ts = priv->base_ts +
864 gst_util_uint64_scale (samples, GST_SECOND, ctx->info.rate);
865 GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
866 " samples past base_ts %" GST_TIME_FORMAT
867 ", expected ts %" GST_TIME_FORMAT, samples,
868 GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
869 diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
870 GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
871 /* if within tolerance,
872 * discard buffer ts and carry on producing perfect stream,
873 * otherwise clip or resync to ts */
874 if (G_UNLIKELY (diff < -enc->priv->tolerance ||
875 diff > enc->priv->tolerance)) {
876 GST_DEBUG_OBJECT (enc, "marked discont");
881 /* do some fancy tweaking in hard resync case */
882 if (discont && enc->priv->hard_resync) {
886 GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
887 GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
890 GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->info.rate) * ctx->info.bpf;
891 if (diff_bytes >= GST_BUFFER_SIZE (buffer)) {
892 gst_buffer_unref (buffer);
895 buffer = gst_buffer_make_metadata_writable (buffer);
896 GST_BUFFER_DATA (buffer) += diff_bytes;
897 GST_BUFFER_SIZE (buffer) -= diff_bytes;
899 GST_BUFFER_TIMESTAMP (buffer) += diff;
900 /* care even less about duration after this */
902 /* drain stuff prior to resync */
903 gst_base_audio_encoder_drain (enc);
907 priv->base_ts += diff;
908 gst_base_audio_encoder_set_base_gp (enc);
909 priv->discont |= discont;
912 gst_adapter_push (enc->priv->adapter, buffer);
913 /* new stuff, so we can push subclass again */
914 enc->priv->drained = FALSE;
916 ret = gst_base_audio_encoder_push_buffers (enc, FALSE);
919 GST_LOG_OBJECT (enc, "chain leaving");
925 GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
926 ("encoder not initialized"));
927 gst_buffer_unref (buffer);
928 return GST_FLOW_NOT_NEGOTIATED;
932 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
933 ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
935 gst_buffer_unref (buffer);
936 return GST_FLOW_ERROR;
941 audio_info_is_equal (GstAudioInfo * from, GstAudioInfo * to)
945 if (GST_AUDIO_INFO_FORMAT (from) != GST_AUDIO_INFO_FORMAT (to))
947 if (GST_AUDIO_INFO_RATE (from) != GST_AUDIO_INFO_RATE (to))
949 if (GST_AUDIO_INFO_CHANNELS (from) != GST_AUDIO_INFO_CHANNELS (to))
951 if (GST_AUDIO_INFO_CHANNELS (from) > 64)
953 return memcmp (from->position, to->position,
954 GST_AUDIO_INFO_CHANNELS (from) * sizeof (to->position[0]));
958 gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
960 GstBaseAudioEncoder *enc;
961 GstBaseAudioEncoderClass *klass;
962 GstBaseAudioEncoderContext *ctx;
963 GstAudioInfo *state, *old_state;
964 gboolean res = TRUE, changed = FALSE;
967 enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
968 klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
970 /* subclass must do something here ... */
971 g_return_val_if_fail (klass->set_format != NULL, FALSE);
973 ctx = &enc->priv->ctx;
976 GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
978 if (!gst_caps_is_fixed (caps))
981 /* adjust ts tracking to new sample rate */
982 old_rate = GST_AUDIO_INFO_RATE (state);
983 if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
984 enc->priv->base_ts +=
985 GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
986 enc->priv->samples = 0;
989 old_state = gst_audio_info_copy (state);
990 if (!gst_audio_info_from_caps (state, caps))
993 changed = audio_info_is_equal (state, old_state);
994 gst_audio_info_free (old_state);
997 GstClockTime old_min_latency;
998 GstClockTime old_max_latency;
1000 /* drain any pending old data stuff */
1001 gst_base_audio_encoder_drain (enc);
1003 /* context defaults */
1004 enc->priv->ctx.frame_samples = 0;
1005 enc->priv->ctx.frame_max = 0;
1006 enc->priv->ctx.lookahead = 0;
1008 /* element might report latency */
1009 GST_OBJECT_LOCK (enc);
1010 old_min_latency = ctx->min_latency;
1011 old_max_latency = ctx->max_latency;
1012 GST_OBJECT_UNLOCK (enc);
1014 if (klass->set_format)
1015 res = klass->set_format (enc, state);
1017 /* notify if new latency */
1018 GST_OBJECT_LOCK (enc);
1019 if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
1020 (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
1021 GST_OBJECT_UNLOCK (enc);
1022 /* post latency message on the bus */
1023 gst_element_post_message (GST_ELEMENT (enc),
1024 gst_message_new_latency (GST_OBJECT (enc)));
1025 GST_OBJECT_LOCK (enc);
1027 GST_OBJECT_UNLOCK (enc);
1029 GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
1037 GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
1044 * gst_base_audio_encoder_proxy_getcaps:
1045 * @enc: a #GstBaseAudioEncoder
1046 * @caps: initial caps
1048 * Returns caps that express @caps (or sink template caps if @caps == NULL)
1049 * restricted to channel/rate combinations supported by downstream elements
1052 * Returns: a #GstCaps owned by caller
1057 gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc, GstCaps * caps)
1059 const GstCaps *templ_caps;
1060 GstCaps *allowed = NULL;
1061 GstCaps *fcaps, *filter_caps;
1064 /* we want to be able to communicate to upstream elements like audioconvert
1065 * and audioresample any rate/channel restrictions downstream (e.g. muxer
1066 * only accepting certain sample rates) */
1067 templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad);
1068 allowed = gst_pad_get_allowed_caps (enc->srcpad);
1069 if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
1070 fcaps = gst_caps_copy (templ_caps);
1074 GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
1075 GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
1077 filter_caps = gst_caps_new_empty ();
1079 for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
1082 q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
1084 /* pick rate + channel fields from allowed caps */
1085 for (j = 0; j < gst_caps_get_size (allowed); j++) {
1086 const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
1090 s = gst_structure_id_empty_new (q_name);
1091 if ((val = gst_structure_get_value (allowed_s, "rate")))
1092 gst_structure_set_value (s, "rate", val);
1093 if ((val = gst_structure_get_value (allowed_s, "channels")))
1094 gst_structure_set_value (s, "channels", val);
1096 gst_caps_merge_structure (filter_caps, s);
1100 fcaps = gst_caps_intersect (filter_caps, templ_caps);
1101 gst_caps_unref (filter_caps);
1104 gst_caps_replace (&allowed, NULL);
1106 GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
1112 gst_base_audio_encoder_sink_getcaps (GstPad * pad)
1114 GstBaseAudioEncoder *enc;
1115 GstBaseAudioEncoderClass *klass;
1118 enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
1119 klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
1120 g_assert (pad == enc->sinkpad);
1123 caps = klass->getcaps (enc);
1125 caps = gst_base_audio_encoder_proxy_getcaps (enc, NULL);
1126 gst_object_unref (enc);
1128 GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
1134 gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
1137 GstBaseAudioEncoderClass *klass;
1138 gboolean handled = FALSE;
1140 klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
1142 switch (GST_EVENT_TYPE (event)) {
1143 case GST_EVENT_NEWSEGMENT:
1146 gdouble rate, arate;
1147 gint64 start, stop, time;
1150 gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
1151 &start, &stop, &time);
1153 if (format == GST_FORMAT_TIME) {
1154 GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
1155 " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
1156 ", rate %g, applied_rate %g",
1157 GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
1160 GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
1161 " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
1162 ", rate %g, applied_rate %g", start, stop, time, rate, arate);
1163 GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
1167 /* finish current segment */
1168 gst_base_audio_encoder_drain (enc);
1169 /* reset partially for new segment */
1170 gst_base_audio_encoder_reset (enc, FALSE);
1171 /* and follow along with segment */
1172 gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
1173 format, start, stop, time);
1177 case GST_EVENT_FLUSH_START:
1180 case GST_EVENT_FLUSH_STOP:
1181 /* discard any pending stuff */
1182 /* TODO route through drain ?? */
1183 if (!enc->priv->drained && klass->flush)
1185 /* and get (re)set for the sequel */
1186 gst_base_audio_encoder_reset (enc, FALSE);
1190 gst_base_audio_encoder_drain (enc);
1201 gst_base_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
1203 GstBaseAudioEncoder *enc;
1204 GstBaseAudioEncoderClass *klass;
1205 gboolean handled = FALSE;
1206 gboolean ret = TRUE;
1208 enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
1209 klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
1211 GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
1212 GST_EVENT_TYPE_NAME (event));
1215 handled = klass->event (enc, event);
1218 handled = gst_base_audio_encoder_sink_eventfunc (enc, event);
1221 ret = gst_pad_event_default (pad, event);
1223 GST_DEBUG_OBJECT (enc, "event handled");
1225 gst_object_unref (enc);
1230 gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
1232 gboolean res = TRUE;
1233 GstBaseAudioEncoder *enc;
1235 enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
1237 switch (GST_QUERY_TYPE (query)) {
1238 case GST_QUERY_FORMATS:
1240 gst_query_set_formats (query, 3,
1241 GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
1245 case GST_QUERY_CONVERT:
1247 GstFormat src_fmt, dest_fmt;
1248 gint64 src_val, dest_val;
1250 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1251 if (!(res = gst_audio_info_convert (&enc->priv->ctx.info,
1252 src_fmt, src_val, dest_fmt, &dest_val)))
1254 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1258 res = gst_pad_query_default (pad, query);
1263 gst_object_unref (enc);
1267 static const GstQueryType *
1268 gst_base_audio_encoder_get_query_types (GstPad * pad)
1270 static const GstQueryType gst_base_audio_encoder_src_query_types[] = {
1278 return gst_base_audio_encoder_src_query_types;
1282 * gst_base_audio_encoded_audio_convert:
1283 * @fmt: audio format of the encoded audio
1284 * @bytes: number of encoded bytes
1285 * @samples: number of encoded samples
1286 * @src_format: source format
1287 * @src_value: source value
1288 * @dest_format: destination format
1289 * @dest_value: destination format
1291 * Helper function to convert @src_value in @src_format to @dest_value in
1292 * @dest_format for encoded audio data. Conversion is possible between
1293 * BYTE and TIME format by using estimated bitrate based on
1294 * @samples and @bytes (and @fmt).
1298 /* FIXME: make gst_base_audio_encoded_audio_convert() public? */
1300 gst_base_audio_encoded_audio_convert (GstAudioInfo * fmt,
1301 gint64 bytes, gint64 samples, GstFormat src_format,
1302 gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
1304 gboolean res = FALSE;
1306 g_return_val_if_fail (dest_format != NULL, FALSE);
1307 g_return_val_if_fail (dest_value != NULL, FALSE);
1309 if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
1312 *dest_value = src_value;
1316 if (samples == 0 || bytes == 0 || fmt->rate == 0) {
1317 GST_DEBUG ("not enough metadata yet to convert");
1323 switch (src_format) {
1324 case GST_FORMAT_BYTES:
1325 switch (*dest_format) {
1326 case GST_FORMAT_TIME:
1327 *dest_value = gst_util_uint64_scale (src_value,
1328 GST_SECOND * samples, bytes);
1335 case GST_FORMAT_TIME:
1336 switch (*dest_format) {
1337 case GST_FORMAT_BYTES:
1338 *dest_value = gst_util_uint64_scale (src_value, bytes,
1339 samples * GST_SECOND);
1354 /* FIXME ? are any of these queries (other than latency) an encoder's business
1355 * also, the conversion stuff might seem to make sense, but seems to not mind
1356 * segment stuff etc at all
1357 * Supposedly that's backward compatibility ... */
1359 gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
1361 GstBaseAudioEncoder *enc;
1363 gboolean res = FALSE;
1365 enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
1366 peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
1368 GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
1370 switch (GST_QUERY_TYPE (query)) {
1371 case GST_QUERY_POSITION:
1373 GstFormat fmt, req_fmt;
1376 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1377 GST_LOG_OBJECT (enc, "returning peer response");
1382 GST_LOG_OBJECT (enc, "no peer");
1386 gst_query_parse_position (query, &req_fmt, NULL);
1387 fmt = GST_FORMAT_TIME;
1388 if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
1391 if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) {
1392 gst_query_set_position (query, req_fmt, val);
1396 case GST_QUERY_DURATION:
1398 GstFormat fmt, req_fmt;
1401 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1402 GST_LOG_OBJECT (enc, "returning peer response");
1407 GST_LOG_OBJECT (enc, "no peer");
1411 gst_query_parse_duration (query, &req_fmt, NULL);
1412 fmt = GST_FORMAT_TIME;
1413 if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
1416 if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) {
1417 gst_query_set_duration (query, req_fmt, val);
1421 case GST_QUERY_FORMATS:
1423 gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
1427 case GST_QUERY_CONVERT:
1429 GstFormat src_fmt, dest_fmt;
1430 gint64 src_val, dest_val;
1432 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
1433 if (!(res = gst_base_audio_encoded_audio_convert (&enc->priv->ctx.info,
1434 enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
1435 &dest_fmt, &dest_val)))
1437 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
1440 case GST_QUERY_LATENCY:
1442 if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
1444 GstClockTime min_latency, max_latency;
1446 gst_query_parse_latency (query, &live, &min_latency, &max_latency);
1447 GST_DEBUG_OBJECT (enc, "Peer latency: live %d, min %"
1448 GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
1449 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
1451 GST_OBJECT_LOCK (enc);
1452 /* add our latency */
1453 if (min_latency != -1)
1454 min_latency += enc->priv->ctx.min_latency;
1455 if (max_latency != -1)
1456 max_latency += enc->priv->ctx.max_latency;
1457 GST_OBJECT_UNLOCK (enc);
1459 gst_query_set_latency (query, live, min_latency, max_latency);
1464 res = gst_pad_query_default (pad, query);
1468 gst_object_unref (peerpad);
1473 gst_base_audio_encoder_set_property (GObject * object, guint prop_id,
1474 const GValue * value, GParamSpec * pspec)
1476 GstBaseAudioEncoder *enc;
1478 enc = GST_BASE_AUDIO_ENCODER (object);
1481 case PROP_PERFECT_TS:
1482 if (enc->priv->granule && !g_value_get_boolean (value))
1483 GST_WARNING_OBJECT (enc, "perfect-ts can not be set FALSE");
1485 enc->priv->perfect_ts = g_value_get_boolean (value);
1487 case PROP_HARD_RESYNC:
1488 enc->priv->hard_resync = g_value_get_boolean (value);
1490 case PROP_TOLERANCE:
1491 enc->priv->tolerance = g_value_get_int64 (value);
1494 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1500 gst_base_audio_encoder_get_property (GObject * object, guint prop_id,
1501 GValue * value, GParamSpec * pspec)
1503 GstBaseAudioEncoder *enc;
1505 enc = GST_BASE_AUDIO_ENCODER (object);
1508 case PROP_PERFECT_TS:
1509 g_value_set_boolean (value, enc->priv->perfect_ts);
1512 g_value_set_boolean (value, enc->priv->granule);
1514 case PROP_HARD_RESYNC:
1515 g_value_set_boolean (value, enc->priv->hard_resync);
1517 case PROP_TOLERANCE:
1518 g_value_set_int64 (value, enc->priv->tolerance);
1521 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1527 gst_base_audio_encoder_activate (GstBaseAudioEncoder * enc, gboolean active)
1529 GstBaseAudioEncoderClass *klass;
1530 gboolean result = FALSE;
1532 klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
1534 g_return_val_if_fail (!enc->priv->granule || enc->priv->perfect_ts, FALSE);
1536 GST_DEBUG_OBJECT (enc, "activate %d", active);
1539 if (!enc->priv->active && klass->start)
1540 result = klass->start (enc);
1542 /* We must make sure streaming has finished before resetting things
1543 * and calling the ::stop vfunc */
1544 GST_PAD_STREAM_LOCK (enc->sinkpad);
1545 GST_PAD_STREAM_UNLOCK (enc->sinkpad);
1547 if (enc->priv->active && klass->stop)
1548 result = klass->stop (enc);
1551 gst_base_audio_encoder_reset (enc, TRUE);
1553 GST_DEBUG_OBJECT (enc, "activate return: %d", result);
1559 gst_base_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
1561 gboolean result = TRUE;
1562 GstBaseAudioEncoder *enc;
1564 enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
1566 GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
1568 result = gst_base_audio_encoder_activate (enc, active);
1571 enc->priv->active = active;
1573 GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
1575 gst_object_unref (enc);
1580 * gst_base_audio_encoder_get_audio_info:
1581 * @enc: a #GstBaseAudioEncoder
1583 * Returns: a #GstAudioInfo describing the input audio format
1588 gst_base_audio_encoder_get_audio_info (GstBaseAudioEncoder * enc)
1590 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), NULL);
1592 return &enc->priv->ctx.info;
1596 * gst_base_audio_encoder_set_frame_samples:
1597 * @enc: a #GstBaseAudioEncoder
1598 * @num: number of samples per frame
1600 * Sets number of samples (per channel) subclass needs to be handed,
1601 * or will be handed all available if 0.
1606 gst_base_audio_encoder_set_frame_samples (GstBaseAudioEncoder * enc, gint num)
1608 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1610 enc->priv->ctx.frame_samples = num;
1614 * gst_base_audio_encoder_get_frame_samples:
1615 * @enc: a #GstBaseAudioEncoder
1617 * Returns: currently requested samples per frame
1622 gst_base_audio_encoder_get_frame_samples (GstBaseAudioEncoder * enc)
1624 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), 0);
1626 return enc->priv->ctx.frame_samples;
1630 * gst_base_audio_encoder_set_frame_max:
1631 * @enc: a #GstBaseAudioEncoder
1632 * @num: number of frames
1634 * Sets max number of frames accepted at once (assumed minimally 1)
1639 gst_base_audio_encoder_set_frame_max (GstBaseAudioEncoder * enc, gint num)
1641 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1643 enc->priv->ctx.frame_max = num;
1647 * gst_base_audio_encoder_get_frame_max:
1648 * @enc: a #GstBaseAudioEncoder
1650 * Returns: currently configured maximum handled frames
1655 gst_base_audio_encoder_get_frame_max (GstBaseAudioEncoder * enc)
1657 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), 0);
1659 return enc->priv->ctx.frame_max;
1663 * gst_base_audio_encoder_set_lookahead:
1664 * @enc: a #GstBaseAudioEncoder
1667 * Sets encoder lookahead (in units of input rate samples)
1672 gst_base_audio_encoder_set_lookahead (GstBaseAudioEncoder * enc, gint num)
1674 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1676 enc->priv->ctx.lookahead = num;
1680 * gst_base_audio_encoder_get_lookahead:
1681 * @enc: a #GstBaseAudioEncoder
1683 * Returns: currently configured encoder lookahead
1686 gst_base_audio_encoder_get_lookahead (GstBaseAudioEncoder * enc)
1688 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), 0);
1690 return enc->priv->ctx.lookahead;
1694 * gst_base_audio_encoder_set_latency:
1695 * @enc: a #GstBaseAudioEncoder
1696 * @min: minimum latency
1697 * @max: maximum latency
1699 * Sets encoder latency.
1704 gst_base_audio_encoder_set_latency (GstBaseAudioEncoder * enc,
1705 GstClockTime min, GstClockTime max)
1707 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1709 GST_OBJECT_LOCK (enc);
1710 enc->priv->ctx.min_latency = min;
1711 enc->priv->ctx.max_latency = max;
1712 GST_OBJECT_UNLOCK (enc);
1716 * gst_base_audio_encoder_get_latency:
1717 * @enc: a #GstBaseAudioEncoder
1718 * @min: a pointer to storage to hold minimum latency
1719 * @max: a pointer to storage to hold maximum latency
1721 * Returns currently configured latency.
1726 gst_base_audio_encoder_get_latency (GstBaseAudioEncoder * enc,
1727 GstClockTime * min, GstClockTime * max)
1729 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1731 GST_OBJECT_LOCK (enc);
1733 *min = enc->priv->ctx.min_latency;
1735 *max = enc->priv->ctx.max_latency;
1736 GST_OBJECT_UNLOCK (enc);
1740 * gst_base_audio_encoder_set_mark_granule:
1741 * @enc: a #GstBaseAudioEncoder
1742 * @enabled: new state
1744 * Enable or disable encoder granule handling.
1751 gst_base_audio_encoder_set_mark_granule (GstBaseAudioEncoder * enc,
1754 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1756 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1758 GST_OBJECT_LOCK (enc);
1759 enc->priv->granule = enabled;
1760 GST_OBJECT_UNLOCK (enc);
1764 * gst_base_audio_encoder_get_mark_granule:
1765 * @enc: a #GstBaseAudioEncoder
1767 * Queries if the encoder will handle granule marking.
1769 * Returns: TRUE if granule marking is enabled.
1776 gst_base_audio_encoder_get_mark_granule (GstBaseAudioEncoder * enc)
1780 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), FALSE);
1782 GST_OBJECT_LOCK (enc);
1783 result = enc->priv->granule;
1784 GST_OBJECT_UNLOCK (enc);
1790 * gst_base_audio_encoder_set_perfect_timestamp:
1791 * @enc: a #GstBaseAudioEncoder
1792 * @enabled: new state
1794 * Enable or disable encoder perfect output timestamp preference.
1801 gst_base_audio_encoder_set_perfect_timestamp (GstBaseAudioEncoder * enc,
1804 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1806 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1808 GST_OBJECT_LOCK (enc);
1809 enc->priv->perfect_ts = enabled;
1810 GST_OBJECT_UNLOCK (enc);
1814 * gst_base_audio_encoder_get_perfect_timestamp:
1815 * @enc: a #GstBaseAudioEncoder
1817 * Queries encoder perfect timestamp behaviour.
1819 * Returns: TRUE if pefect timestamp setting enabled.
1826 gst_base_audio_encoder_get_perfect_timestamp (GstBaseAudioEncoder * enc)
1830 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), FALSE);
1832 GST_OBJECT_LOCK (enc);
1833 result = enc->priv->perfect_ts;
1834 GST_OBJECT_UNLOCK (enc);
1840 * gst_base_audio_encoder_set_hard_sync:
1841 * @enc: a #GstBaseAudioEncoder
1842 * @enabled: new state
1844 * Sets encoder hard resync handling.
1851 gst_base_audio_encoder_set_hard_resync (GstBaseAudioEncoder * enc,
1854 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1856 GST_LOG_OBJECT (enc, "enabled: %d", enabled);
1858 GST_OBJECT_LOCK (enc);
1859 enc->priv->hard_resync = enabled;
1860 GST_OBJECT_UNLOCK (enc);
1864 * gst_base_audio_encoder_get_hard_sync:
1865 * @enc: a #GstBaseAudioEncoder
1867 * Queries encoder's hard resync setting.
1869 * Returns: TRUE if hard resync is enabled.
1876 gst_base_audio_encoder_get_hard_resync (GstBaseAudioEncoder * enc)
1880 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), FALSE);
1882 GST_OBJECT_LOCK (enc);
1883 result = enc->priv->hard_resync;
1884 GST_OBJECT_UNLOCK (enc);
1890 * gst_base_audio_encoder_set_tolerance:
1891 * @enc: a #GstBaseAudioEncoder
1892 * @tolerance: new tolerance
1894 * Configures encoder audio jitter tolerance threshold.
1901 gst_base_audio_encoder_set_tolerance (GstBaseAudioEncoder * enc,
1904 g_return_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc));
1906 GST_OBJECT_LOCK (enc);
1907 enc->priv->tolerance = tolerance;
1908 GST_OBJECT_UNLOCK (enc);
1912 * gst_base_audio_encoder_get_tolerance:
1913 * @enc: a #GstBaseAudioEncoder
1915 * Queries current audio jitter tolerance threshold.
1917 * Returns: encoder audio jitter tolerance threshold.
1924 gst_base_audio_encoder_get_tolerance (GstBaseAudioEncoder * enc)
1928 g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), 0);
1930 GST_OBJECT_LOCK (enc);
1931 result = enc->priv->tolerance;
1932 GST_OBJECT_UNLOCK (enc);