Taken from http://cgit.collabora.com/git/user/manauw/gst-plugins-bad.git/log/?h=baseaudio
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral Quiroga <itoral@igalia.com>
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:gstbaseaudiodecoder
+ * @short_description: Base class for audio decoders
+ * @see_also: #GstBaseTransform
+ *
+ * This base class is for audio decoders turning encoded data into
+ * raw audio samples.
+ *
+ * GstBaseAudioDecoder and subclass should cooperate as follows.
+ * <orderedlist>
+ * <listitem>
+ * <itemizedlist><title>Configuration</title>
+ * <listitem><para>
+ * Initially, GstBaseAudioDecoder calls @start when the decoder element
+ * is activated, which allows subclass to perform any global setup.
+ * Base class context parameters can already be set according to subclass
+ * capabilities (or possibly upon receive more information in subsequent
+ * @set_format).
+ * </para></listitem>
+ * <listitem><para>
+ * GstBaseAudioDecoder calls @set_format to inform subclass of the format
+ * of input audio data that it is about to receive.
+ * While unlikely, it might be called more than once, if changing input
+ * parameters require reconfiguration.
+ * </para></listitem>
+ * <listitem><para>
+ * GstBaseAudioDecoder calls @stop at end of all processing.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * As of configuration stage, and throughout processing, GstBaseAudioDecoder
+ * provides a GstBaseAudioDecoderContext that provides required context,
+ * e.g. describing the format of output audio data
+ * (valid when output caps have been caps) or current parsing state.
+ * Conversely, subclass can and should configure context to inform
+ * base class of its expectation w.r.t. buffer handling.
+ * <listitem>
+ * <itemizedlist>
+ * <title>Data processing</title>
+ * <listitem><para>
+ * Base class gathers input data, and optionally allows subclass
+ * to parse this into subsequently manageable (as defined by subclass)
+ * chunks. Such chunks are subsequently referred to as 'frames',
+ * though they may or may not correspond to 1 (or more) audio format frame.
+ * </para></listitem>
+ * <listitem><para>
+ * Input frame is provided to subclass' @handle_frame.
+ * </para></listitem>
+ * <listitem><para>
+ * If codec processing results in decoded data, subclass should call
+ * @gst_base_audio_decoder_finish_frame to have decoded data pushed
+ * downstream.
+ * </para></listitem>
+ * <listitem><para>
+ * Just prior to actually pushing a buffer downstream,
+ * it is passed to @pre_push. Subclass should either use this callback
+ * to arrange for additional downstream pushing or otherwise ensure such
+ * custom pushing occurs after at least a method call has finished since
+ * setting src pad caps.
+ * </para></listitem>
+ * <listitem><para>
+ * During the parsing process GstBaseAudioDecoderClass will handle both
+ * srcpad and sinkpad events. Sink events will be passed to subclass
+ * if @event callback has been provided.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * <listitem>
+ * <itemizedlist><title>Shutdown phase</title>
+ * <listitem><para>
+ * GstBaseAudioDecoder class calls @stop to inform the subclass that data
+ * parsing will be stopped.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * </orderedlist>
+ *
+ * Subclass is responsible for providing pad template caps for
+ * source and sink pads. The pads need to be named "sink" and "src". It also
+ * needs to set the fixed caps on srcpad, when the format is ensured. This
+ * is typically when base class calls subclass' @set_format function, though
+ * it might be delayed until calling @gst_base_audio_decoder_finish_frame.
+ *
+ * In summary, above process should have subclass concentrating on
+ * codec data processing while leaving other matters to base class,
+ * such as most notably timestamp handling. While it may exert more control
+ * in this area (see e.g. @pre_push), it is very much not recommended.
+ *
+ * In particular, base class will try to arrange for perfect output timestamps
+ * as much as possible while tracking upstream timestamps.
+ * To this end, if deviation between the next ideal expected perfect timestamp
+ * and upstream exceeds #GstBaseAudioDecoder:tolerance, then resync to upstream
+ * occurs (which would happen always if the tolerance mechanism is disabled).
+ *
+ * In non-live pipelines, baseclass can also (configurably) arrange for
+ * output buffer aggregation which may help to redue large(r) numbers of
+ * small(er) buffers being pushed and processed downstream.
+ *
+ * On the other hand, it should be noted that baseclass only provides limited
+ * seeking support (upon explicit subclass request), as full-fledged support
+ * should rather be left to upstream demuxer, parser or alike. This simple
+ * approach caters for seeking and duration reporting using estimated input
+ * bitrates.
+ *
+ * Things that subclass need to take care of:
+ * <itemizedlist>
+ * <listitem><para>Provide pad templates</para></listitem>
+ * <listitem><para>
+ * Set source pad caps when appropriate
+ * </para></listitem>
+ * <listitem><para>
+ * Set user-configurable properties to sane defaults for format and
+ * implementing codec at hand, and convey some subclass capabilities and
+ * expectations in context.
+ * </para></listitem>
+ * <listitem><para>
+ * Accept data in @handle_frame and provide encoded results to
+ * @gst_base_audio_decoder_finish_frame. If it is prepared to perform
+ * PLC, it should also accept NULL data in @handle_frame and provide for
+ * data for indicated duration.
+ * </para></listitem>
+ * </itemizedlist>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstbaseaudiodecoder.h"
+#include <gst/audio/audio.h>
+#include <gst/base/gstadapter.h>
+#include <gst/pbutils/descriptions.h>
+
+#include <string.h>
+
+GST_DEBUG_CATEGORY (baseaudiodecoder_debug);
+#define GST_CAT_DEFAULT baseaudiodecoder_debug
+
+#define GST_BASE_AUDIO_DECODER_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_DECODER, \
+ GstBaseAudioDecoderPrivate))
+
+enum
+{
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_LATENCY,
+ PROP_TOLERANCE,
+ PROP_PLC
+};
+
+#define DEFAULT_LATENCY 0
+#define DEFAULT_TOLERANCE 0
+#define DEFAULT_PLC FALSE
+
+struct _GstBaseAudioDecoderPrivate
+{
+ /* activation status */
+ gboolean active;
+
+ /* input base/first ts as basis for output ts */
+ GstClockTime base_ts;
+ /* input samples processed and sent downstream so far (w.r.t. base_ts) */
+ guint64 samples;
+
+ /* collected input data */
+ GstAdapter *adapter;
+ /* tracking input ts for changes */
+ GstClockTime prev_ts;
+ /* frames obtained from input */
+ GQueue frames;
+ /* collected output data */
+ GstAdapter *adapter_out;
+ /* ts and duration for output data collected above */
+ GstClockTime out_ts, out_dur;
+ /* mark outgoing discont */
+ gboolean discont;
+
+ /* subclass gave all it could already */
+ gboolean drained;
+ /* subclass currently being forcibly drained */
+ gboolean force;
+
+ /* input bps estimatation */
+ /* global in bytes seen */
+ guint64 bytes_in;
+ /* global samples sent out */
+ guint64 samples_out;
+ /* bytes flushed during parsing */
+ guint sync_flush;
+ /* error count */
+ gint error_count;
+ /* codec id tag */
+ GstTagList *taglist;
+
+ /* whether circumstances allow output aggregation */
+ gint agg;
+
+ /* reverse playback queues */
+ /* collect input */
+ GList *gather;
+ /* to-be-decoded */
+ GList *decode;
+ /* reversed output */
+ GList *queued;
+
+ /* context storage */
+ GstBaseAudioDecoderContext ctx;
+};
+
+
+static void gst_base_audio_decoder_finalize (GObject * object);
+static void gst_base_audio_decoder_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_base_audio_decoder_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static void gst_base_audio_decoder_clear_queues (GstBaseAudioDecoder * dec);
+static GstFlowReturn gst_base_audio_decoder_chain_reverse (GstBaseAudioDecoder *
+ dec, GstBuffer * buf);
+
+static GstStateChangeReturn gst_base_audio_decoder_change_state (GstElement *
+ element, GstStateChange transition);
+static gboolean gst_base_audio_decoder_sink_event (GstPad * pad,
+ GstEvent * event);
+static gboolean gst_base_audio_decoder_src_event (GstPad * pad,
+ GstEvent * event);
+static gboolean gst_base_audio_decoder_sink_setcaps (GstPad * pad,
+ GstCaps * caps);
+static gboolean gst_base_audio_decoder_src_setcaps (GstPad * pad,
+ GstCaps * caps);
+static GstFlowReturn gst_base_audio_decoder_chain (GstPad * pad,
+ GstBuffer * buf);
+static gboolean gst_base_audio_decoder_src_query (GstPad * pad,
+ GstQuery * query);
+static gboolean gst_base_audio_decoder_sink_query (GstPad * pad,
+ GstQuery * query);
+static const GstQueryType *gst_base_audio_decoder_get_query_types (GstPad *
+ pad);
+static void gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec,
+ gboolean full);
+
+
+GST_BOILERPLATE (GstBaseAudioDecoder, gst_base_audio_decoder, GstElement,
+ GST_TYPE_ELEMENT);
+
+static void
+gst_base_audio_decoder_base_init (gpointer g_class)
+{
+}
+
+static void
+gst_base_audio_decoder_class_init (GstBaseAudioDecoderClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *element_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ element_class = GST_ELEMENT_CLASS (klass);
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ g_type_class_add_private (klass, sizeof (GstBaseAudioDecoderPrivate));
+
+ GST_DEBUG_CATEGORY_INIT (baseaudiodecoder_debug, "baseaudiodecoder", 0,
+ "baseaudiodecoder element");
+
+ gobject_class->set_property = gst_base_audio_decoder_set_property;
+ gobject_class->get_property = gst_base_audio_decoder_get_property;
+ gobject_class->finalize = gst_base_audio_decoder_finalize;
+
+ element_class->change_state = gst_base_audio_decoder_change_state;
+
+ /* Properties */
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_int64 ("latency", "Latency",
+ "Aggregate output data to a minimum of latency time (ns)",
+ 0, G_MAXINT64, DEFAULT_LATENCY,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TOLERANCE,
+ g_param_spec_int64 ("tolerance", "Tolerance",
+ "Perfect ts while timestamp jitter/imperfection within tolerance (ns)",
+ 0, G_MAXINT64, DEFAULT_TOLERANCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PLC,
+ g_param_spec_boolean ("plc", "Packet Loss Concealment",
+ "Perform packet loss concealment (if supported)",
+ DEFAULT_PLC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_base_audio_decoder_init (GstBaseAudioDecoder * dec,
+ GstBaseAudioDecoderClass * klass)
+{
+ GstPadTemplate *pad_template;
+
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_init");
+
+ dec->priv = GST_BASE_AUDIO_DECODER_GET_PRIVATE (dec);
+
+ /* Setup sink pad */
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
+ g_return_if_fail (pad_template != NULL);
+
+ dec->sinkpad = gst_pad_new_from_template (pad_template, "sink");
+ gst_pad_set_event_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_event));
+ gst_pad_set_setcaps_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_setcaps));
+ gst_pad_set_chain_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_chain));
+ gst_pad_set_query_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_query));
+ gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
+ GST_DEBUG_OBJECT (dec, "sinkpad created");
+
+ /* Setup source pad */
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
+ g_return_if_fail (pad_template != NULL);
+
+ dec->srcpad = gst_pad_new_from_template (pad_template, "src");
+ gst_pad_set_setcaps_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_setcaps));
+ gst_pad_set_event_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_event));
+ gst_pad_set_query_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_query));
+ gst_pad_set_query_type_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_decoder_get_query_types));
+ gst_pad_use_fixed_caps (dec->srcpad);
+ gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
+ GST_DEBUG_OBJECT (dec, "srcpad created");
+
+ dec->priv->adapter = gst_adapter_new ();
+ dec->priv->adapter_out = gst_adapter_new ();
+ g_queue_init (&dec->priv->frames);
+ dec->ctx = &dec->priv->ctx;
+
+ /* property default */
+ dec->latency = DEFAULT_LATENCY;
+ dec->tolerance = DEFAULT_TOLERANCE;
+
+ /* init state */
+ gst_base_audio_decoder_reset (dec, TRUE);
+ GST_DEBUG_OBJECT (dec, "init ok");
+}
+
+static void
+gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec, gboolean full)
+{
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_reset");
+
+ GST_OBJECT_LOCK (dec);
+
+ if (full) {
+ dec->priv->active = FALSE;
+ dec->priv->bytes_in = 0;
+ dec->priv->samples_out = 0;
+ dec->priv->agg = -1;
+ dec->priv->error_count = 0;
+ gst_base_audio_decoder_clear_queues (dec);
+
+ g_free (dec->ctx->state.channel_pos);
+ memset (dec->ctx, 0, sizeof (dec->ctx));
+
+ if (dec->priv->taglist) {
+ gst_tag_list_free (dec->priv->taglist);
+ dec->priv->taglist = NULL;
+ }
+
+ gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+ }
+
+ g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
+ g_queue_clear (&dec->priv->frames);
+ gst_adapter_clear (dec->priv->adapter);
+ gst_adapter_clear (dec->priv->adapter_out);
+ dec->priv->out_ts = GST_CLOCK_TIME_NONE;
+ dec->priv->out_dur = 0;
+ dec->priv->prev_ts = GST_CLOCK_TIME_NONE;
+ dec->priv->drained = TRUE;
+ dec->priv->base_ts = GST_CLOCK_TIME_NONE;
+ dec->priv->samples = 0;
+ dec->priv->discont = TRUE;
+ dec->priv->sync_flush = FALSE;
+
+ GST_OBJECT_UNLOCK (dec);
+}
+
+static void
+gst_base_audio_decoder_finalize (GObject * object)
+{
+ GstBaseAudioDecoder *dec;
+
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (object));
+ dec = GST_BASE_AUDIO_DECODER (object);
+
+ if (dec->priv->adapter) {
+ g_object_unref (dec->priv->adapter);
+ }
+ if (dec->priv->adapter_out) {
+ g_object_unref (dec->priv->adapter_out);
+ }
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* automagically perform sanity checking of src caps;
+ * also extracts output data format */
+static gboolean
+gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstBaseAudioDecoder *dec;
+ GstAudioState *state;
+ gboolean res = TRUE, changed;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ state = &dec->ctx->state;
+
+ GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
+
+ /* parse caps here to check subclass;
+ * also makes us aware of output format */
+ if (!gst_caps_is_fixed (caps))
+ goto refuse_caps;
+
+ /* adjust ts tracking to new sample rate */
+ if (GST_CLOCK_TIME_IS_VALID (dec->priv->base_ts) && state->rate) {
+ dec->priv->base_ts +=
+ GST_FRAMES_TO_CLOCK_TIME (dec->priv->samples, state->rate);
+ dec->priv->samples = 0;
+ }
+
+ if (!gst_base_audio_parse_caps (caps, state, &changed))
+ goto refuse_caps;
+
+ gst_object_unref (dec);
+ return res;
+
+ /* ERRORS */
+refuse_caps:
+ {
+ GST_WARNING_OBJECT (dec, "rejected caps %" GST_PTR_FORMAT, caps);
+ gst_object_unref (dec);
+ return res;
+ }
+}
+
+static gboolean
+gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean res = TRUE;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
+
+ /* NOTE pbutils only needed here */
+ /* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
+ if (dec->priv->taglist)
+ gst_tag_list_free (dec->priv->taglist);
+ dec->priv->taglist = gst_tag_list_new ();
+ gst_pb_utils_add_codec_description_to_tag_list (dec->priv->taglist,
+ GST_TAG_AUDIO_CODEC, caps);
+
+ if (klass->set_format)
+ res = klass->set_format (dec, caps);
+
+ g_object_unref (dec);
+ return res;
+}
+
+static void
+gst_base_audio_decoder_setup (GstBaseAudioDecoder * dec)
+{
+ GstQuery *query;
+ gboolean res;
+
+ /* check if in live pipeline, then latency messing is no-no */
+ query = gst_query_new_latency ();
+ res = gst_pad_peer_query (dec->sinkpad, query);
+ if (res) {
+ gst_query_parse_latency (query, &res, NULL, NULL);
+ res = !res;
+ }
+ gst_query_unref (query);
+
+ /* normalize to bool */
+ dec->priv->agg = ! !res;
+}
+
+/* mini aggregator combining output buffers into fewer larger ones,
+ * if so allowed/configured */
+static GstFlowReturn
+gst_base_audio_decoder_output (GstBaseAudioDecoder * dec, GstBuffer * buf)
+{
+ GstBaseAudioDecoderClass *klass;
+ GstBaseAudioDecoderPrivate *priv;
+ GstBaseAudioDecoderContext *ctx;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBuffer *inbuf = NULL;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+ priv = dec->priv;
+ ctx = dec->ctx;
+
+ if (G_UNLIKELY (priv->agg < 0))
+ gst_base_audio_decoder_setup (dec);
+
+ if (G_LIKELY (buf)) {
+ g_return_val_if_fail (ctx->state.bpf != 0, GST_FLOW_ERROR);
+
+ GST_LOG_OBJECT (dec, "output buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+ /* clip buffer */
+ buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->state.rate,
+ ctx->state.bpf);
+ if (G_UNLIKELY (!buf)) {
+ GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
+ } else {
+ GST_LOG_OBJECT (dec,
+ "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+ }
+ } else {
+ GST_DEBUG_OBJECT (dec, "no output buffer");
+ }
+
+again:
+ inbuf = NULL;
+ if (priv->agg && dec->latency > 0) {
+ gint av;
+ gboolean assemble = FALSE;
+ const GstClockTimeDiff tol = 10 * GST_MSECOND;
+ GstClockTimeDiff diff = -100 * GST_MSECOND;
+
+ av = gst_adapter_available (priv->adapter_out);
+ if (G_UNLIKELY (!buf)) {
+ /* forcibly send current */
+ assemble = TRUE;
+ GST_LOG_OBJECT (dec, "forcing fragment flush");
+ } else if (av && (!GST_BUFFER_TIMESTAMP_IS_VALID (buf) ||
+ !GST_CLOCK_TIME_IS_VALID (priv->out_ts) ||
+ ((diff = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP (buf),
+ priv->out_ts + priv->out_dur)) > tol) || diff < -tol)) {
+ assemble = TRUE;
+ GST_LOG_OBJECT (dec, "buffer %d ms apart from current fragment",
+ (gint) (diff / GST_MSECOND));
+ } else {
+ /* add or start collecting */
+ if (!av) {
+ GST_LOG_OBJECT (dec, "starting new fragment");
+ priv->out_ts = GST_BUFFER_TIMESTAMP (buf);
+ } else {
+ GST_LOG_OBJECT (dec, "adding to fragment");
+ }
+ gst_adapter_push (priv->adapter_out, buf);
+ priv->out_dur += GST_BUFFER_DURATION (buf);
+ av += GST_BUFFER_SIZE (buf);
+ buf = NULL;
+ }
+ if (priv->out_dur > dec->latency)
+ assemble = TRUE;
+ if (av && assemble) {
+ GST_LOG_OBJECT (dec, "assembling fragment");
+ inbuf = buf;
+ buf = gst_adapter_take_buffer (priv->adapter_out, av);
+ GST_BUFFER_TIMESTAMP (buf) = priv->out_ts;
+ GST_BUFFER_DURATION (buf) = priv->out_dur;
+ priv->out_ts = GST_CLOCK_TIME_NONE;
+ priv->out_dur = 0;
+ }
+ }
+
+ if (G_LIKELY (buf)) {
+
+ /* decorate */
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
+
+ if (G_UNLIKELY (priv->discont)) {
+ GST_LOG_OBJECT (dec, "marking discont");
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+ priv->discont = FALSE;
+ }
+
+ if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
+ /* duration should always be valid for raw audio */
+ g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
+ dec->segment.last_stop =
+ GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+ }
+
+ if (klass->pre_push) {
+ /* last chance for subclass to do some dirty stuff */
+ ret = klass->pre_push (dec, &buf);
+ if (ret != GST_FLOW_OK || !buf) {
+ GST_DEBUG_OBJECT (dec, "subclass returned %s, buf %p",
+ gst_flow_get_name (ret), buf);
+ if (buf)
+ gst_buffer_unref (buf);
+ goto exit;
+ }
+ }
+
+ GST_LOG_OBJECT (dec, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+ if (dec->segment.rate > 0.0) {
+ ret = gst_pad_push (dec->srcpad, buf);
+ GST_LOG_OBJECT (dec, "buffer pushed: %s", gst_flow_get_name (ret));
+ } else {
+ ret = GST_FLOW_OK;
+ priv->queued = g_list_prepend (priv->queued, buf);
+ GST_LOG_OBJECT (dec, "buffer queued");
+ }
+
+ exit:
+ if (inbuf) {
+ buf = inbuf;
+ goto again;
+ }
+ }
+
+ return ret;
+}
+
+GstFlowReturn
+gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf,
+ gint frames)
+{
+ GstBaseAudioDecoderPrivate *priv;
+ GstBaseAudioDecoderContext *ctx;
+ gint samples = 0;
+ GstClockTime ts, next_ts;
+
+ /* subclass should know what it is producing by now */
+ g_return_val_if_fail (buf == NULL || GST_PAD_CAPS (dec->srcpad) != NULL,
+ GST_FLOW_ERROR);
+ /* subclass should not hand us no data */
+ g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
+ GST_FLOW_ERROR);
+ /* no dummy calls please */
+ g_return_val_if_fail (frames != 0, GST_FLOW_ERROR);
+
+ priv = dec->priv;
+ ctx = dec->ctx;
+
+ GST_LOG_OBJECT (dec, "accepting %d bytes == %d samples for %d frames",
+ buf ? GST_BUFFER_SIZE (buf) : -1,
+ buf ? GST_BUFFER_SIZE (buf) / ctx->state.bpf : -1, frames);
+
+ /* output shoud be whole number of sample frames */
+ if (G_LIKELY (buf && ctx->state.bpf)) {
+ if (GST_BUFFER_SIZE (buf) % ctx->state.bpf)
+ goto wrong_buffer;
+ /* per channel least */
+ samples = GST_BUFFER_SIZE (buf) / ctx->state.bpf;
+ }
+
+ /* frame and ts book-keeping */
+ if (G_UNLIKELY (frames < 0)) {
+ if (G_UNLIKELY (-frames - 1 > priv->frames.length))
+ goto overflow;
+ frames = priv->frames.length + frames + 1;
+ } else if (G_UNLIKELY (frames > priv->frames.length)) {
+ if (G_LIKELY (!priv->force)) {
+ /* no way we can let this pass */
+ g_assert_not_reached ();
+ /* really no way */
+ goto overflow;
+ }
+ }
+
+ if (G_LIKELY (priv->frames.length))
+ ts = GST_BUFFER_TIMESTAMP (priv->frames.head->data);
+ else
+ ts = GST_CLOCK_TIME_NONE;
+
+ GST_DEBUG_OBJECT (dec, "leading frame ts %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (ts));
+
+ while (priv->frames.length && frames) {
+ gst_buffer_unref (g_queue_pop_head (&priv->frames));
+ dec->ctx->delay = dec->priv->frames.length;
+ frames--;
+ }
+
+ /* lock on */
+ if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
+ priv->base_ts = ts;
+ GST_DEBUG_OBJECT (dec, "base_ts now %" GST_TIME_FORMAT, GST_TIME_ARGS (ts));
+ }
+
+ if (G_UNLIKELY (!buf))
+ goto exit;
+
+ /* slightly convoluted approach caters for perfect ts if subclass desires */
+ if (GST_CLOCK_TIME_IS_VALID (ts)) {
+ if (dec->tolerance > 0) {
+ GstClockTimeDiff diff;
+
+ g_assert (GST_CLOCK_TIME_IS_VALID (priv->base_ts));
+ next_ts = priv->base_ts +
+ gst_util_uint64_scale (samples, GST_SECOND, ctx->state.rate);
+ GST_LOG_OBJECT (dec, "buffer is %d samples past base_ts %" GST_TIME_FORMAT
+ ", expected ts %" GST_TIME_FORMAT, samples,
+ GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
+ diff = GST_CLOCK_DIFF (next_ts, ts);
+ GST_LOG_OBJECT (dec, "ts diff %d ms", (gint) (diff / GST_MSECOND));
+ /* if within tolerance,
+ * discard buffer ts and carry on producing perfect stream,
+ * otherwise resync to ts */
+ if (G_UNLIKELY (diff < -dec->tolerance || diff > dec->tolerance)) {
+ GST_DEBUG_OBJECT (dec, "base_ts resync");
+ priv->base_ts = ts;
+ priv->samples = 0;
+ }
+ } else {
+ GST_DEBUG_OBJECT (dec, "base_ts resync");
+ priv->base_ts = ts;
+ priv->samples = 0;
+ }
+ }
+
+ /* delayed one-shot stuff until confirmed data */
+ if (priv->taglist) {
+ GST_DEBUG_OBJECT (dec, "codec tag %" GST_PTR_FORMAT, priv->taglist);
+ if (gst_tag_list_is_empty (priv->taglist)) {
+ gst_tag_list_free (priv->taglist);
+ } else {
+ gst_element_found_tags (GST_ELEMENT (dec), priv->taglist);
+ }
+ priv->taglist = NULL;
+ }
+
+ buf = gst_buffer_make_metadata_writable (buf);
+ if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
+ GST_BUFFER_TIMESTAMP (buf) =
+ priv->base_ts +
+ GST_FRAMES_TO_CLOCK_TIME (priv->samples, ctx->state.rate);
+ GST_BUFFER_DURATION (buf) = priv->base_ts +
+ GST_FRAMES_TO_CLOCK_TIME (priv->samples + samples, ctx->state.rate) -
+ GST_BUFFER_TIMESTAMP (buf);
+ } else {
+ GST_BUFFER_TIMESTAMP (buf) = GST_CLOCK_TIME_NONE;
+ GST_BUFFER_DURATION (buf) =
+ GST_FRAMES_TO_CLOCK_TIME (samples, ctx->state.rate);
+ }
+ priv->samples += samples;
+ priv->samples_out += samples;
+
+ /* we got data, so note things are looking up */
+ if (G_UNLIKELY (dec->priv->error_count))
+ dec->priv->error_count--;
+
+exit:
+ return gst_base_audio_decoder_output (dec, buf);
+
+ /* ERRORS */
+wrong_buffer:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, ENCODE, (NULL),
+ ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
+ ctx->state.bpf));
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
+ }
+overflow:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, ENCODE,
+ ("received more decoded frames %d than provided %d", frames,
+ priv->frames.length), (NULL));
+ if (buf)
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
+ }
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_handle_frame (GstBaseAudioDecoder * dec,
+ GstBaseAudioDecoderClass * klass, GstBuffer * buffer)
+{
+ if (G_LIKELY (buffer)) {
+ /* keep around for admin */
+ GST_LOG_OBJECT (dec, "tracking frame size %d, ts %" GST_TIME_FORMAT,
+ GST_BUFFER_SIZE (buffer),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+ g_queue_push_tail (&dec->priv->frames, buffer);
+ dec->ctx->delay = dec->priv->frames.length;
+ dec->priv->bytes_in += GST_BUFFER_SIZE (buffer);
+ } else {
+ GST_LOG_OBJECT (dec, "providing subclass with NULL frame");
+ }
+
+ return klass->handle_frame (dec, buffer);
+}
+
+/* maybe subclass configurable instead, but this allows for a whole lot of
+ * raw samples, so at least quite some encoded ... */
+#define GST_BASE_AUDIO_DECODER_MAX_SYNC 10 * 8 * 2 * 1024
+
+static GstFlowReturn
+gst_base_audio_decoder_push_buffers (GstBaseAudioDecoder * dec, gboolean force)
+{
+ GstBaseAudioDecoderClass *klass;
+ GstBaseAudioDecoderPrivate *priv;
+ GstBaseAudioDecoderContext *ctx;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBuffer *buffer;
+ gint av, flush;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+ priv = dec->priv;
+ ctx = dec->ctx;
+
+ g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
+
+ av = gst_adapter_available (priv->adapter);
+ GST_DEBUG_OBJECT (dec, "available: %d", av);
+
+ while (ret == GST_FLOW_OK) {
+
+ flush = 0;
+ ctx->eos = force;
+
+ if (G_LIKELY (av)) {
+ gint len;
+ GstClockTime ts;
+
+ /* parse if needed */
+ if (klass->parse) {
+ gint offset = 0;
+
+ /* limited (legacy) parsing; avoid whole of baseparse */
+ GST_DEBUG_OBJECT (dec, "parsing available: %d", av);
+ /* piggyback sync state on discont */
+ ctx->sync = !priv->discont;
+ ret = klass->parse (dec, priv->adapter, &offset, &len);
+
+ g_assert (offset <= av);
+ if (offset) {
+ /* jumped a bit */
+ GST_DEBUG_OBJECT (dec, "setting DISCONT");
+ gst_adapter_flush (priv->adapter, offset);
+ flush = offset;
+ /* avoid parsing indefinitely */
+ priv->sync_flush += offset;
+ if (priv->sync_flush > GST_BASE_AUDIO_DECODER_MAX_SYNC)
+ goto parse_failed;
+ }
+
+ if (ret == GST_FLOW_UNEXPECTED) {
+ GST_LOG_OBJECT (dec, "no frame yet");
+ ret = GST_FLOW_OK;
+ break;
+ } else if (ret == GST_FLOW_OK) {
+ GST_LOG_OBJECT (dec, "frame at offset %d of length %d", offset, len);
+ g_assert (offset + len <= av);
+ priv->sync_flush = 0;
+ } else {
+ break;
+ }
+ } else {
+ len = av;
+ }
+ /* track upstream ts, but do not get stuck if nothing new upstream */
+ ts = gst_adapter_prev_timestamp (priv->adapter, NULL);
+ if (ts == priv->prev_ts) {
+ GST_LOG_OBJECT (dec, "ts == prev_ts; discarding");
+ ts = GST_CLOCK_TIME_NONE;
+ } else {
+ priv->prev_ts = ts;
+ }
+ buffer = gst_adapter_take_buffer (priv->adapter, len);
+ buffer = gst_buffer_make_metadata_writable (buffer);
+ GST_BUFFER_TIMESTAMP (buffer) = ts;
+ flush += len;
+ } else {
+ if (!force)
+ break;
+ buffer = NULL;
+ }
+
+ ret = gst_base_audio_decoder_handle_frame (dec, klass, buffer);
+
+ /* do not keep pushing it ... */
+ if (G_UNLIKELY (!av)) {
+ priv->drained = TRUE;
+ break;
+ }
+
+ av -= flush;
+ g_assert (av >= 0);
+ }
+
+ GST_LOG_OBJECT (dec, "done pushing to subclass");
+ return ret;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("failed to parse stream"));
+ return GST_FLOW_ERROR;
+ }
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_drain (GstBaseAudioDecoder * dec)
+{
+ GstFlowReturn ret;
+
+ if (dec->priv->drained)
+ return GST_FLOW_OK;
+ else {
+ /* dispatch reverse pending buffers */
+ /* chain eventually calls upon drain as well, but by that time
+ * gather list should be clear, so ok ... */
+ if (dec->segment.rate < 0.0 && dec->priv->gather)
+ gst_base_audio_decoder_chain_reverse (dec, NULL);
+ /* have subclass give all it can */
+ ret = gst_base_audio_decoder_push_buffers (dec, TRUE);
+ /* ensure all output sent */
+ ret = gst_base_audio_decoder_output (dec, NULL);
+ /* everything should be away now */
+ if (dec->priv->frames.length) {
+ /* not fatal/impossible though if subclass/codec eats stuff */
+ GST_WARNING_OBJECT (dec, "still %d frames left after draining",
+ dec->priv->frames.length);
+ g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
+ g_queue_clear (&dec->priv->frames);
+ }
+ /* discard (unparsed) leftover */
+ gst_adapter_clear (dec->priv->adapter);
+
+ return ret;
+ }
+}
+
+/* hard == FLUSH, otherwise discont */
+static GstFlowReturn
+gst_base_audio_decoder_flush (GstBaseAudioDecoder * dec, gboolean hard)
+{
+ GstBaseAudioDecoderClass *klass;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ GST_LOG_OBJECT (dec, "flush hard %d", hard);
+
+ if (!hard) {
+ ret = gst_base_audio_decoder_drain (dec);
+ } else {
+ gst_base_audio_decoder_clear_queues (dec);
+ gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+ dec->priv->error_count = 0;
+ }
+ /* only bother subclass with flushing if known it is already alive
+ * and kicking out stuff */
+ if (klass->flush && dec->priv->samples_out > 0)
+ klass->flush (dec, hard);
+ /* and get (re)set for the sequel */
+ gst_base_audio_decoder_reset (dec, FALSE);
+
+ return ret;
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_chain_forward (GstBaseAudioDecoder * dec,
+ GstBuffer * buffer)
+{
+ GstFlowReturn ret;
+
+ /* grab buffer */
+ gst_adapter_push (dec->priv->adapter, buffer);
+ buffer = NULL;
+ /* new stuff, so we can push subclass again */
+ dec->priv->drained = FALSE;
+
+ /* hand to subclass */
+ ret = gst_base_audio_decoder_push_buffers (dec, FALSE);
+
+ GST_LOG_OBJECT (dec, "chain-done");
+ return ret;
+}
+
+static void
+gst_base_audio_decoder_clear_queues (GstBaseAudioDecoder * dec)
+{
+ GstBaseAudioDecoderPrivate *priv = dec->priv;
+
+ g_list_foreach (priv->queued, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (priv->queued);
+ priv->queued = NULL;
+ g_list_foreach (priv->gather, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (priv->gather);
+ priv->gather = NULL;
+ g_list_foreach (priv->decode, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (priv->decode);
+ priv->decode = NULL;
+}
+
+/*
+ * Input:
+ * Buffer decoding order: 7 8 9 4 5 6 3 1 2 EOS
+ * Discont flag: D D D D
+ *
+ * - Each Discont marks a discont in the decoding order.
+ *
+ * for vorbis, each buffer is a keyframe when we have the previous
+ * buffer. This means that to decode buffer 7, we need buffer 6, which
+ * arrives out of order.
+ *
+ * we first gather buffers in the gather queue until we get a DISCONT. We
+ * prepend each incomming buffer so that they are in reversed order.
+ *
+ * gather queue: 9 8 7
+ * decode queue:
+ * output queue:
+ *
+ * When a DISCONT is received (buffer 4), we move the gather queue to the
+ * decode queue. This is simply done be taking the head of the gather queue
+ * and prepending it to the decode queue. This yields:
+ *
+ * gather queue:
+ * decode queue: 7 8 9
+ * output queue:
+ *
+ * Then we decode each buffer in the decode queue in order and put the output
+ * buffer in the output queue. The first buffer (7) will not produce any output
+ * because it needs the previous buffer (6) which did not arrive yet. This
+ * yields:
+ *
+ * gather queue:
+ * decode queue: 7 8 9
+ * output queue: 9 8
+ *
+ * Then we remove the consumed buffers from the decode queue. Buffer 7 is not
+ * completely consumed, we need to keep it around for when we receive buffer
+ * 6. This yields:
+ *
+ * gather queue:
+ * decode queue: 7
+ * output queue: 9 8
+ *
+ * Then we accumulate more buffers:
+ *
+ * gather queue: 6 5 4
+ * decode queue: 7
+ * output queue:
+ *
+ * prepending to the decode queue on DISCONT yields:
+ *
+ * gather queue:
+ * decode queue: 4 5 6 7
+ * output queue:
+ *
+ * after decoding and keeping buffer 4:
+ *
+ * gather queue:
+ * decode queue: 4
+ * output queue: 7 6 5
+ *
+ * Etc..
+ */
+static GstFlowReturn
+gst_base_audio_decoder_flush_decode (GstBaseAudioDecoder * dec)
+{
+ GstBaseAudioDecoderPrivate *priv = dec->priv;
+ GstFlowReturn res = GST_FLOW_OK;
+ GList *walk;
+
+ walk = priv->decode;
+
+ GST_DEBUG_OBJECT (dec, "flushing buffers to decoder");
+
+ /* clear buffer and decoder state */
+ gst_base_audio_decoder_flush (dec, FALSE);
+
+ while (walk) {
+ GList *next;
+ GstBuffer *buf = GST_BUFFER_CAST (walk->data);
+
+ GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT,
+ buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
+
+ next = g_list_next (walk);
+ /* decode buffer, resulting data prepended to output queue */
+ gst_buffer_ref (buf);
+ res = gst_base_audio_decoder_chain_forward (dec, buf);
+
+ /* if we generated output, we can discard the buffer, else we
+ * keep it in the queue */
+ if (priv->queued) {
+ GST_DEBUG_OBJECT (dec, "decoded buffer to %p", priv->queued->data);
+ priv->decode = g_list_delete_link (priv->decode, walk);
+ gst_buffer_unref (buf);
+ } else {
+ GST_DEBUG_OBJECT (dec, "buffer did not decode, keeping");
+ }
+ walk = next;
+ }
+
+ /* drain any aggregation (or otherwise) leftover */
+ gst_base_audio_decoder_drain (dec);
+
+ /* now send queued data downstream */
+ while (priv->queued) {
+ GstBuffer *buf = GST_BUFFER_CAST (priv->queued->data);
+
+ if (G_LIKELY (res == GST_FLOW_OK)) {
+ GST_DEBUG_OBJECT (dec, "pushing buffer %p of size %u, "
+ "time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
+ GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+ /* should be already, but let's be sure */
+ buf = gst_buffer_make_metadata_writable (buf);
+ /* avoid stray DISCONT from forward processing,
+ * which have no meaning in reverse pushing */
+ GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
+ res = gst_pad_push (dec->srcpad, buf);
+ } else {
+ gst_buffer_unref (buf);
+ }
+
+ priv->queued = g_list_delete_link (priv->queued, priv->queued);
+ }
+
+ return res;
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_chain_reverse (GstBaseAudioDecoder * dec,
+ GstBuffer * buf)
+{
+ GstBaseAudioDecoderPrivate *priv = dec->priv;
+ GstFlowReturn result = GST_FLOW_OK;
+
+ /* if we have a discont, move buffers to the decode list */
+ if (!buf || GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
+ GST_DEBUG_OBJECT (dec, "received discont");
+ while (priv->gather) {
+ GstBuffer *gbuf;
+
+ gbuf = GST_BUFFER_CAST (priv->gather->data);
+ /* remove from the gather list */
+ priv->gather = g_list_delete_link (priv->gather, priv->gather);
+ /* copy to decode queue */
+ priv->decode = g_list_prepend (priv->decode, gbuf);
+ }
+ /* decode stuff in the decode queue */
+ gst_base_audio_decoder_flush_decode (dec);
+ }
+
+ if (G_LIKELY (buf)) {
+ GST_DEBUG_OBJECT (dec, "gathering buffer %p of size %u, "
+ "time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
+ GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+ /* add buffer to gather queue */
+ priv->gather = g_list_prepend (priv->gather, buf);
+ }
+
+ return result;
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
+{
+ GstBaseAudioDecoder *dec;
+ GstFlowReturn ret;
+
+ dec = GST_BASE_AUDIO_DECODER (GST_PAD_PARENT (pad));
+
+ GST_LOG_OBJECT (dec,
+ "received buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
+
+ if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+ gint64 samples, ts;
+
+ /* track present position */
+ ts = dec->priv->base_ts;
+ samples = dec->priv->samples;
+
+ GST_DEBUG_OBJECT (dec, "handling discont");
+ gst_base_audio_decoder_flush (dec, FALSE);
+ dec->priv->discont = TRUE;
+
+ /* buffer may claim DISCONT loudly, if it can't tell us where we are now,
+ * we'll stick to where we were ...
+ * Particularly useful/needed for upstream BYTE based */
+ if (dec->segment.rate > 0.0 && !GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
+ GST_DEBUG_OBJECT (dec, "... but restoring previous ts tracking");
+ dec->priv->base_ts = ts;
+ dec->priv->samples = samples;
+ }
+ }
+
+ if (dec->segment.rate > 0.0)
+ ret = gst_base_audio_decoder_chain_forward (dec, buffer);
+ else
+ ret = gst_base_audio_decoder_chain_reverse (dec, buffer);
+
+ return ret;
+}
+
+/* perform upstream byte <-> time conversion (duration, seeking)
+ * if subclass allows and if enough data for moderately decent conversion */
+static inline gboolean
+gst_base_audio_decoder_do_byte (GstBaseAudioDecoder * dec)
+{
+ return dec->ctx->do_byte_time && dec->ctx->state.bpf &&
+ dec->ctx->state.rate <= dec->priv->samples_out;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
+ GstEvent * event)
+{
+ gboolean handled = FALSE;
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_NEWSEGMENT:
+ {
+ GstFormat format;
+ gdouble rate, arate;
+ gint64 start, stop, time;
+ gboolean update;
+
+ gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+ &start, &stop, &time);
+
+ if (format == GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (dec, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
+ " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
+ ", rate %g, applied_rate %g",
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
+ rate, arate);
+ } else {
+ GstFormat dformat = GST_FORMAT_TIME;
+
+ GST_DEBUG_OBJECT (dec, "received NEW_SEGMENT %" G_GINT64_FORMAT
+ " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
+ ", rate %g, applied_rate %g", start, stop, time, rate, arate);
+ /* handle newsegment resulting from legacy simple seeking */
+ /* note that we need to convert this whether or not enough data
+ * to handle initial newsegment */
+ if (dec->ctx->do_byte_time &&
+ gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, start,
+ &dformat, &start)) {
+ /* best attempt convert */
+ /* as these are only estimates, stop is kept open-ended to avoid
+ * premature cutting */
+ GST_DEBUG_OBJECT (dec, "converted to TIME start %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start));
+ format = GST_FORMAT_TIME;
+ time = start;
+ stop = GST_CLOCK_TIME_NONE;
+ /* replace event */
+ gst_event_unref (event);
+ event = gst_event_new_new_segment_full (update, rate, arate,
+ GST_FORMAT_TIME, start, stop, time);
+ } else {
+ GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
+ break;
+ }
+ }
+
+ /* finish current segment */
+ gst_base_audio_decoder_drain (dec);
+
+ if (update) {
+ /* time progressed without data, see if we can fill the gap with
+ * some concealment data */
+ GST_DEBUG_OBJECT (dec,
+ "segment update: plc %d, do_plc %d, last_stop %" GST_TIME_FORMAT,
+ dec->plc, dec->ctx->do_plc, GST_TIME_ARGS (dec->segment.last_stop));
+ if (dec->plc && dec->ctx->do_plc && dec->segment.rate > 0.0 &&
+ dec->segment.last_stop < start) {
+ GstBaseAudioDecoderClass *klass;
+ GstBuffer *buf;
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+ /* hand subclass empty frame with duration that needs covering */
+ buf = gst_buffer_new ();
+ GST_BUFFER_DURATION (buf) = start - dec->segment.last_stop;
+ /* best effort, not much error handling */
+ gst_base_audio_decoder_handle_frame (dec, klass, buf);
+ }
+ } else {
+ /* prepare for next one */
+ gst_base_audio_decoder_flush (dec, FALSE);
+ /* and that's where we time from,
+ * in case upstream does not come up with anything better
+ * (e.g. upstream BYTE) */
+ if (format != GST_FORMAT_TIME) {
+ dec->priv->base_ts = start;
+ dec->priv->samples = 0;
+ }
+ }
+
+ /* and follow along with segment */
+ gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
+ format, start, stop, time);
+
+ gst_pad_push_event (dec->srcpad, event);
+ handled = TRUE;
+ break;
+ }
+
+ case GST_EVENT_FLUSH_START:
+ break;
+
+ case GST_EVENT_FLUSH_STOP:
+ /* prepare for fresh start */
+ gst_base_audio_decoder_flush (dec, TRUE);
+ break;
+
+ case GST_EVENT_EOS:
+ gst_base_audio_decoder_drain (dec);
+ break;
+
+ default:
+ break;
+ }
+
+ return handled;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean handled = FALSE;
+ gboolean ret = TRUE;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
+ GST_EVENT_TYPE_NAME (event));
+
+ if (klass->event)
+ handled = klass->event (dec, event);
+
+ if (!handled)
+ handled = gst_base_audio_decoder_sink_eventfunc (dec, event);
+
+ if (!handled)
+ ret = gst_pad_event_default (pad, event);
+
+ GST_DEBUG_OBJECT (dec, "event handled");
+
+ gst_object_unref (dec);
+ return ret;
+}
+
+static gboolean
+gst_base_audio_decoder_do_seek (GstBaseAudioDecoder * dec, GstEvent * event)
+{
+ GstSeekFlags flags;
+ GstSeekType start_type, end_type;
+ GstFormat format;
+ gdouble rate;
+ gint64 start, start_time, end_time;
+ GstSegment seek_segment;
+ guint32 seqnum;
+
+ gst_event_parse_seek (event, &rate, &format, &flags, &start_type,
+ &start_time, &end_type, &end_time);
+
+ /* we'll handle plain open-ended flushing seeks with the simple approach */
+ if (rate != 1.0) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: rate");
+ return FALSE;
+ }
+
+ if (start_type != GST_SEEK_TYPE_SET) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: start time");
+ return FALSE;
+ }
+
+ if (end_type != GST_SEEK_TYPE_NONE ||
+ (end_type == GST_SEEK_TYPE_SET && end_time != GST_CLOCK_TIME_NONE)) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: end time");
+ return FALSE;
+ }
+
+ if (!(flags & GST_SEEK_FLAG_FLUSH)) {
+ GST_DEBUG_OBJECT (dec, "unsupported seek: not flushing");
+ return FALSE;
+ }
+
+ memcpy (&seek_segment, &dec->segment, sizeof (seek_segment));
+ gst_segment_set_seek (&seek_segment, rate, format, flags, start_type,
+ start_time, end_type, end_time, NULL);
+ start_time = seek_segment.last_stop;
+
+ format = GST_FORMAT_BYTES;
+ if (!gst_pad_query_convert (dec->sinkpad, GST_FORMAT_TIME, start_time,
+ &format, &start)) {
+ GST_DEBUG_OBJECT (dec, "conversion failed");
+ return FALSE;
+ }
+
+ seqnum = gst_event_get_seqnum (event);
+ event = gst_event_new_seek (1.0, GST_FORMAT_BYTES, flags,
+ GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE, -1);
+ gst_event_set_seqnum (event, seqnum);
+
+ GST_DEBUG_OBJECT (dec, "seeking to %" GST_TIME_FORMAT " at byte offset %"
+ G_GINT64_FORMAT, GST_TIME_ARGS (start_time), start);
+
+ return gst_pad_push_event (dec->sinkpad, event);
+}
+
+static gboolean
+gst_base_audio_decoder_src_event (GstPad * pad, GstEvent * event)
+{
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean res = FALSE;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
+ GST_EVENT_TYPE_NAME (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEEK:
+ {
+ GstFormat format, tformat;
+ gdouble rate;
+ GstSeekFlags flags;
+ GstSeekType cur_type, stop_type;
+ gint64 cur, stop;
+ gint64 tcur, tstop;
+ guint32 seqnum;
+
+ gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
+ &stop_type, &stop);
+ seqnum = gst_event_get_seqnum (event);
+
+ /* upstream gets a chance first */
+ if ((res = gst_pad_push_event (dec->sinkpad, event)))
+ break;
+
+ /* if upstream fails for a time seek, maybe we can help if allowed */
+ if (format == GST_FORMAT_TIME) {
+ if (gst_base_audio_decoder_do_byte (dec))
+ res = gst_base_audio_decoder_do_seek (dec, event);
+ break;
+ }
+
+ /* ... though a non-time seek can be aided as well */
+ /* First bring the requested format to time */
+ tformat = GST_FORMAT_TIME;
+ if (!(res = gst_pad_query_convert (pad, format, cur, &tformat, &tcur)))
+ goto convert_error;
+ if (!(res = gst_pad_query_convert (pad, format, stop, &tformat, &tstop)))
+ goto convert_error;
+
+ /* then seek with time on the peer */
+ event = gst_event_new_seek (rate, GST_FORMAT_TIME,
+ flags, cur_type, tcur, stop_type, tstop);
+ gst_event_set_seqnum (event, seqnum);
+
+ res = gst_pad_push_event (dec->sinkpad, event);
+ break;
+ }
+ default:
+ res = gst_pad_push_event (dec->sinkpad, event);
+ break;
+ }
+done:
+ gst_object_unref (dec);
+
+ return res;
+
+ /* ERRORS */
+convert_error:
+ {
+ GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
+ goto done;
+ }
+}
+
+static gboolean
+gst_base_audio_decoder_sink_query (GstPad * pad, GstQuery * query)
+{
+ gboolean res = TRUE;
+ GstBaseAudioDecoder *dec;
+
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_FORMATS:
+ {
+ gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
+ res = TRUE;
+ break;
+ }
+ case GST_QUERY_CONVERT:
+ {
+ GstFormat src_fmt, dest_fmt;
+ gint64 src_val, dest_val;
+
+ gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+ if (!(res = gst_base_audio_encoded_audio_convert (&dec->ctx->state,
+ dec->priv->bytes_in, dec->priv->samples_out,
+ src_fmt, src_val, &dest_fmt, &dest_val)))
+ goto error;
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+error:
+ gst_object_unref (dec);
+ return res;
+}
+
+static const GstQueryType *
+gst_base_audio_decoder_get_query_types (GstPad * pad)
+{
+ static const GstQueryType gst_base_audio_decoder_src_query_types[] = {
+ GST_QUERY_POSITION,
+ GST_QUERY_DURATION,
+ GST_QUERY_CONVERT,
+ GST_QUERY_LATENCY,
+ 0
+ };
+
+ return gst_base_audio_decoder_src_query_types;
+}
+
+/* FIXME ? are any of these queries (other than latency) a decoder's business ??
+ * also, the conversion stuff might seem to make sense, but seems to not mind
+ * segment stuff etc at all
+ * Supposedly that's backward compatibility ... */
+static gboolean
+gst_base_audio_decoder_src_query (GstPad * pad, GstQuery * query)
+{
+ GstBaseAudioDecoder *dec;
+ GstPad *peerpad;
+ gboolean res = FALSE;
+
+ dec = GST_BASE_AUDIO_DECODER (GST_PAD_PARENT (pad));
+ peerpad = gst_pad_get_peer (GST_PAD (dec->sinkpad));
+
+ GST_LOG_OBJECT (dec, "handling query: %" GST_PTR_FORMAT, query);
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_DURATION:
+ {
+ GstFormat format;
+
+ /* upstream in any case */
+ if ((res = gst_pad_query_default (pad, query)))
+ break;
+
+ gst_query_parse_duration (query, &format, NULL);
+ /* try answering TIME by converting from BYTE if subclass allows */
+ if (format == GST_FORMAT_TIME && gst_base_audio_decoder_do_byte (dec)) {
+ gint64 value;
+
+ format = GST_FORMAT_BYTES;
+ if (gst_pad_query_peer_duration (dec->sinkpad, &format, &value)) {
+ GST_LOG_OBJECT (dec, "upstream size %" G_GINT64_FORMAT, value);
+ format = GST_FORMAT_TIME;
+ if (gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, value,
+ &format, &value)) {
+ gst_query_set_duration (query, GST_FORMAT_TIME, value);
+ res = TRUE;
+ }
+ }
+ }
+ break;
+ }
+ case GST_QUERY_POSITION:
+ {
+ GstFormat format;
+ gint64 time, value;
+
+ if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
+ GST_LOG_OBJECT (dec, "returning peer response");
+ break;
+ }
+
+ /* we start from the last seen time */
+ time = dec->segment.last_stop;
+ /* correct for the segment values */
+ time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time);
+
+ GST_LOG_OBJECT (dec,
+ "query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time));
+
+ /* and convert to the final format */
+ gst_query_parse_position (query, &format, NULL);
+ if (!(res = gst_pad_query_convert (pad, GST_FORMAT_TIME, time,
+ &format, &value)))
+ break;
+
+ gst_query_set_position (query, format, value);
+
+ GST_LOG_OBJECT (dec,
+ "query %p: we return %" G_GINT64_FORMAT " (format %u)", query, value,
+ format);
+ break;
+ }
+ case GST_QUERY_FORMATS:
+ {
+ gst_query_set_formats (query, 3,
+ GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
+ res = TRUE;
+ break;
+ }
+ case GST_QUERY_CONVERT:
+ {
+ GstFormat src_fmt, dest_fmt;
+ gint64 src_val, dest_val;
+
+ gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+ if (!(res = gst_base_audio_raw_audio_convert (&dec->ctx->state,
+ src_fmt, src_val, &dest_fmt, &dest_val)))
+ break;
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ break;
+ }
+ case GST_QUERY_LATENCY:
+ {
+ if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
+ gboolean live;
+ GstClockTime min_latency, max_latency;
+
+ gst_query_parse_latency (query, &live, &min_latency, &max_latency);
+ GST_DEBUG_OBJECT (dec, "Peer latency: live %d, min %"
+ GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
+ GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
+
+ GST_OBJECT_LOCK (dec);
+ /* add our latency */
+ if (min_latency != -1)
+ min_latency += dec->ctx->min_latency;
+ if (max_latency != -1)
+ max_latency += dec->ctx->max_latency;
+ GST_OBJECT_UNLOCK (dec);
+
+ gst_query_set_latency (query, live, min_latency, max_latency);
+ }
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+ gst_object_unref (peerpad);
+ return res;
+}
+
+static gboolean
+gst_base_audio_decoder_stop (GstBaseAudioDecoder * dec)
+{
+ GstBaseAudioDecoderClass *klass;
+ gboolean ret = TRUE;
+
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_stop");
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ if (klass->stop) {
+ ret = klass->stop (dec);
+ }
+
+ /* clean up */
+ gst_base_audio_decoder_reset (dec, TRUE);
+
+ if (ret)
+ dec->priv->active = FALSE;
+
+ return TRUE;
+}
+
+static gboolean
+gst_base_audio_decoder_start (GstBaseAudioDecoder * dec)
+{
+ GstBaseAudioDecoderClass *klass;
+ gboolean ret = TRUE;
+
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_start");
+
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ /* arrange clean state */
+ gst_base_audio_decoder_reset (dec, TRUE);
+
+ if (klass->start) {
+ ret = klass->start (dec);
+ }
+
+ if (ret)
+ dec->priv->active = TRUE;
+
+ return TRUE;
+}
+
+static void
+gst_base_audio_decoder_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstBaseAudioDecoder *dec;
+
+ dec = GST_BASE_AUDIO_DECODER (object);
+
+ switch (prop_id) {
+ case PROP_LATENCY:
+ g_value_set_int64 (value, dec->latency);
+ break;
+ case PROP_TOLERANCE:
+ g_value_set_int64 (value, dec->tolerance);
+ break;
+ case PROP_PLC:
+ g_value_set_boolean (value, dec->plc);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_base_audio_decoder_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstBaseAudioDecoder *dec;
+
+ dec = GST_BASE_AUDIO_DECODER (object);
+
+ switch (prop_id) {
+ case PROP_LATENCY:
+ dec->latency = g_value_get_int64 (value);
+ break;
+ case PROP_TOLERANCE:
+ dec->tolerance = g_value_get_int64 (value);
+ break;
+ case PROP_PLC:
+ dec->plc = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstStateChangeReturn
+gst_base_audio_decoder_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstBaseAudioDecoder *codec;
+ GstBaseAudioDecoderClass *codec_class;
+ GstStateChangeReturn ret;
+
+ codec = GST_BASE_AUDIO_DECODER (element);
+ codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ if (!gst_base_audio_decoder_start (codec)) {
+ goto start_failed;
+ }
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ break;
+ default:
+ break;
+ }
+
+ ret = parent_class->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ if (!gst_base_audio_decoder_stop (codec)) {
+ goto stop_failed;
+ }
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+
+start_failed:
+ {
+ GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to start codec"));
+ return GST_STATE_CHANGE_FAILURE;
+ }
+stop_failed:
+ {
+ GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to stop codec"));
+ return GST_STATE_CHANGE_FAILURE;
+ }
+}
+
+GstFlowReturn
+_gst_base_audio_decoder_error (GstBaseAudioDecoder * dec, gint weight,
+ GQuark domain, gint code, gchar * txt, gchar * dbg, const gchar * file,
+ const gchar * function, gint line)
+{
+ if (txt)
+ GST_WARNING_OBJECT (dec, "error: %s", txt);
+ if (dbg)
+ GST_WARNING_OBJECT (dec, "error: %s", dbg);
+ dec->priv->error_count += weight;
+ dec->priv->discont = TRUE;
+ if (dec->ctx->max_errors < dec->priv->error_count) {
+ gst_element_message_full (GST_ELEMENT (dec), GST_MESSAGE_ERROR,
+ domain, code, txt, dbg, file, function, line);
+ return GST_FLOW_ERROR;
+ } else {
+ return GST_FLOW_OK;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral Quiroga <itoral@igalia.com>
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef _GST_BASE_AUDIO_DECODER_H_
+#define _GST_BASE_AUDIO_DECODER_H_
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "GstBaseAudioDecoder is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/gst.h>
+#include <gst/audio/gstbaseaudioutils.h>
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_BASE_AUDIO_DECODER \
+ (gst_base_audio_decoder_get_type())
+#define GST_BASE_AUDIO_DECODER(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoder))
+#define GST_BASE_AUDIO_DECODER_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
+#define GST_BASE_AUDIO_DECODER_GET_CLASS(obj) \
+ (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
+#define GST_IS_BASE_AUDIO_DECODER(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_DECODER))
+#define GST_IS_BASE_AUDIO_DECODER_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_DECODER))
+
+/**
+ * GST_BASE_AUDIO_DECODER_SINK_NAME:
+ *
+ * The name of the templates for the sink pad.
+ */
+#define GST_BASE_AUDIO_DECODER_SINK_NAME "sink"
+/**
+ * GST_BASE_AUDIO_DECODER_SRC_NAME:
+ *
+ * The name of the templates for the source pad.
+ */
+#define GST_BASE_AUDIO_DECODER_SRC_NAME "src"
+
+/**
+ * GST_BASE_AUDIO_DECODER_SRC_PAD:
+ * @obj: base audio codec instance
+ *
+ * Gives the pointer to the source #GstPad object of the element.
+ */
+#define GST_BASE_AUDIO_DECODER_SRC_PAD(obj) (((GstBaseAudioDecoder *) (obj))->srcpad)
+
+/**
+ * GST_BASE_AUDIO_DECODER_SINK_PAD:
+ * @obj: base audio codec instance
+ *
+ * Gives the pointer to the sink #GstPad object of the element.
+ */
+#define GST_BASE_AUDIO_DECODER_SINK_PAD(obj) (((GstBaseAudioDecoder *) (obj))->sinkpad)
+
+typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
+typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
+
+typedef struct _GstBaseAudioDecoderPrivate GstBaseAudioDecoderPrivate;
+typedef struct _GstBaseAudioDecoderContext GstBaseAudioDecoderContext;
+
+/* do not use this one, use macro below */
+GstFlowReturn _gst_base_audio_decoder_error (GstBaseAudioDecoder *dec, gint weight,
+ GQuark domain, gint code,
+ gchar *txt, gchar *debug,
+ const gchar *file, const gchar *function,
+ gint line);
+
+/**
+ * GST_BASE_AUDIO_DECODER_ERROR:
+ * @el: the base audio decoder element that generates the error
+ * @weight: element defined weight of the error, added to error count
+ * @domain: like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError)
+ * @code: error code defined for that domain (see #gstreamer-GstGError)
+ * @text: the message to display (format string and args enclosed in
+ * parentheses)
+ * @debug: debugging information for the message (format string and args
+ * enclosed in parentheses)
+ * @ret: variable to receive return value
+ *
+ * Utility function that audio decoder elements can use in case they encountered
+ * a data processing error that may be fatal for the current "data unit" but
+ * need not prevent subsequent decoding. Such errors are counted and if there
+ * are too many, as configured in the context's max_errors, the pipeline will
+ * post an error message and the application will be requested to stop further
+ * media processing. Otherwise, it is considered a "glitch" and only a warning
+ * is logged. In either case, @ret is set to the proper value to
+ * return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).
+ */
+#define GST_BASE_AUDIO_DECODER_ERROR(el, w, domain, code, text, debug, ret) \
+G_STMT_START { \
+ gchar *__txt = _gst_element_error_printf text; \
+ gchar *__dbg = _gst_element_error_printf debug; \
+ GstBaseAudioDecoder *dec = GST_BASE_AUDIO_DECODER (el); \
+ ret = _gst_base_audio_decoder_error (dec, w, GST_ ## domain ## _ERROR, \
+ GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \
+ GST_FUNCTION, __LINE__); \
+} G_STMT_END
+
+/**
+ * GstBaseAudioDecoderContext:
+ * @state: a #GstAudioState describing input audio format
+ * @eos: no (immediate) subsequent data in stream
+ * @sync: stream parsing in sync
+ * @delay: number of frames pending decoding (typically at least 1 for current)
+ * @do_plc: whether subclass is prepared to handle (packet) loss concealment
+ * @min_latency: min latency of element
+ * @max_latency: max latency of element
+ * @lookahead: decoder lookahead (in units of input rate samples)
+ *
+ * Transparent #GstBaseAudioEncoderContext data structure.
+ */
+struct _GstBaseAudioDecoderContext {
+ /* input */
+ /* (output) audio format */
+ GstAudioState state;
+
+ /* parsing state */
+ gboolean eos;
+ gboolean sync;
+
+ /* misc */
+ gint delay;
+
+ /* output */
+ gboolean do_plc;
+ gboolean do_byte_time;
+ gint max_errors;
+ /* MT-protected (with LOCK) */
+ GstClockTime min_latency;
+ GstClockTime max_latency;
+};
+
+/**
+ * GstBaseAudioDecoder:
+ *
+ * The opaque #GstBaseAudioDecoder data structure.
+ */
+struct _GstBaseAudioDecoder
+{
+ GstElement element;
+
+ /*< protected >*/
+ /* source and sink pads */
+ GstPad *sinkpad;
+ GstPad *srcpad;
+
+ /* MT-protected (with STREAM_LOCK) */
+ GstSegment segment;
+ GstBaseAudioDecoderContext *ctx;
+
+ /* properties */
+ GstClockTime latency;
+ GstClockTime tolerance;
+ gboolean plc;
+
+ /*< private >*/
+ GstBaseAudioDecoderPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+/**
+ * GstBaseAudioDecoderClass:
+ * @start: Optional.
+ * Called when the element starts processing.
+ * Allows opening external resources.
+ * @stop: Optional.
+ * Called when the element stops processing.
+ * Allows closing external resources.
+ * @set_format: Notifies subclass of incoming data format (caps).
+ * @parse: Optional.
+ * Allows chopping incoming data into manageable units (frames)
+ * for subsequent decoding. This division is at subclass
+ * discretion and may or may not correspond to 1 (or more)
+ * frames as defined by audio format.
+ * @handle_frame: Provides input data (or NULL to clear any remaining data)
+ * to subclass. Input data ref management is performed by
+ * base class, subclass should not care or intervene.
+ * @flush: Optional.
+ * Instructs subclass to clear any codec caches and discard
+ * any pending samples and not yet returned encoded data.
+ * @hard indicates whether a FLUSH is being processed,
+ * or otherwise a DISCONT (or conceptually similar).
+ * @event: Optional.
+ * Event handler on the sink pad. This function should return
+ * TRUE if the event was handled and should be discarded
+ * (i.e. not unref'ed).
+ * @pre_push: Optional.
+ * Called just prior to pushing (encoded data) buffer downstream.
+ * Subclass has full discretionary access to buffer,
+ * and a not OK flow return will abort downstream pushing.
+ *
+ * Subclasses can override any of the available virtual methods or not, as
+ * needed. At minimum @handle_frame (and likely @set_format) needs to be
+ * overridden.
+ */
+struct _GstBaseAudioDecoderClass
+{
+ GstElementClass parent_class;
+
+ /*< public >*/
+ /* virtual methods for subclasses */
+
+ gboolean (*start) (GstBaseAudioDecoder *dec);
+
+ gboolean (*stop) (GstBaseAudioDecoder *dec);
+
+ gboolean (*set_format) (GstBaseAudioDecoder *dec,
+ GstCaps *caps);
+
+ GstFlowReturn (*parse) (GstBaseAudioDecoder *dec,
+ GstAdapter *adapter,
+ gint *offset, gint *length);
+
+ GstFlowReturn (*handle_frame) (GstBaseAudioDecoder *dec,
+ GstBuffer *buffer);
+
+ void (*flush) (GstBaseAudioDecoder *dec, gboolean hard);
+
+ GstFlowReturn (*pre_push) (GstBaseAudioDecoder *dec,
+ GstBuffer **buffer);
+
+ gboolean (*event) (GstBaseAudioDecoder *dec,
+ GstEvent *event);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GstFlowReturn gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec,
+ GstBuffer * buf, gint frames);
+
+GType gst_base_audio_decoder_get_type (void);
+
+G_END_DECLS
+
+#endif
+
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:gstbaseaudioencoder
+ * @short_description: Base class for audio encoders
+ * @see_also: #GstBaseTransform
+ *
+ * This base class is for audio encoders turning raw audio samples into
+ * encoded audio data.
+ *
+ * GstBaseAudioEncoder and subclass should cooperate as follows.
+ * <orderedlist>
+ * <listitem>
+ * <itemizedlist><title>Configuration</title>
+ * <listitem><para>
+ * Initially, GstBaseAudioEncoder calls @start when the encoder element
+ * is activated, which allows subclass to perform any global setup.
+ * </para></listitem>
+ * <listitem><para>
+ * GstBaseAudioEncoder calls @set_format to inform subclass of the format
+ * of input audio data that it is about to receive. Subclass should
+ * setup for encoding and configure various base class context parameters
+ * appropriately, notably those directing desired input data handling.
+ * While unlikely, it might be called more than once, if changing input
+ * parameters require reconfiguration.
+ * </para></listitem>
+ * <listitem><para>
+ * GstBaseAudioEncoder calls @stop at end of all processing.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * As of configuration stage, and throughout processing, GstBaseAudioEncoder
+ * provides a GstBaseAudioEncoderContext that provides required context,
+ * e.g. describing the format of input audio data.
+ * Conversely, subclass can and should configure context to inform
+ * base class of its expectation w.r.t. buffer handling.
+ * <listitem>
+ * <itemizedlist>
+ * <title>Data processing</title>
+ * <listitem><para>
+ * Base class gathers input sample data (as directed by the context's
+ * frame_samples and frame_max) and provides this to subclass' @handle_frame.
+ * </para></listitem>
+ * <listitem><para>
+ * If codec processing results in encoded data, subclass should call
+ * @gst_base_audio_encoder_finish_frame to have encoded data pushed
+ * downstream. Alternatively, it might also call to indicate dropped
+ * (non-encoded) samples.
+ * </para></listitem>
+ * <listitem><para>
+ * Just prior to actually pushing a buffer downstream,
+ * it is passed to @pre_push.
+ * </para></listitem>
+ * <listitem><para>
+ * During the parsing process GstBaseAudioEncoderClass will handle both
+ * srcpad and sinkpad events. Sink events will be passed to subclass
+ * if @event callback has been provided.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * <listitem>
+ * <itemizedlist><title>Shutdown phase</title>
+ * <listitem><para>
+ * GstBaseAudioEncoder class calls @stop to inform the subclass that data
+ * parsing will be stopped.
+ * </para></listitem>
+ * </itemizedlist>
+ * </listitem>
+ * </orderedlist>
+ *
+ * Subclass is responsible for providing pad template caps for
+ * source and sink pads. The pads need to be named "sink" and "src". It also
+ * needs to set the fixed caps on srcpad, when the format is ensured. This
+ * is typically when base class calls subclass' @set_format function, though
+ * it might be delayed until calling @gst_base_audio_encoder_finish_frame.
+ *
+ * In summary, above process should have subclass concentrating on
+ * codec data processing while leaving other matters to base class,
+ * such as most notably timestamp handling. While it may exert more control
+ * in this area (see e.g. @pre_push), it is very much not recommended.
+ *
+ * In particular, base class will either favor tracking upstream timestamps
+ * (at the possible expense of jitter) or aim to arrange for a perfect stream of
+ * output timestamps, depending on #GstBaseAudioEncoder:perfect-ts.
+ * However, in the latter case, the input may not be so perfect or ideal, which
+ * is handled as follows. An input timestamp is compared with the expected
+ * timestamp as dictated by input sample stream and if the deviation is less
+ * than #GstBaseAudioEncoder:tolerance, the deviation is discarded.
+ * Otherwise, it is considered a discontuinity and subsequent output timestamp
+ * is resynced to the new position after performing configured discontinuity
+ * processing. In the non-perfect-ts case, an upstream variation exceeding
+ * tolerance only leads to marking DISCONT on subsequent outgoing
+ * (while timestamps are adjusted to upstream regardless of variation).
+ * While DISCONT is also marked in the perfect-ts case, this one optionally
+ * (see #GstBaseAudioEncoder:hard-resync)
+ * performs some additional steps, such as clipping of (early) input samples
+ * or draining all currently remaining input data, depending on the direction
+ * of the discontuinity.
+ *
+ * If perfect timestamps are arranged, it is also possible to request baseclass
+ * (usually set by subclass) to provide additional buffer metadata (in OFFSET
+ * and OFFSET_END) fields according to granule defined semantics currently
+ * needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count
+ * including buffer) and OFFSET_END to corresponding timestamp (as determined
+ * by same sample count and sample rate).
+ *
+ * Things that subclass need to take care of:
+ * <itemizedlist>
+ * <listitem><para>Provide pad templates</para></listitem>
+ * <listitem><para>
+ * Set source pad caps when appropriate
+ * </para></listitem>
+ * <listitem><para>
+ * Inform base class of buffer processing needs using context's
+ * frame_samples and frame_bytes.
+ * </para></listitem>
+ * <listitem><para>
+ * Set user-configurable properties to sane defaults for format and
+ * implementing codec at hand, e.g. those controlling timestamp behaviour
+ * and discontinuity processing.
+ * </para></listitem>
+ * <listitem><para>
+ * Accept data in @handle_frame and provide encoded results to
+ * @gst_base_audio_encoder_finish_frame.
+ * </para></listitem>
+ * </itemizedlist>
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "gstbaseaudioencoder.h"
+#include <gst/base/gstadapter.h>
+#include <gst/audio/audio.h>
+
+#include <stdlib.h>
+#include <string.h>
+
+
+GST_DEBUG_CATEGORY_STATIC (gst_base_audio_encoder_debug);
+#define GST_CAT_DEFAULT gst_base_audio_encoder_debug
+
+#define GST_BASE_AUDIO_ENCODER_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_ENCODER, \
+ GstBaseAudioEncoderPrivate))
+
+enum
+{
+ PROP_0,
+ PROP_PERFECT_TS,
+ PROP_GRANULE,
+ PROP_HARD_RESYNC,
+ PROP_TOLERANCE
+};
+
+#define DEFAULT_PERFECT_TS FALSE
+#define DEFAULT_GRANULE FALSE
+#define DEFAULT_HARD_RESYNC FALSE
+#define DEFAULT_TOLERANCE 40000000
+
+struct _GstBaseAudioEncoderPrivate
+{
+ /* activation status */
+ gboolean active;
+
+ /* input base/first ts as basis for output ts;
+ * kept nearly constant for perfect_ts,
+ * otherwise resyncs to upstream ts */
+ GstClockTime base_ts;
+ /* corresponding base granulepos */
+ gint64 base_gp;
+ /* input samples processed and sent downstream so far (w.r.t. base_ts) */
+ guint64 samples;
+
+ /* currently collected sample data */
+ GstAdapter *adapter;
+ /* offset in adapter up to which already supplied to encoder */
+ gint offset;
+ /* mark outgoing discont */
+ gboolean discont;
+ /* to guess duration of drained data */
+ GstClockTime last_duration;
+
+ /* subclass provided data in processing round */
+ gboolean got_data;
+ /* subclass gave all it could already */
+ gboolean drained;
+ /* subclass currently being forcibly drained */
+ gboolean force;
+
+ /* output bps estimatation */
+ /* global in samples seen */
+ guint64 samples_in;
+ /* global bytes sent out */
+ guint64 bytes_out;
+
+ /* context storage */
+ GstBaseAudioEncoderContext ctx;
+};
+
+
+static GstElementClass *parent_class = NULL;
+
+static void gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass *
+ klass);
+static void gst_base_audio_encoder_init (GstBaseAudioEncoder * parse,
+ GstBaseAudioEncoderClass * klass);
+
+GType
+gst_base_audio_encoder_get_type (void)
+{
+ static GType base_audio_encoder_type = 0;
+
+ if (!base_audio_encoder_type) {
+ static const GTypeInfo base_audio_encoder_info = {
+ sizeof (GstBaseAudioEncoderClass),
+ (GBaseInitFunc) NULL,
+ (GBaseFinalizeFunc) NULL,
+ (GClassInitFunc) gst_base_audio_encoder_class_init,
+ NULL,
+ NULL,
+ sizeof (GstBaseAudioEncoder),
+ 0,
+ (GInstanceInitFunc) gst_base_audio_encoder_init,
+ };
+ const GInterfaceInfo preset_interface_info = {
+ NULL, /* interface_init */
+ NULL, /* interface_finalize */
+ NULL /* interface_data */
+ };
+
+ base_audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
+ "GstBaseAudioEncoder", &base_audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
+
+ g_type_add_interface_static (base_audio_encoder_type, GST_TYPE_PRESET,
+ &preset_interface_info);
+ }
+ return base_audio_encoder_type;
+}
+
+static void gst_base_audio_encoder_finalize (GObject * object);
+static void gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc,
+ gboolean full);
+
+static void gst_base_audio_encoder_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_base_audio_encoder_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_base_audio_encoder_sink_activate_push (GstPad * pad,
+ gboolean active);
+
+static gboolean gst_base_audio_encoder_sink_event (GstPad * pad,
+ GstEvent * event);
+static gboolean gst_base_audio_encoder_sink_setcaps (GstPad * pad,
+ GstCaps * caps);
+static GstFlowReturn gst_base_audio_encoder_chain (GstPad * pad,
+ GstBuffer * buffer);
+static gboolean gst_base_audio_encoder_src_query (GstPad * pad,
+ GstQuery * query);
+static gboolean gst_base_audio_encoder_sink_query (GstPad * pad,
+ GstQuery * query);
+static const GstQueryType *gst_base_audio_encoder_get_query_types (GstPad *
+ pad);
+static GstCaps *gst_base_audio_encoder_sink_getcaps (GstPad * pad);
+
+
+static void
+gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ parent_class = g_type_class_peek_parent (klass);
+
+ GST_DEBUG_CATEGORY_INIT (gst_base_audio_encoder_debug, "baseaudioencoder", 0,
+ "baseaudioencoder element");
+
+ g_type_class_add_private (klass, sizeof (GstBaseAudioEncoderPrivate));
+
+ gobject_class->set_property = gst_base_audio_encoder_set_property;
+ gobject_class->get_property = gst_base_audio_encoder_get_property;
+
+ gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_audio_encoder_finalize);
+
+ /* properties */
+ g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
+ g_param_spec_boolean ("perfect-ts", "Perfect Timestamps",
+ "Favour perfect timestamps over tracking upstream timestamps",
+ DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_GRANULE,
+ g_param_spec_boolean ("granule", "Granule Marking",
+ "Apply granule semantics to buffer metadata (implies perfect-ts)",
+ DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
+ g_param_spec_boolean ("hard-resync", "Hard Resync",
+ "Perform clipping and sample flushing upon discontinuity",
+ DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_TOLERANCE,
+ g_param_spec_int64 ("tolerance", "Tolerance",
+ "Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
+ 0, G_MAXINT64, DEFAULT_TOLERANCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_base_audio_encoder_init (GstBaseAudioEncoder * enc,
+ GstBaseAudioEncoderClass * bclass)
+{
+ GstPadTemplate *pad_template;
+
+ GST_DEBUG_OBJECT (enc, "gst_base_audio_encoder_init");
+
+ enc->priv = GST_BASE_AUDIO_ENCODER_GET_PRIVATE (enc);
+
+ /* only push mode supported */
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
+ g_return_if_fail (pad_template != NULL);
+ enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
+ gst_pad_set_event_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_event));
+ gst_pad_set_setcaps_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_setcaps));
+ gst_pad_set_getcaps_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_getcaps));
+ gst_pad_set_query_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_query));
+ gst_pad_set_chain_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_chain));
+ gst_pad_set_activatepush_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_activate_push));
+ gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
+
+ GST_DEBUG_OBJECT (enc, "sinkpad created");
+
+ /* and we don't mind upstream traveling stuff that much ... */
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
+ g_return_if_fail (pad_template != NULL);
+ enc->srcpad = gst_pad_new_from_template (pad_template, "src");
+ gst_pad_set_query_function (enc->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_src_query));
+ gst_pad_set_query_type_function (enc->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_encoder_get_query_types));
+ gst_pad_use_fixed_caps (enc->srcpad);
+ gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
+ GST_DEBUG_OBJECT (enc, "src created");
+
+ enc->priv->adapter = gst_adapter_new ();
+ enc->ctx = &enc->priv->ctx;
+
+ /* property default */
+ enc->perfect_ts = DEFAULT_PERFECT_TS;
+ enc->hard_resync = DEFAULT_HARD_RESYNC;
+ enc->tolerance = DEFAULT_TOLERANCE;
+
+ /* init state */
+ gst_base_audio_encoder_reset (enc, TRUE);
+ GST_DEBUG_OBJECT (enc, "init ok");
+}
+
+static void
+gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
+{
+ GST_OBJECT_LOCK (enc);
+
+ if (full) {
+ enc->priv->active = FALSE;
+ enc->priv->samples_in = 0;
+ enc->priv->bytes_out = 0;
+ g_free (enc->ctx->state.channel_pos);
+ memset (enc->ctx, 0, sizeof (enc->ctx));
+ }
+
+ gst_segment_init (&enc->segment, GST_FORMAT_TIME);
+
+ gst_adapter_clear (enc->priv->adapter);
+ enc->priv->got_data = FALSE;
+ enc->priv->drained = TRUE;
+ enc->priv->offset = 0;
+ enc->priv->base_ts = GST_CLOCK_TIME_NONE;
+ enc->priv->base_gp = -1;
+ enc->priv->samples = 0;
+ enc->priv->discont = FALSE;
+
+ GST_OBJECT_UNLOCK (enc);
+}
+
+static void
+gst_base_audio_encoder_finalize (GObject * object)
+{
+ GstBaseAudioEncoder *enc = GST_BASE_AUDIO_ENCODER (object);
+
+ g_object_unref (enc->priv->adapter);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/**
+ * gst_base_audio_encoder_finish_frame:
+ * @enc: a #GstBaseAudioEncoder
+ * @buffer: encoded data
+ * @samples: number of samples (per channel) represented by encoded data
+ *
+ * Collects encoded data and/or pushes encoded data downstream.
+ * Source pad caps must be set when this is called. Depending on the nature
+ * of the (framing of) the format, subclass can decide whether to push
+ * encoded data directly or to collect various "frames" in a single buffer.
+ * Note that the latter behaviour is recommended whenever the format is allowed,
+ * as it incurs no additional latency and avoids otherwise generating a
+ * a multitude of (small) output buffers. If not explicitly pushed,
+ * any available encoded data is pushed at the end of each processing cycle,
+ * i.e. which encodes as much data as available input data allows.
+ *
+ * If @samples < 0, then best estimate is all samples provided to encoder
+ * (subclass) so far. @buf may be NULL, in which case next number of @samples
+ * are considered discarded, e.g. as a result of discontinuous transmission,
+ * and a discontinuity is marked (note that @buf == NULL => push == TRUE).
+ *
+ * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
+ */
+GstFlowReturn
+gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
+ gint samples)
+{
+ GstBaseAudioEncoderClass *klass;
+ GstBaseAudioEncoderPrivate *priv;
+ GstBaseAudioEncoderContext *ctx;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+ priv = enc->priv;
+ ctx = enc->ctx;
+
+ /* subclass should know what it is producing by now */
+ g_return_val_if_fail (GST_PAD_CAPS (enc->srcpad) != NULL, GST_FLOW_ERROR);
+ /* subclass should not hand us no data */
+ g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
+ GST_FLOW_ERROR);
+
+ GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
+ buf ? GST_BUFFER_SIZE (buf) : -1, samples);
+
+ /* mark subclass still alive and providing */
+ priv->got_data = TRUE;
+
+ /* remove corresponding samples from input */
+ if (samples < 0)
+ samples = (enc->priv->offset / ctx->state.bpf);
+
+ if (G_LIKELY (samples)) {
+ /* track upstream ts if so configured */
+ if (!enc->perfect_ts) {
+ guint64 ts, distance;
+
+ ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
+ g_assert (distance % ctx->state.bpf == 0);
+ distance /= ctx->state.bpf;
+ GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
+ GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
+ GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
+ GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
+ /* when draining adapter might be empty and no ts to offer */
+ if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
+ GstClockTimeDiff diff;
+ GstClockTime old_ts, next_ts;
+
+ /* passed into another buffer;
+ * mild check for discontinuity and only mark if so */
+ next_ts = ts +
+ gst_util_uint64_scale (distance, GST_SECOND, ctx->state.rate);
+ old_ts = priv->base_ts +
+ gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->state.rate);
+ diff = GST_CLOCK_DIFF (next_ts, old_ts);
+ GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
+ /* only mark discontinuity if beyond tolerance */
+ if (G_UNLIKELY (diff < -enc->tolerance || diff > enc->tolerance)) {
+ GST_DEBUG_OBJECT (enc, "marked discont");
+ priv->discont = TRUE;
+ }
+ GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
+ " at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
+ /* re-sync to upstream ts */
+ priv->base_ts = ts;
+ priv->samples = distance;
+ }
+ }
+ /* advance sample view */
+ if (G_UNLIKELY (samples * ctx->state.bpf > priv->offset)) {
+ if (G_LIKELY (!priv->force)) {
+ /* no way we can let this pass */
+ g_assert_not_reached ();
+ /* really no way */
+ goto overflow;
+ } else {
+ priv->offset = 0;
+ if (samples * ctx->state.bpf >= gst_adapter_available (priv->adapter))
+ gst_adapter_clear (priv->adapter);
+ else
+ gst_adapter_flush (priv->adapter, samples * ctx->state.bpf);
+ }
+ } else {
+ gst_adapter_flush (priv->adapter, samples * ctx->state.bpf);
+ priv->offset -= samples * ctx->state.bpf;
+ /* avoid subsequent stray prev_ts */
+ if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
+ gst_adapter_clear (priv->adapter);
+ }
+ /* sample count advanced below after buffer handling */
+ }
+
+ /* collect output */
+ if (G_LIKELY (buf)) {
+ GST_LOG_OBJECT (enc, "taking %d bytes for output", GST_BUFFER_SIZE (buf));
+ buf = gst_buffer_make_metadata_writable (buf);
+
+ /* decorate */
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (enc->srcpad));
+ if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
+ /* FIXME ? lookahead could lead to weird ts and duration ?
+ * (particularly if not in perfect mode) */
+ /* mind sample rounding and produce perfect output */
+ GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
+ gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
+ ctx->state.rate);
+ GST_DEBUG_OBJECT (enc, "out samples %d", samples);
+ if (G_LIKELY (samples > 0)) {
+ priv->samples += samples;
+ GST_BUFFER_DURATION (buf) = priv->base_ts +
+ gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
+ ctx->state.rate) - GST_BUFFER_TIMESTAMP (buf);
+ priv->last_duration = GST_BUFFER_DURATION (buf);
+ } else {
+ /* duration forecast in case of handling remainder;
+ * the last one is probably like the previous one ... */
+ GST_BUFFER_DURATION (buf) = priv->last_duration;
+ }
+ if (priv->base_gp >= 0) {
+ /* pamper oggmux */
+ /* FIXME: in longer run, muxer should take care of this ... */
+ /* offset_end = granulepos for ogg muxer */
+ GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples -
+ enc->ctx->lookahead;
+ /* offset = timestamp corresponding to granulepos for ogg muxer */
+ GST_BUFFER_OFFSET (buf) =
+ GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf),
+ ctx->state.rate);
+ } else {
+ GST_BUFFER_OFFSET (buf) = priv->bytes_out;
+ GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf);
+ }
+ }
+
+ priv->bytes_out += GST_BUFFER_SIZE (buf);
+
+ if (G_UNLIKELY (priv->discont)) {
+ GST_LOG_OBJECT (enc, "marking discont");
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+ priv->discont = FALSE;
+ }
+
+ if (klass->pre_push) {
+ /* last chance for subclass to do some dirty stuff */
+ ret = klass->pre_push (enc, &buf);
+ if (ret != GST_FLOW_OK || !buf) {
+ GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
+ gst_flow_get_name (ret), buf);
+ if (buf)
+ gst_buffer_unref (buf);
+ goto exit;
+ }
+ }
+
+ GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+ ret = gst_pad_push (enc->srcpad, buf);
+ GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
+ } else {
+ /* merely advance samples, most work for that already done above */
+ priv->samples += samples;
+ }
+
+exit:
+ return ret;
+
+ /* ERRORS */
+overflow:
+ {
+ GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
+ ("received more encoded samples %d than provided %d",
+ samples, priv->offset / ctx->state.bpf), (NULL));
+ if (buf)
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
+ }
+}
+
+ /* adapter tracking idea:
+ * - start of adapter corresponds with what has already been encoded
+ * (i.e. really returned by encoder subclass)
+ * - start + offset is what needs to be fed to subclass next */
+static GstFlowReturn
+gst_base_audio_encoder_push_buffers (GstBaseAudioEncoder * enc, gboolean force)
+{
+ GstBaseAudioEncoderClass *klass;
+ GstBaseAudioEncoderPrivate *priv;
+ GstBaseAudioEncoderContext *ctx;
+ gint av, need;
+ GstBuffer *buf;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+
+ g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
+
+ priv = enc->priv;
+ ctx = enc->ctx;
+
+ while (ret == GST_FLOW_OK) {
+
+ buf = NULL;
+ av = gst_adapter_available (priv->adapter);
+
+ g_assert (priv->offset <= av);
+ av -= priv->offset;
+
+ need = ctx->frame_samples > 0 ? ctx->frame_samples * ctx->state.bpf : av;
+ GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d",
+ av, need, force);
+
+ if ((need > av) || !av) {
+ if (G_UNLIKELY (force)) {
+ priv->force = TRUE;
+ need = av;
+ } else {
+ break;
+ }
+ } else {
+ priv->force = FALSE;
+ }
+
+ /* if we have some extra metadata,
+ * provide for integer multiple of frames to allow for better granularity
+ * of processing */
+ if (ctx->frame_samples > 0 && need) {
+ if (ctx->frame_max > 1)
+ need = need * MIN ((av / need), ctx->frame_max);
+ else if (ctx->frame_max == 0)
+ need = need * (av / need);
+ }
+
+ if (need) {
+ buf = gst_buffer_new ();
+ GST_BUFFER_DATA (buf) = (guint8 *)
+ gst_adapter_peek (priv->adapter, priv->offset + need) + priv->offset;
+ GST_BUFFER_SIZE (buf) = need;
+ }
+
+ GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
+ need, priv->offset);
+
+ /* mark this already as consumed,
+ * which it should be when subclass gives us data in exchange for samples */
+ priv->offset += need;
+ priv->samples_in += need / ctx->state.bpf;
+
+ priv->got_data = FALSE;
+ ret = klass->handle_frame (enc, buf);
+
+ if (G_LIKELY (buf))
+ gst_buffer_unref (buf);
+
+ /* no data to feed, no leftover provided, then bail out */
+ if (G_UNLIKELY (!buf && !priv->got_data)) {
+ priv->drained = TRUE;
+ GST_LOG_OBJECT (enc, "no more data drained from subclass");
+ break;
+ }
+ }
+
+ return ret;
+}
+
+static GstFlowReturn
+gst_base_audio_encoder_drain (GstBaseAudioEncoder * enc)
+{
+ if (enc->priv->drained)
+ return GST_FLOW_OK;
+ else
+ return gst_base_audio_encoder_push_buffers (enc, TRUE);
+}
+
+static void
+gst_base_audio_encoder_set_base_gp (GstBaseAudioEncoder * enc)
+{
+ GstClockTime ts;
+
+ if (!enc->granule)
+ return;
+
+ /* use running time for granule */
+ /* incoming data is clipped, so a valid input should yield a valid output */
+ ts = gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME,
+ enc->priv->base_ts);
+ if (GST_CLOCK_TIME_IS_VALID (ts)) {
+ enc->priv->base_gp =
+ GST_CLOCK_TIME_TO_FRAMES (enc->priv->base_ts, enc->ctx->state.rate);
+ GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, enc->priv->base_gp);
+ } else {
+ /* should reasonably have a valid base,
+ * otherwise start at 0 if we did not already start there earlier */
+ if (enc->priv->base_gp < 0) {
+ enc->priv->base_gp = 0;
+ GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
+ enc->priv->base_gp);
+ }
+ }
+}
+
+static GstFlowReturn
+gst_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
+{
+ GstBaseAudioEncoderClass *bclass;
+ GstBaseAudioEncoder *enc;
+ GstBaseAudioEncoderPrivate *priv;
+ GstBaseAudioEncoderContext *ctx;
+ GstFlowReturn ret = GST_FLOW_OK;
+ gboolean discont;
+
+ enc = GST_BASE_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
+ bclass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+
+ priv = enc->priv;
+ ctx = enc->ctx;
+
+ /* should know what is coming by now */
+ if (!ctx->state.bpf)
+ goto not_negotiated;
+
+ GST_LOG_OBJECT (enc,
+ "received buffer of size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
+
+ /* input shoud be whole number of sample frames */
+ if (GST_BUFFER_SIZE (buffer) % ctx->state.bpf)
+ goto wrong_buffer;
+
+#ifndef GST_DISABLE_GST_DEBUG
+ {
+ GstClockTime duration;
+ GstClockTimeDiff diff;
+
+ /* verify buffer duration */
+ duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND,
+ ctx->state.rate * ctx->state.bpf);
+ diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
+ if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
+ (diff > GST_SECOND / ctx->state.rate / 2 ||
+ diff < -GST_SECOND / ctx->state.rate / 2)) {
+ GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
+ GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
+ GST_TIME_ARGS (duration));
+ }
+ }
+#endif
+
+ discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
+ if (G_UNLIKELY (discont)) {
+ GST_LOG_OBJECT (buffer, "marked discont");
+ enc->priv->discont = discont;
+ }
+
+ /* clip to segment */
+ /* NOTE: slightly painful linking -laudio only for this one ... */
+ buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->state.rate,
+ ctx->state.bpf);
+ if (G_UNLIKELY (!buffer)) {
+ GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
+ goto done;
+ }
+
+ GST_LOG_OBJECT (enc,
+ "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
+
+ if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
+ priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
+ GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (priv->base_ts));
+ gst_base_audio_encoder_set_base_gp (enc);
+ }
+
+ /* check for continuity;
+ * checked elsewhere in non-perfect case */
+ if (enc->perfect_ts) {
+ GstClockTimeDiff diff = 0;
+ GstClockTime next_ts = 0;
+
+ if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
+ GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
+ guint64 samples;
+
+ samples = priv->samples +
+ gst_adapter_available (priv->adapter) / ctx->state.bpf;
+ next_ts = priv->base_ts +
+ gst_util_uint64_scale (samples, GST_SECOND, ctx->state.rate);
+ GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
+ " samples past base_ts %" GST_TIME_FORMAT
+ ", expected ts %" GST_TIME_FORMAT, samples,
+ GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
+ diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
+ GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
+ /* if within tolerance,
+ * discard buffer ts and carry on producing perfect stream,
+ * otherwise clip or resync to ts */
+ if (G_UNLIKELY (diff < -enc->tolerance || diff > enc->tolerance)) {
+ GST_DEBUG_OBJECT (enc, "marked discont");
+ discont = TRUE;
+ }
+ }
+
+ /* do some fancy tweaking in hard resync case */
+ if (discont && enc->hard_resync) {
+ if (diff < 0) {
+ guint64 diff_bytes;
+
+ GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
+ GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
+
+ diff_bytes =
+ GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->state.rate) * ctx->state.bpf;
+ if (diff_bytes >= GST_BUFFER_SIZE (buffer)) {
+ gst_buffer_unref (buffer);
+ goto done;
+ }
+ buffer = gst_buffer_make_metadata_writable (buffer);
+ GST_BUFFER_DATA (buffer) += diff_bytes;
+ GST_BUFFER_SIZE (buffer) -= diff_bytes;
+
+ GST_BUFFER_TIMESTAMP (buffer) += diff;
+ /* care even less about duration after this */
+ } else {
+ /* drain stuff prior to resync */
+ gst_base_audio_encoder_drain (enc);
+ }
+ }
+ /* now re-sync ts */
+ priv->base_ts += diff;
+ gst_base_audio_encoder_set_base_gp (enc);
+ priv->discont |= discont;
+ }
+
+ gst_adapter_push (enc->priv->adapter, buffer);
+ /* new stuff, so we can push subclass again */
+ enc->priv->drained = FALSE;
+
+ ret = gst_base_audio_encoder_push_buffers (enc, FALSE);
+
+done:
+ GST_LOG_OBJECT (enc, "chain leaving");
+ return ret;
+
+ /* ERRORS */
+not_negotiated:
+ {
+ GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
+ ("encoder not initialized"));
+ gst_buffer_unref (buffer);
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+wrong_buffer:
+ {
+ GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
+ ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
+ ctx->state.bpf));
+ gst_buffer_unref (buffer);
+ return GST_FLOW_ERROR;
+ }
+}
+
+static gboolean
+gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstBaseAudioEncoder *enc;
+ GstBaseAudioEncoderClass *klass;
+ GstBaseAudioEncoderContext *ctx;
+ GstAudioState *state;
+ gboolean res = TRUE, changed = FALSE;
+
+ enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
+ klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+
+ /* subclass must do something here ... */
+ g_return_val_if_fail (klass->set_format != NULL, FALSE);
+
+ ctx = enc->ctx;
+ state = &ctx->state;
+
+ GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
+
+ if (!gst_caps_is_fixed (caps))
+ goto refuse_caps;
+
+ /* adjust ts tracking to new sample rate */
+ if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && state->rate) {
+ enc->priv->base_ts +=
+ GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, state->rate);
+ enc->priv->samples = 0;
+ }
+
+ if (!gst_base_audio_parse_caps (caps, state, &changed))
+ goto refuse_caps;
+
+ if (changed) {
+ GstClockTime old_min_latency;
+ GstClockTime old_max_latency;
+
+ /* drain any pending old data stuff */
+ gst_base_audio_encoder_drain (enc);
+
+ /* context defaults */
+ enc->ctx->frame_samples = 0;
+ enc->ctx->frame_max = 0;
+ enc->ctx->lookahead = 0;
+
+ /* element might report latency */
+ GST_OBJECT_LOCK (enc);
+ old_min_latency = ctx->min_latency;
+ old_max_latency = ctx->max_latency;
+ GST_OBJECT_UNLOCK (enc);
+
+ if (klass->set_format)
+ res = klass->set_format (enc, state);
+
+ /* notify if new latency */
+ GST_OBJECT_LOCK (enc);
+ if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
+ (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
+ GST_OBJECT_UNLOCK (enc);
+ /* post latency message on the bus */
+ gst_element_post_message (GST_ELEMENT (enc),
+ gst_message_new_latency (GST_OBJECT (enc)));
+ GST_OBJECT_LOCK (enc);
+ }
+ GST_OBJECT_UNLOCK (enc);
+ } else {
+ GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
+ }
+
+ return res;
+
+ /* ERRORS */
+refuse_caps:
+ {
+ GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
+ return res;
+ }
+}
+
+
+/**
+ * gst_base_audio_encoder_proxy_getcaps:
+ * @enc: a #GstBaseAudioEncoder
+ * @caps: initial
+ *
+ * Returns caps that express @caps (or sink template caps if @caps == NULL)
+ * restricted to channel/rate combinations supported by downstream elements
+ * (e.g. muxers).
+ *
+ * Returns: a #GstCaps owned by caller
+ */
+GstCaps *
+gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc, GstCaps * caps)
+{
+ const GstCaps *templ_caps;
+ GstCaps *allowed = NULL;
+ GstCaps *fcaps, *filter_caps;
+ gint i, j;
+
+ /* we want to be able to communicate to upstream elements like audioconvert
+ * and audioresample any rate/channel restrictions downstream (e.g. muxer
+ * only accepting certain sample rates) */
+ templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad);
+ allowed = gst_pad_get_allowed_caps (enc->srcpad);
+ if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
+ fcaps = gst_caps_copy (templ_caps);
+ goto done;
+ }
+
+ GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
+ GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
+
+ filter_caps = gst_caps_new_empty ();
+
+ for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
+ GQuark q_name;
+
+ q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
+
+ /* pick rate + channel fields from allowed caps */
+ for (j = 0; j < gst_caps_get_size (allowed); j++) {
+ const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
+ const GValue *val;
+ GstStructure *s;
+
+ s = gst_structure_id_empty_new (q_name);
+ if ((val = gst_structure_get_value (allowed_s, "rate")))
+ gst_structure_set_value (s, "rate", val);
+ if ((val = gst_structure_get_value (allowed_s, "channels")))
+ gst_structure_set_value (s, "channels", val);
+
+ gst_caps_merge_structure (filter_caps, s);
+ }
+ }
+
+ fcaps = gst_caps_intersect (filter_caps, templ_caps);
+ gst_caps_unref (filter_caps);
+
+done:
+ gst_caps_replace (&allowed, NULL);
+
+ GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
+
+ return fcaps;
+}
+
+static GstCaps *
+gst_base_audio_encoder_sink_getcaps (GstPad * pad)
+{
+ GstBaseAudioEncoder *enc;
+ GstBaseAudioEncoderClass *klass;
+ GstCaps *caps;
+
+ enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+ g_assert (pad == enc->sinkpad);
+
+ if (klass->getcaps)
+ caps = klass->getcaps (enc);
+ else
+ caps = gst_base_audio_encoder_proxy_getcaps (enc, NULL);
+ gst_object_unref (enc);
+
+ GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
+
+ return caps;
+}
+
+static gboolean
+gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
+ GstEvent * event)
+{
+ GstBaseAudioEncoderClass *klass;
+ gboolean handled = FALSE;
+
+ klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_NEWSEGMENT:
+ {
+ GstFormat format;
+ gdouble rate, arate;
+ gint64 start, stop, time;
+ gboolean update;
+
+ gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+ &start, &stop, &time);
+
+ if (format == GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
+ " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
+ ", rate %g, applied_rate %g",
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
+ rate, arate);
+ } else {
+ GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
+ " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
+ ", rate %g, applied_rate %g", start, stop, time, rate, arate);
+ GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
+ break;
+ }
+
+ /* finish current segment */
+ gst_base_audio_encoder_drain (enc);
+ /* reset partially for new segment */
+ gst_base_audio_encoder_reset (enc, FALSE);
+ /* and follow along with segment */
+ gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
+ format, start, stop, time);
+ break;
+ }
+
+ case GST_EVENT_FLUSH_START:
+ break;
+
+ case GST_EVENT_FLUSH_STOP:
+ /* discard any pending stuff */
+ /* TODO route through drain ?? */
+ if (!enc->priv->drained && klass->flush)
+ klass->flush (enc);
+ /* and get (re)set for the sequel */
+ gst_base_audio_encoder_reset (enc, FALSE);
+ break;
+
+ case GST_EVENT_EOS:
+ gst_base_audio_encoder_drain (enc);
+ break;
+
+ default:
+ break;
+ }
+
+ return handled;
+}
+
+static gboolean
+gst_base_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstBaseAudioEncoder *enc;
+ GstBaseAudioEncoderClass *klass;
+ gboolean handled = FALSE;
+ gboolean ret = TRUE;
+
+ enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+
+ GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
+ GST_EVENT_TYPE_NAME (event));
+
+ if (klass->event)
+ handled = klass->event (enc, event);
+
+ if (!handled)
+ handled = gst_base_audio_encoder_sink_eventfunc (enc, event);
+
+ if (!handled)
+ ret = gst_pad_event_default (pad, event);
+
+ GST_DEBUG_OBJECT (enc, "event handled");
+
+ gst_object_unref (enc);
+ return ret;
+}
+
+static gboolean
+gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
+{
+ gboolean res = TRUE;
+ GstBaseAudioEncoder *enc;
+
+ enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_FORMATS:
+ {
+ gst_query_set_formats (query, 3,
+ GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
+ res = TRUE;
+ break;
+ }
+ case GST_QUERY_CONVERT:
+ {
+ GstFormat src_fmt, dest_fmt;
+ gint64 src_val, dest_val;
+
+ gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+ if (!(res = gst_base_audio_raw_audio_convert (&enc->ctx->state,
+ src_fmt, src_val, &dest_fmt, &dest_val)))
+ goto error;
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+error:
+ gst_object_unref (enc);
+ return res;
+}
+
+static const GstQueryType *
+gst_base_audio_encoder_get_query_types (GstPad * pad)
+{
+ static const GstQueryType gst_base_audio_encoder_src_query_types[] = {
+ GST_QUERY_POSITION,
+ GST_QUERY_DURATION,
+ GST_QUERY_CONVERT,
+ GST_QUERY_LATENCY,
+ 0
+ };
+
+ return gst_base_audio_encoder_src_query_types;
+}
+
+/* FIXME ? are any of these queries (other than latency) an encoder's business
+ * also, the conversion stuff might seem to make sense, but seems to not mind
+ * segment stuff etc at all
+ * Supposedly that's backward compatibility ... */
+static gboolean
+gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
+{
+ GstBaseAudioEncoder *enc;
+ GstPad *peerpad;
+ gboolean res = FALSE;
+
+ enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
+ peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
+
+ GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_POSITION:
+ {
+ GstFormat fmt, req_fmt;
+ gint64 pos, val;
+
+ if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
+ GST_LOG_OBJECT (enc, "returning peer response");
+ break;
+ }
+
+ if (!peerpad) {
+ GST_LOG_OBJECT (enc, "no peer");
+ break;
+ }
+
+ gst_query_parse_position (query, &req_fmt, NULL);
+ fmt = GST_FORMAT_TIME;
+ if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
+ break;
+
+ if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) {
+ gst_query_set_position (query, req_fmt, val);
+ }
+ break;
+ }
+ case GST_QUERY_DURATION:
+ {
+ GstFormat fmt, req_fmt;
+ gint64 dur, val;
+
+ if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
+ GST_LOG_OBJECT (enc, "returning peer response");
+ break;
+ }
+
+ if (!peerpad) {
+ GST_LOG_OBJECT (enc, "no peer");
+ break;
+ }
+
+ gst_query_parse_duration (query, &req_fmt, NULL);
+ fmt = GST_FORMAT_TIME;
+ if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
+ break;
+
+ if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) {
+ gst_query_set_duration (query, req_fmt, val);
+ }
+ break;
+ }
+ case GST_QUERY_FORMATS:
+ {
+ gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
+ res = TRUE;
+ break;
+ }
+ case GST_QUERY_CONVERT:
+ {
+ GstFormat src_fmt, dest_fmt;
+ gint64 src_val, dest_val;
+
+ gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
+ if (!(res = gst_base_audio_encoded_audio_convert (&enc->ctx->state,
+ enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
+ &dest_fmt, &dest_val)))
+ break;
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ break;
+ }
+ case GST_QUERY_LATENCY:
+ {
+ if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
+ gboolean live;
+ GstClockTime min_latency, max_latency;
+
+ gst_query_parse_latency (query, &live, &min_latency, &max_latency);
+ GST_DEBUG_OBJECT (enc, "Peer latency: live %d, min %"
+ GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
+ GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
+
+ GST_OBJECT_LOCK (enc);
+ /* add our latency */
+ if (min_latency != -1)
+ min_latency += enc->ctx->min_latency;
+ if (max_latency != -1)
+ max_latency += enc->ctx->max_latency;
+ GST_OBJECT_UNLOCK (enc);
+
+ gst_query_set_latency (query, live, min_latency, max_latency);
+ }
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+ gst_object_unref (peerpad);
+ return res;
+}
+
+static void
+gst_base_audio_encoder_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstBaseAudioEncoder *enc;
+
+ enc = GST_BASE_AUDIO_ENCODER (object);
+
+ switch (prop_id) {
+ case PROP_PERFECT_TS:
+ if (enc->granule && !g_value_get_boolean (value))
+ GST_WARNING_OBJECT (enc, "perfect-ts can not be set FALSE");
+ else
+ enc->perfect_ts = g_value_get_boolean (value);
+ break;
+ case PROP_HARD_RESYNC:
+ enc->hard_resync = g_value_get_boolean (value);
+ break;
+ case PROP_TOLERANCE:
+ enc->tolerance = g_value_get_int64 (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_base_audio_encoder_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstBaseAudioEncoder *enc;
+
+ enc = GST_BASE_AUDIO_ENCODER (object);
+
+ switch (prop_id) {
+ case PROP_PERFECT_TS:
+ g_value_set_boolean (value, enc->perfect_ts);
+ break;
+ case PROP_GRANULE:
+ g_value_set_boolean (value, enc->granule);
+ break;
+ case PROP_HARD_RESYNC:
+ g_value_set_boolean (value, enc->hard_resync);
+ break;
+ case PROP_TOLERANCE:
+ g_value_set_int64 (value, enc->tolerance);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static gboolean
+gst_base_audio_encoder_activate (GstBaseAudioEncoder * enc, gboolean active)
+{
+ GstBaseAudioEncoderClass *klass;
+ gboolean result = FALSE;
+
+ klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+
+ g_return_val_if_fail (!enc->granule || enc->perfect_ts, FALSE);
+
+ GST_DEBUG_OBJECT (enc, "activate %d", active);
+
+ if (active) {
+ if (!enc->priv->active && klass->start)
+ result = klass->start (enc);
+ } else {
+ /* We must make sure streaming has finished before resetting things
+ * and calling the ::stop vfunc */
+ GST_PAD_STREAM_LOCK (enc->sinkpad);
+ GST_PAD_STREAM_UNLOCK (enc->sinkpad);
+
+ if (enc->priv->active && klass->stop)
+ result = klass->stop (enc);
+
+ /* clean up */
+ gst_base_audio_encoder_reset (enc, TRUE);
+ }
+ GST_DEBUG_OBJECT (enc, "activate return: %d", result);
+ return result;
+}
+
+
+static gboolean
+gst_base_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
+{
+ gboolean result = TRUE;
+ GstBaseAudioEncoder *enc;
+
+ enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
+
+ GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
+
+ result = gst_base_audio_encoder_activate (enc, active);
+
+ if (result)
+ enc->priv->active = active;
+
+ GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
+
+ gst_object_unref (enc);
+ return result;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_BASE_AUDIO_ENCODER_H__
+#define __GST_BASE_AUDIO_ENCODER_H__
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "GstBaseAudioEncoder is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/gst.h>
+#include <gst/audio/gstbaseaudioutils.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_BASE_AUDIO_ENCODER (gst_base_audio_encoder_get_type())
+#define GST_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoder))
+#define GST_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
+#define GST_BASE_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
+#define GST_IS_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_ENCODER))
+#define GST_IS_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_ENCODER))
+#define GST_BASE_AUDIO_ENCODER_CAST(obj) ((GstBaseAudioEncoder *)(obj))
+
+/**
+ * GST_BASE_AUDIO_ENCODER_SINK_NAME:
+ *
+ * the name of the templates for the sink pad
+ */
+#define GST_BASE_AUDIO_ENCODER_SINK_NAME "sink"
+/**
+ * GST_BASE_AUDIO_ENCODER_SRC_NAME:
+ *
+ * the name of the templates for the source pad
+ */
+#define GST_BASE_AUDIO_ENCODER_SRC_NAME "src"
+
+/**
+ * GST_BASE_AUDIO_ENCODER_SRC_PAD:
+ * @obj: base parse instance
+ *
+ * Gives the pointer to the source #GstPad object of the element.
+ *
+ * Since: 0.10.x
+ */
+#define GST_BASE_AUDIO_ENCODER_SRC_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->srcpad)
+
+/**
+ * GST_BASE_AUDIO_ENCODER_SINK_PAD:
+ * @obj: base parse instance
+ *
+ * Gives the pointer to the sink #GstPad object of the element.
+ *
+ * Since: 0.10.x
+ */
+#define GST_BASE_AUDIO_ENCODER_SINK_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->sinkpad)
+
+/**
+ * GST_BASE_AUDIO_ENCODER_SEGMENT:
+ * @obj: base parse instance
+ *
+ * Gives the segment of the element.
+ *
+ * Since: 0.10.x
+ */
+#define GST_BASE_AUDIO_ENCODER_SEGMENT(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->segment)
+
+
+typedef struct _GstBaseAudioEncoder GstBaseAudioEncoder;
+typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass;
+
+typedef struct _GstBaseAudioEncoderPrivate GstBaseAudioEncoderPrivate;
+typedef struct _GstBaseAudioEncoderContext GstBaseAudioEncoderContext;
+
+/**
+ * GstBaseAudioEncoderContext:
+ * @state: a #GstAudioState describing input audio format
+ * @frame_samples: number of samples (per channel) subclass needs to be handed,
+ * or will be handed all available if 0.
+ * @frame_max: max number of frames of size @frame_bytes accepted at once
+ * (assumed minimally 1)
+ * @min_latency: min latency of element
+ * @max_latency: max latency of element
+ * @lookahead: encoder lookahead (in units of input rate samples)
+ *
+ * Transparent #GstBaseAudioEncoderContext data structure.
+ */
+struct _GstBaseAudioEncoderContext {
+ /* input */
+ GstAudioState state;
+
+ /* output */
+ gint frame_samples;
+ gint frame_max;
+ gint lookahead;
+ /* MT-protected (with LOCK) */
+ GstClockTime min_latency;
+ GstClockTime max_latency;
+};
+
+/**
+ * GstBaseAudioEncoder:
+ * @element: the parent element.
+ *
+ * The opaque #GstBaseAudioEncoder data structure.
+ */
+struct _GstBaseAudioEncoder {
+ GstElement element;
+
+ /*< protected >*/
+ /* source and sink pads */
+ GstPad *sinkpad;
+ GstPad *srcpad;
+
+ /* MT-protected (with STREAM_LOCK) */
+ GstSegment segment;
+ GstBaseAudioEncoderContext *ctx;
+
+ /* properties */
+ gint64 tolerance;
+ gboolean perfect_ts;
+ gboolean hard_resync;
+ gboolean granule;
+
+ /*< private >*/
+ GstBaseAudioEncoderPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+/**
+ * GstBaseAudioEncoderClass:
+ * @start: Optional.
+ * Called when the element starts processing.
+ * Allows opening external resources.
+ * @stop: Optional.
+ * Called when the element stops processing.
+ * Allows closing external resources.
+ * @set_format: Notifies subclass of incoming data format.
+ * GstBaseAudioEncoderContext fields have already been
+ * set according to provided caps.
+ * @handle_frame: Provides input samples (or NULL to clear any remaining data)
+ * according to directions as provided by subclass in the
+ * #GstBaseAudioEncoderContext. Input data ref management
+ * is performed by base class, subclass should not care or
+ * intervene.
+ * @flush: Optional.
+ * Instructs subclass to clear any codec caches and discard
+ * any pending samples and not yet returned encoded data.
+ * @event: Optional.
+ * Event handler on the sink pad. This function should return
+ * TRUE if the event was handled and should be discarded
+ * (i.e. not unref'ed).
+ * @pre_push: Optional.
+ * Called just prior to pushing (encoded data) buffer downstream.
+ * Subclass has full discretionary access to buffer,
+ * and a not OK flow return will abort downstream pushing.
+ * @getcaps: Optional.
+ * Allows for a custom sink getcaps implementation (e.g.
+ * for multichannel input specification). If not implemented,
+ * default returns gst_base_audio_encoder_proxy_getcaps
+ * applied to sink template caps.
+ *
+ * Subclasses can override any of the available virtual methods or not, as
+ * needed. At minimum @set_format and @handle_frame needs to be overridden.
+ */
+struct _GstBaseAudioEncoderClass {
+ GstElementClass parent_class;
+
+ /*< public >*/
+ /* virtual methods for subclasses */
+
+ gboolean (*start) (GstBaseAudioEncoder *enc);
+
+ gboolean (*stop) (GstBaseAudioEncoder *enc);
+
+ gboolean (*set_format) (GstBaseAudioEncoder *enc,
+ GstAudioState *state);
+
+ GstFlowReturn (*handle_frame) (GstBaseAudioEncoder *enc,
+ GstBuffer *buffer);
+
+ void (*flush) (GstBaseAudioEncoder *enc);
+
+ GstFlowReturn (*pre_push) (GstBaseAudioEncoder *enc,
+ GstBuffer **buffer);
+
+ gboolean (*event) (GstBaseAudioEncoder *enc,
+ GstEvent *event);
+
+ GstCaps * (*getcaps) (GstBaseAudioEncoder *enc);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_base_audio_encoder_get_type (void);
+
+GstFlowReturn gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc,
+ GstBuffer *buffer, gint samples);
+
+GstCaps * gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc,
+ GstCaps * caps);
+
+G_END_DECLS
+
+#endif /* __GST_BASE_AUDIO_ENCODER_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include "gstbaseaudioutils.h"
+
+#include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
+
+#define CHECK_VALUE(var, val) \
+G_STMT_START { \
+ if (!res) \
+ goto fail; \
+ if (var != val) \
+ changed = TRUE; \
+ var = val; \
+} G_STMT_END
+
+/**
+ * gst_base_audio_parse_caps:
+ * @caps: a #GstCaps
+ * @state: a #GstAudioState
+ * @changed: whether @caps introduced a change in current @state
+ *
+ * Parses audio format as represented by @caps into a more concise form
+ * as represented by @state, while checking if for changes to currently
+ * defined audio format.
+ *
+ * Returns: TRUE if parsing succeeded, otherwise FALSE
+ */
+gboolean
+gst_base_audio_parse_caps (GstCaps * caps, GstAudioState * state,
+ gboolean * _changed)
+{
+ gboolean res = TRUE, changed = FALSE;
+ GstStructure *s;
+ gboolean vb;
+ gint vi;
+
+ g_return_val_if_fail (caps != NULL, FALSE);
+ g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
+
+ s = gst_caps_get_structure (caps, 0);
+ if (gst_structure_has_name (s, "audio/x-raw-int"))
+ state->is_int = TRUE;
+ else if (gst_structure_has_name (s, "audio/x-raw-float"))
+ state->is_int = FALSE;
+ else
+ goto fail;
+
+ res = gst_structure_get_int (s, "rate", &vi);
+ CHECK_VALUE (state->rate, vi);
+ res &= gst_structure_get_int (s, "channels", &vi);
+ CHECK_VALUE (state->channels, vi);
+ res &= gst_structure_get_int (s, "width", &vi);
+ CHECK_VALUE (state->width, vi);
+ res &= (!state->is_int || gst_structure_get_int (s, "depth", &vi));
+ CHECK_VALUE (state->depth, vi);
+ res &= gst_structure_get_int (s, "endianness", &vi);
+ CHECK_VALUE (state->endian, vi);
+ res &= (!state->is_int || gst_structure_get_boolean (s, "signed", &vb));
+ CHECK_VALUE (state->sign, vb);
+
+ state->bpf = (state->width / 8) * state->channels;
+ GST_LOG ("bpf: %d", state->bpf);
+ if (!state->bpf)
+ goto fail;
+
+ g_free (state->channel_pos);
+ state->channel_pos = gst_audio_get_channel_positions (s);
+
+ if (_changed)
+ *_changed = changed;
+
+ return res;
+
+ /* ERRORS */
+fail:
+ {
+ /* there should not be caps out there that fail parsing ... */
+ GST_WARNING ("failed to parse caps %" GST_PTR_FORMAT, caps);
+ return res;
+ }
+}
+
+/**
+ * gst_base_audio_add_streamheader:
+ * @caps: a #GstCaps
+ * @buf: header buffers
+ *
+ * Adds given buffers to an array of buffers set as streamheader field
+ * on the given @caps. List of buffer arguments must be NULL-terminated.
+ *
+ * Returns: input caps with a streamheader field added, or NULL if some error
+ */
+GstCaps *
+gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...)
+{
+ GstStructure *structure = NULL;
+ va_list va;
+ GValue array = { 0 };
+ GValue value = { 0 };
+
+ g_return_val_if_fail (caps != NULL, NULL);
+ g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
+
+ caps = gst_caps_make_writable (caps);
+ structure = gst_caps_get_structure (caps, 0);
+
+ g_value_init (&array, GST_TYPE_ARRAY);
+
+ va_start (va, buf);
+ /* put buffers in a fixed list */
+ while (buf) {
+ g_assert (gst_buffer_is_metadata_writable (buf));
+
+ /* mark buffer */
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+
+ g_value_init (&value, GST_TYPE_BUFFER);
+ buf = gst_buffer_copy (buf);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ gst_value_set_buffer (&value, buf);
+ gst_buffer_unref (buf);
+ gst_value_array_append_value (&array, &value);
+ g_value_unset (&value);
+
+ buf = va_arg (va, GstBuffer *);
+ }
+
+ gst_structure_set_value (structure, "streamheader", &array);
+ g_value_unset (&array);
+
+ return caps;
+}
+
+/**
+ * gst_base_audio_encoded_audio_convert:
+ * @fmt: audio format of the encoded audio
+ * @bytes: number of encoded bytes
+ * @samples: number of encoded samples
+ * @src_format: source format
+ * @src_value: source value
+ * @dest_format: destination format
+ * @dest_value: destination format
+ *
+ * Helper function to convert @src_value in @src_format to @dest_value in
+ * @dest_format for encoded audio data. Conversion is possible between
+ * BYTE and TIME format by using estimated bitrate based on
+ * @samples and @bytes (and @fmt).
+ */
+gboolean
+gst_base_audio_encoded_audio_convert (GstAudioState * fmt,
+ gint64 bytes, gint64 samples, GstFormat src_format,
+ gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
+{
+ gboolean res = FALSE;
+
+ g_return_val_if_fail (dest_format != NULL, FALSE);
+ g_return_val_if_fail (dest_value != NULL, FALSE);
+
+ if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
+ src_value == -1)) {
+ if (dest_value)
+ *dest_value = src_value;
+ return TRUE;
+ }
+
+ if (samples == 0 || bytes == 0 || fmt->rate == 0) {
+ GST_DEBUG ("not enough metadata yet to convert");
+ goto exit;
+ }
+
+ bytes *= fmt->rate;
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale (src_value,
+ GST_SECOND * samples, bytes);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ *dest_value = gst_util_uint64_scale (src_value, bytes,
+ samples * GST_SECOND);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ }
+
+exit:
+ return res;
+}
+
+/**
+ * gst_base_audio_raw_audio_convert:
+ * @fmt: audio format of the encoded audio
+ * @src_format: source format
+ * @src_value: source value
+ * @dest_format: destination format
+ * @dest_value: destination format
+ *
+ * Helper function to convert @src_value in @src_format to @dest_value in
+ * @dest_format for encoded audio data. Conversion is possible between
+ * BYTE, DEFAULT and TIME format based on audio characteristics provided
+ * by @fmt.
+ */
+gboolean
+gst_base_audio_raw_audio_convert (GstAudioState * fmt, GstFormat src_format,
+ gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
+{
+ gboolean res = FALSE;
+ guint scale = 1;
+ gint bytes_per_sample, rate, byterate;
+
+ g_return_val_if_fail (dest_format != NULL, FALSE);
+ g_return_val_if_fail (dest_value != NULL, FALSE);
+
+ if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
+ src_value == -1)) {
+ if (dest_value)
+ *dest_value = src_value;
+ return TRUE;
+ }
+
+ bytes_per_sample = fmt->bpf;
+ rate = fmt->rate;
+ byterate = bytes_per_sample * rate;
+
+ if (G_UNLIKELY (bytes_per_sample == 0 || rate == 0)) {
+ GST_DEBUG ("not enough metadata yet to convert");
+ goto exit;
+ }
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+ case GST_FORMAT_DEFAULT:
+ *dest_value = src_value / bytes_per_sample;
+ res = TRUE;
+ break;
+ case GST_FORMAT_TIME:
+ *dest_value =
+ gst_util_uint64_scale_int (src_value, GST_SECOND, byterate);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_DEFAULT:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ *dest_value = src_value * bytes_per_sample;
+ res = TRUE;
+ break;
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, rate);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ scale = bytes_per_sample;
+ /* fallthrough */
+ case GST_FORMAT_DEFAULT:
+ *dest_value = gst_util_uint64_scale_int (src_value,
+ scale * rate, GST_SECOND);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ }
+
+exit:
+ return res;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef _GST_BASE_AUDIO_UTILS_H_
+#define _GST_BASE_AUDIO_UTILS_H_
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "Base audio utils provide unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstAudioState:
+ * @is_int: whether sample data is int or float
+ * @rate: rate of sample data
+ * @channels: number of channels in sample data
+ * @width: width (in bits) of sample data
+ * @depth: used bits in sample data (if integer)
+ * @sign: sign of sample data (if integer)
+ * @endian: endianness of sample data
+ * @bpf: bytes per audio frame
+ */
+typedef struct _GstAudioState {
+ gboolean is_int;
+ gint rate;
+ gint channels;
+ gint width;
+ gint depth;
+ gboolean sign;
+ gint endian;
+ GstAudioChannelPosition *channel_pos;
+
+ gint bpf;
+} GstAudioState;
+
+gboolean gst_base_audio_parse_caps (GstCaps * caps,
+ GstAudioState * state, gboolean * changed);
+
+GstCaps *gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...);
+
+gboolean gst_base_audio_encoded_audio_convert (GstAudioState * fmt,
+ gint64 bytes, gint64 samples, GstFormat src_format,
+ gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
+
+gboolean gst_base_audio_raw_audio_convert (GstAudioState * fmt, GstFormat src_format,
+ gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
+
+G_END_DECLS
+
+#endif
+
projects / platform / upstream / gstreamer.git / commitdiff