/* 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
/**
* SECTION:gstbaseaudiodecoder
- * @short_description: Base class for codec elements
- * @see_also: #GstBaseTransform, #GstBaseSource, #GstBaseSink
+ * @short_description: Base class for audio decoders
+ * @see_also: #GstBaseTransform
*
- * #GstBaseAudioDecoder is the base class for codec elements ion GStreamer. It is
- * a layer on top of #GstElement that provides simplified interface to plugin
- * writers, hangling many details for you. Its way of operation is explained
- * below.
+ * This base class is for audio decoders turning encoded data into
+ * raw audio samples.
*
- * Subclasses are responsible for specifying the codec's source pad caps. For
- * that purpose they should provide an implementation of ::negotiate_src_caps.
- * If the subclass provides an implementation of this method, it will be
- * invoked by #GstBaseAudioDecoder on its sink_setcaps function. Otherwise, if
- * the subclass does not provide an implementation of this method, the subclass
- * will be responsible for calling gst_base_audio_decoder_set_src_caps() to
- * complete the caps negotiation before any buffers are pushed out.
+ * GstBaseAudioDecoder and subclass should cooperate as follows.
+ * <orderedlist>
+ * <listitem>
+ * <itemizedlist><title>Configuration</title>
+ * <listitem><para>
+ * Initially, GstBaseAudioEncoder 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>
+ * GstBaseAudioEncoder 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>
+ * GstBaseAudioEncoder 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 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>
*
- * Each buffer received on the codec's sink pad is pushed to its input
- * adapter. When there is enough data present in the input adapter
- * (configured in the #GstBaseAudioDecoder:input-buffer-size
- * property), the method ::process_data is called on the subclass. Subclasses
- * must provide an implementation of this method, which would read from the
- * input adapter, encode or decode the data, and push it to the output adapter.
- * If #GstBaseAudioDecoder:input-buffer-size is set to 0 ::process_data will be
- * invoked as soon as there is any data on the input adapter.
+ * 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.
*
- * Similarly, when there is enough data present on the output adapter,
- * (configured in the #GstBaseAudioDecoder:output-buffer-size property),
- * buffers will be pushed out through the codec's source pad. If
- * #GstBaseAudioDecoder:output-buffer-size is set to 0 a buffer will be pushed
- * out as soon as there is any data present on the output adapter. Notice
- * that if no implementation of ::negotiate_src_caps has been provided by the
- * subclass, it must call gst_base_audio_decoder_set_src_caps() to complete
- * the caps negotiation process or otherwise attempting to push buffers
- * through the codec's source pad will fail.
+ * 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.
*
- * It is possible for subclasses to take control on how and when buffers
- * are pushed out by overriding the ::push_data method. If subclasses
- * provide an implementation of this method #GstBaseAudioDecoder will
- * not push buffers out by itself, instead, whenever there* is data present
- * in the output adapter, it will invoke ::push_data on subclass, which
- * will implement there any logic necessary for pushing buffers out when
- * appropriate. In this mode of operation, the property
- * ::output_buffer_size is ignored in #GstBaseAudioDecoder. In any case,
- * buffers should be pushed using gst_base_audio_decoder_push_buffer().
+ * 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).
*
- * #GstBaseAudioDecoder checks for discontinuities and handles them
- * appropriately when pushing buffers out (setting the discontinuous
- * flag on the output buffers when necessary). Subclasses can check if
- * the data present on the adapters represents a discontinuity by checking
- * the discont field of #GstBaseAudioDecoder. Also, subclasses can provide
- * an implementation for the ::handle_discont method, which will be invoked
- * whenever a discontinuity is detected on the source stream.
+ * 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.
*
- * Because data is not processed immediately and is stored in adapters,
- * depending on how the actual codec operates it may be possible to
- * receive an end-of-stream event before all the data in the adapters
- * has been processed and pushed out. If this can happen, the subclass
- * must provide implementation of the ::flush_input method, which should
- * then read the data present int the input adapter, process it and
- * store the result in the output adapter. The subclass may also want
- * provide an implementation for the ::flush_output method, which would
- * take care of reading the data from the output adapter and push it
- * out through the codec's source pad. If no implementation is provided
- * for the ::flush_out method, #GstBaseAudioDecoder will create a single
- * buffer with all the data present in the output adapter and push it
- * out. If a subclass needs to force a flush on the adapters for some
- * reason, it should call gst_base_audio_decoder_flush(), which will then
- * invoke ::flush_input and/or ::flush_output appropriately.
+ * On the other hand, it should be noted that baseclass does not handle seeking,
+ * nor are subclasses expected to do so, as this should rather be left to
+ * upstream demuxer, parser or alike.
*
- * Subclasses may provide an implementation for the ::start, ::stop
- * and ::reset methods when needed. This methods will be called
- * from #GstBaseAudioDecoder when needed (on state changes,
- * discontinuities, etc), so they must never invoke the
- * implementation on the parent class. When a subclass needs to
- * start, stop or reset the codec itself, it should use the public
- * functions gst_base_audio_decoder_{start,stop,reset}(), which call
- * the corresponding methods on the parent class, which will then
- * call the functions provided by the subclass (if any).
- *
- * #GstBaseAudioDecoder also provides an sink event handler.
- * Subclasses that want to be notified on these events, can provide
- * an implementation of the ::event function, which will be called after
- * #GstBaseAudioDecoder has processed the event itself.
+ * 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 "gstbaseaudiodecoder.h"
#include <gst/audio/audio.h>
-#include <string.h>
+#include <gst/base/gstadapter.h>
-/*
- * FIXME: maybe we need more work with the segments (see ac3 decoder)
- */
+#include <string.h>
GST_DEBUG_CATEGORY (baseaudiodecoder_debug);
#define GST_CAT_DEFAULT baseaudiodecoder_debug
-/* ----- Signals and properties ----- */
+#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_INPUT_BUFFER_SIZE,
- PROP_OUTPUT_BUFFER_SIZE
+ 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;
+
+ /* 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;
};
-/* ----- Function prototypes ----- */
static void gst_base_audio_decoder_finalize (GObject * object);
static void gst_base_audio_decoder_set_property (GObject * object,
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 void gst_base_audio_decoder_handle_discont (GstBaseAudioDecoder * codec,
- 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);
-/* ----- GObject setup ----- */
GST_BOILERPLATE (GstBaseAudioDecoder, gst_base_audio_decoder, GstElement,
GST_TYPE_ELEMENT);
static void
gst_base_audio_decoder_base_init (gpointer g_class)
{
- GST_DEBUG_CATEGORY_INIT (baseaudiodecoder_debug, "baseaudiodecoder", 0,
- "Base Audio Codec Classes");
}
static void
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;
- klass->start = NULL;
- klass->stop = NULL;
- klass->reset = NULL;
- klass->event = NULL;
- klass->handle_discont = NULL;
- klass->flush_input = NULL;
- klass->flush_output = NULL;
- klass->process_data = NULL;
- klass->push_data = NULL;
- klass->negotiate_src_caps = NULL;
-
/* Properties */
- g_object_class_install_property (gobject_class, PROP_INPUT_BUFFER_SIZE,
- g_param_spec_uint ("input-buffer-size", "Input buffer size",
- "Size of the input buffers in bytes (0 for not setting a "
- "particular size)", 0, G_MAXUINT, 0, G_PARAM_READWRITE));
-
- g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_SIZE,
- g_param_spec_uint ("output-buffer-size", "Output buffer size",
- "Size of the output buffers in bytes (0 for not setting a "
- "particular size)", 0, G_MAXUINT, 0, G_PARAM_READWRITE));
+ 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 * codec,
+gst_base_audio_decoder_init (GstBaseAudioDecoder * dec,
GstBaseAudioDecoderClass * klass)
{
GstPadTemplate *pad_template;
- GST_DEBUG ("gst_base_audio_decoder_init");
+ 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);
- codec->sinkpad = gst_pad_new_from_template (pad_template, "sink");
- gst_pad_set_event_function (codec->sinkpad,
- gst_base_audio_decoder_sink_event);
- gst_pad_set_setcaps_function (codec->sinkpad,
- gst_base_audio_decoder_sink_setcaps);
- gst_pad_set_chain_function (codec->sinkpad, gst_base_audio_decoder_chain);
- gst_element_add_pad (GST_ELEMENT (codec), codec->sinkpad);
+ 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);
- codec->srcpad = gst_pad_new_from_template (pad_template, "src");
- gst_pad_use_fixed_caps (codec->srcpad);
- gst_element_add_pad (GST_ELEMENT (codec), codec->srcpad);
-
- /* Setup adapters */
- codec->input_adapter = gst_adapter_new ();
- codec->output_adapter = gst_adapter_new ();
- codec->input_buffer_size = 0;
- codec->output_buffer_size = 0;
-
- /* Setup state */
- memset (&codec->state, 0, sizeof (GstAudioState));
- gst_segment_init (&codec->state.segment, GST_FORMAT_TIME);
-
- codec->started = FALSE;
- codec->bytes_in = 0;
- codec->bytes_out = 0;
- codec->discont = TRUE;
- codec->caps_set = FALSE;
- codec->first_ts = -1;
- codec->last_ts = -1;
+ 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_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
+gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec, gboolean full)
{
- GstBaseAudioDecoder *codec;
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_reset");
- codec = GST_BASE_AUDIO_DECODER (object);
+ GST_OBJECT_LOCK (dec);
- switch (prop_id) {
- case PROP_INPUT_BUFFER_SIZE:
- g_value_set_uint (value, codec->input_buffer_size);
- break;
- case PROP_OUTPUT_BUFFER_SIZE:
- g_value_set_uint (value, codec->output_buffer_size);
- 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 *codec;
+ if (full) {
+ dec->priv->active = FALSE;
+ dec->priv->bytes_in = 0;
+ dec->priv->samples_out = 0;
+ dec->priv->agg = -1;
+ gst_base_audio_decoder_clear_queues (dec);
- codec = GST_BASE_AUDIO_DECODER (object);
+ g_free (dec->ctx->state.channel_pos);
+ memset (dec->ctx, 0, sizeof (dec->ctx));
- switch (prop_id) {
- case PROP_INPUT_BUFFER_SIZE:
- codec->input_buffer_size = g_value_get_uint (value);
- break;
- case PROP_OUTPUT_BUFFER_SIZE:
- codec->output_buffer_size = g_value_get_uint (value);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
+ 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 *codec;
+ GstBaseAudioDecoder *dec;
g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (object));
- codec = GST_BASE_AUDIO_DECODER (object);
+ dec = GST_BASE_AUDIO_DECODER (object);
- if (codec->input_adapter) {
- g_object_unref (codec->input_adapter);
+ if (dec->priv->adapter) {
+ g_object_unref (dec->priv->adapter);
}
- if (codec->output_adapter) {
- g_object_unref (codec->output_adapter);
- }
- if (codec->codec_data) {
- gst_buffer_unref (codec->codec_data);
+ if (dec->priv->adapter_out) {
+ g_object_unref (dec->priv->adapter_out);
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
-/* ----- Private element implementation ----- */
-
-static void
-gst_base_audio_decoder_read_state_from_caps (GstBaseAudioDecoder * codec,
- GstCaps * caps)
+/* automagically perform sanity checking of src caps;
+ * also extracts output data format */
+static gboolean
+gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
{
- GstStructure *structure;
- const GValue *codec_data;
+ GstBaseAudioDecoder *dec;
+ GstAudioState *state;
+ gboolean res = TRUE, changed;
- structure = gst_caps_get_structure (caps, 0);
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ state = &dec->ctx->state;
- if (codec->codec_data) {
- gst_buffer_unref (codec->codec_data);
- codec->codec_data = NULL;
- }
+ 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;
- codec_data = gst_structure_get_value (structure, "codec_data");
- if (codec_data && G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
- codec->codec_data = gst_value_get_buffer (codec_data);
+ /* 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;
}
- gst_structure_get_int (structure, "channels", &codec->state.channels);
- gst_structure_get_int (structure, "rate", &codec->state.rate);
- gst_structure_get_int (structure, "depth", &codec->state.sample_depth);
- gst_structure_get_int (structure, "width", &codec->state.bytes_per_sample);
- codec->state.bytes_per_sample /= 8;
- codec->state.frame_size =
- codec->state.bytes_per_sample * codec->state.channels;
+ 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_event (GstPad * pad, GstEvent * event)
+gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
{
- GstBaseAudioDecoder *codec;
- GstBaseAudioDecoderClass *codec_class;
- gboolean ret = FALSE;
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean res = TRUE;
- codec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_EOS:
- /* Flush any data still present in the adapters */
- gst_base_audio_decoder_flush (codec);
- ret = gst_pad_push_event (codec->srcpad, event);
- break;
- case GST_EVENT_FLUSH_STOP:
- gst_base_audio_decoder_reset (codec);
- ret = gst_pad_push_event (codec->srcpad, event);
- break;
- case GST_EVENT_NEWSEGMENT:
- {
- gboolean update;
- GstFormat format;
- gdouble rate, arate;
- gint64 start, stop, time;
+ GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
- gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
- &start, &stop, &time);
+ if (klass->set_format)
+ res = klass->set_format (dec, caps);
- if (format != GST_FORMAT_TIME)
- goto newseg_wrong_format;
+ g_object_unref (dec);
+ return res;
+}
- if (rate <= 0.0)
- goto newseg_wrong_rate;
+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);
- GST_DEBUG ("news egment %lld %lld", start, time);
- gst_segment_set_newsegment_full (&codec->state.segment,
- update, rate, arate, format, start, stop, time);
- ret = gst_pad_push_event (codec->srcpad, event);
- break;
+ /* 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)));
}
- default:
- ret = gst_pad_push_event (codec->srcpad, event);
- break;
+ } else {
+ GST_DEBUG_OBJECT (dec, "no output buffer");
}
- /* Let the subclass see the event too */
- if (codec_class->event) {
- if (!codec_class->event (codec, event)) {
- ret = FALSE;
- goto subclass_event_error;
+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;
}
}
-done:
- gst_object_unref (codec);
- return ret;
+ 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)));
-newseg_wrong_format:
- GST_DEBUG ("received non TIME newsegment");
- gst_event_unref (event);
- goto done;
+ 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");
+ }
-newseg_wrong_rate:
- GST_DEBUG ("negative rates not supported");
- gst_event_unref (event);
- goto done;
+ exit:
+ if (inbuf) {
+ buf = inbuf;
+ goto again;
+ }
+ }
-subclass_event_error:
- GST_DEBUG ("codec implementation failed to proces event");
- gst_event_unref (event);
- goto done;
+ return ret;
}
-static gboolean
-gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+GstFlowReturn
+gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf,
+ gint frames)
{
- GstBaseAudioDecoder *codec;
- GstBaseAudioDecoderClass *codec_class;
+ 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;
+ }
- codec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
-
- GST_DEBUG ("gst_base_audio_decoder_sink_setcaps %" GST_PTR_FORMAT, caps);
-
- /* Let the subclass provide the source caps and we will set them
- on the codec's source pad */
- if (codec_class->negotiate_src_caps) {
- GstCaps *src_caps;
- src_caps = codec_class->negotiate_src_caps (codec, caps);
- if (!gst_base_audio_decoder_set_src_caps (codec, src_caps)) {
- GST_DEBUG ("Caps negotiation failed!");
- g_object_unref (codec);
- gst_caps_unref (src_caps);
- return FALSE;
+ /* 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;
}
- gst_caps_unref (src_caps);
- } else {
- /* If the subclass does not provide a negotiate_src_caps method, then
- it will be responsible for calling gst_base_audio_decoder_set_src_caps
- with appropriate caps before we try to push buffers out */
- GST_DEBUG ("Subclass does not provide negotiate_src_caps, is that ok?");
}
- gst_base_audio_decoder_start (codec);
+ if (G_LIKELY (priv->frames.length))
+ ts = GST_BUFFER_TIMESTAMP (priv->frames.head->data);
+ else
+ ts = GST_CLOCK_TIME_NONE;
- g_object_unref (codec);
+ GST_DEBUG_OBJECT (dec, "leading frame ts %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (ts));
- return TRUE;
+ 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));
+ }
+
+ /* 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 %" 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 (buf));
+ 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;
+ }
+ }
+
+ if (G_LIKELY (buf)) {
+ 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;
+ }
+
+ 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 GstStateChangeReturn
-gst_base_audio_decoder_change_state (GstElement * element,
- GstStateChange transition)
+static GstFlowReturn
+gst_base_audio_decoder_handle_frame (GstBaseAudioDecoder * dec,
+ GstBaseAudioDecoderClass * klass, GstBuffer * buffer)
{
- GstBaseAudioDecoder *codec;
- GstBaseAudioDecoderClass *codec_class;
- GstStateChangeReturn ret;
+ 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");
+ }
- codec = GST_BASE_AUDIO_DECODER (element);
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (element);
+ return klass->handle_frame (dec, buffer);
+}
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- if (!gst_base_audio_decoder_start (codec)) {
- goto start_failed;
+/* 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;
}
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- if (!gst_base_audio_decoder_reset (codec)) {
- goto reset_failed;
+ /* 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;
}
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- break;
- default:
- break;
- }
+ 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 = parent_class->change_state (element, transition);
+ ret = gst_base_audio_decoder_handle_frame (dec, klass, buffer);
- 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:
+ /* 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;
-start_failed:
- {
- GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to start codec"));
- return GST_STATE_CHANGE_FAILURE;
- }
-reset_failed:
+ /* ERRORS */
+parse_failed:
{
- GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to reset codec"));
- return GST_STATE_CHANGE_FAILURE;
- }
-stop_failed:
- {
- GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to stop codec"));
- return GST_STATE_CHANGE_FAILURE;
+ GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("failed to parse stream"));
+ return GST_FLOW_ERROR;
}
}
-static void
-gst_base_audio_decoder_handle_discont (GstBaseAudioDecoder * codec,
- GstBuffer * buffer)
+static GstFlowReturn
+gst_base_audio_decoder_drain (GstBaseAudioDecoder * dec)
{
- GstBaseAudioDecoderClass *codec_class;
+ GstFlowReturn ret;
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+ 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);
- /* Reset codec on discont */
- if (codec->started) {
- gst_base_audio_decoder_reset (codec);
+ return ret;
}
+}
+
+/* hard == FLUSH, otherwise discont */
+static GstFlowReturn
+gst_base_audio_decoder_flush (GstBaseAudioDecoder * dec, gboolean hard)
+{
+ GstBaseAudioDecoderClass *klass;
+ GstFlowReturn ret = GST_FLOW_OK;
- codec->discont = TRUE;
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
- /* Let the subclass do its stuff too if that is needed */
- if (codec_class->handle_discont) {
- codec_class->handle_discont (codec, buffer);
+ GST_LOG_OBJECT (dec, "flush hard %d", hard);
+
+ if (!hard) {
+ ret = gst_base_audio_decoder_drain (dec);
+ } else {
+ gst_segment_init (&dec->segment, GST_FORMAT_TIME);
}
+ /* 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 (GstPad * pad, GstBuffer * buf)
+gst_base_audio_decoder_chain_forward (GstBaseAudioDecoder * dec,
+ GstBuffer * buffer)
{
- GstBaseAudioDecoder *codec;
- GstBaseAudioDecoderClass *codec_class;
- GstBuffer *outbuf;
GstFlowReturn ret;
- guint bytes_ready;
- guint64 timestamp;
- codec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+ /* 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 ("gst_base_audio_decoder_chain");
+ GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT,
+ buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
- /* Make sure we have started our codec */
- if (G_UNLIKELY (!codec->started)) {
- if (G_UNLIKELY (!gst_base_audio_decoder_start (codec))) {
- GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL),
- ("Failed to start codec"));
- gst_object_unref (codec);
- return GST_FLOW_ERROR;
+ 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;
}
- /* Handle timestamps */
- timestamp = GST_BUFFER_TIMESTAMP (buf);
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- GST_DEBUG ("buffer timestamp %" GST_TIME_FORMAT " duration:%"
- GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
- if (gst_adapter_available (codec->input_adapter) == 0) {
- codec->first_ts = timestamp;
+ /* 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);
}
- codec->last_ts = timestamp;
- }
-
- /* Check for discontinuity */
- if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
- GST_DEBUG ("received DISCONT buffer");
- gst_base_audio_decoder_handle_discont (codec, buf);
- }
-
- /* Push buffer to the input adapter so the codec can
- take data from it as needed */
- codec->bytes_in += GST_BUFFER_SIZE (buf);
- gst_adapter_push (codec->input_adapter, buf);
-
- GST_DEBUG ("Input buffer size: %ld bytes", GST_BUFFER_SIZE (buf));
-
- /* Check if we have enough data to be processed. While we have
- enough data on the input adapter, instruct the element to
- process it */
- ret = GST_FLOW_OK;
- bytes_ready = gst_adapter_available (codec->input_adapter);
- while (ret == GST_FLOW_OK && bytes_ready > 0 &&
- bytes_ready >= codec->input_buffer_size) {
- GST_DEBUG ("Processing data");
- ret = codec_class->process_data (codec);
- bytes_ready = gst_adapter_available (codec->input_adapter);
- GST_DEBUG ("%ld bytes remaining on the input", bytes_ready);
- }
-
- /* FIXME: is it possible that we have enough data in the output
- adapter but we have to wait for more data before we can
- push buffers out? In that case we need a custom GST_FLOW.
- Not sure if we could handle pushing buffers here in that
- case though, since we always push in output_buffer_size
- blocks. */
-
- /* If no error was raised, check if we can push buffers out */
- if (G_LIKELY (ret == GST_FLOW_OK)) {
- bytes_ready = gst_adapter_available (codec->output_adapter);
- GST_DEBUG ("Processed input correctly");
- GST_DEBUG ("%ld bytes on the output", bytes_ready);
-
- /* If the subclass wants to control how buffers are pushed out
- let it do it */
- if (bytes_ready > 0 && codec_class->push_data) {
- GST_DEBUG ("Calling push_data on the subclass");
- codec_class->push_data (codec);
- } else if (bytes_ready > 0 && bytes_ready >= codec->output_buffer_size) {
- /* We have enough data in the output adapter, so take a buffer, apply
- clipping, push it out and repeat while we have enough data */
- guint bytes_to_push;
-
- bytes_to_push =
- codec->output_buffer_size ? codec->output_buffer_size : bytes_ready;
-
- do {
- GST_DEBUG ("Pushing a buffer out (%ld bytes)", bytes_to_push);
-
- outbuf = gst_adapter_take_buffer (codec->output_adapter, bytes_to_push);
-
- /* Set buffer timestamp/duration if needed (and possible) */
- if (!GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) && codec->first_ts != -1) {
- GST_DEBUG ("Computing output buffer timestamp");
- GST_BUFFER_TIMESTAMP (outbuf) = codec->first_ts;
- }
- if (!GST_BUFFER_DURATION_IS_VALID (outbuf) && codec->state.frame_size) {
- guint nsamples;
- GST_DEBUG ("Computing output buffer duration");
- nsamples = GST_BUFFER_SIZE (outbuf) / codec->state.frame_size;
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale_int (GST_SECOND, nsamples,
- codec->state.rate);
- }
+ priv->queued = g_list_delete_link (priv->queued, priv->queued);
+ }
- if (codec->first_ts != -1) {
- codec->first_ts += GST_BUFFER_DURATION (outbuf);
- if (codec->first_ts > codec->last_ts) {
- codec->last_ts = codec->first_ts;
- }
- }
+ return res;
+}
- GST_DEBUG ("out buffer timestamp %" GST_TIME_FORMAT " duration:%"
- GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
+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);
+ }
- /* Clip buffer */
- if (codec->state.segment.format == GST_FORMAT_TIME ||
- codec->state.segment.format == GST_FORMAT_DEFAULT) {
- GST_DEBUG ("Clipping buffer");
- outbuf = gst_audio_buffer_clip (outbuf, &codec->state.segment,
- codec->state.rate, codec->state.frame_size);
- }
+ 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)));
- /* Set DISCONT flag on the output buffer if needed */
- if (G_LIKELY (outbuf)) {
- if (G_UNLIKELY (codec->discont)) {
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- codec->discont = FALSE;
- GST_DEBUG ("Buffer is discont");
- }
+ /* add buffer to gather queue */
+ priv->gather = g_list_prepend (priv->gather, buf);
+ }
- ret = gst_base_audio_decoder_push_buffer (codec, outbuf);
- }
+ return result;
+}
- /* See if we can push another buffer */
- bytes_ready = gst_adapter_available (codec->output_adapter);
- GST_DEBUG ("%ld bytes left on the output", bytes_ready);
- } while (ret == GST_FLOW_OK && bytes_ready >= bytes_to_push);
- } else {
- /* We need more data before we can push a buffer out */
- GST_DEBUG ("Not pushing out, need more data");
- ret = GST_FLOW_OK;
- }
- } else {
- /* We got an error */
- GST_DEBUG ("Got error while processing data");
+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)) {
+ GST_DEBUG_OBJECT (dec, "handling discont");
+ gst_base_audio_decoder_flush (dec, FALSE);
+ dec->priv->discont = TRUE;
}
- GST_DEBUG ("chain-done");
+ 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;
}
-/* ----- Element public API ----- */
-
-/**
- * gst_base_audio_decoder_reset:
- * @codec: The #GstBaseAudioDecoder instance.
- *
- * Resets the codec.
- *
- * This method will also invoke the subclass's reset virtual method
- * if available. Niotice that reseting the codec will clear the
- * input and output adapters.
- *
- * Returns: TRUE if the start operation was successful.
- */
-gboolean
-gst_base_audio_decoder_reset (GstBaseAudioDecoder * codec)
+static gboolean
+gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
+ GstEvent * event)
{
- GstBaseAudioDecoderClass *codec_class;
+ gboolean handled = FALSE;
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_NEWSEGMENT:
+ {
+ GstFormat format;
+ gdouble rate, arate;
+ gint64 start, stop, time;
+ gboolean update;
- GST_DEBUG ("gst_base_audio_decoder_reset");
+ gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+ &start, &stop, &time);
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+ if (format == GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (dec, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
+ " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
+ ", accum %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (dec->segment.start),
+ GST_TIME_ARGS (dec->segment.stop),
+ GST_TIME_ARGS (dec->segment.time),
+ GST_TIME_ARGS (dec->segment.accum));
+ } else {
+ GST_DEBUG_OBJECT (dec, "received NEW_SEGMENT %" G_GINT64_FORMAT
+ " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
+ ", accum %" G_GINT64_FORMAT,
+ dec->segment.start, dec->segment.stop,
+ dec->segment.time, dec->segment.accum);
+ GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
+ break;
+ }
- gst_adapter_clear (codec->input_adapter);
- gst_adapter_clear (codec->output_adapter);
+ /* 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);
+ }
- /* FIXME: is this needed? */
- gst_segment_init (&codec->state.segment, GST_FORMAT_TIME);
+ /* and follow along with segment */
+ gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
+ format, start, stop, time);
+ break;
+ }
- codec->first_ts = -1;
- codec->last_ts = -1;
+ case GST_EVENT_FLUSH_START:
+ break;
+
+ case GST_EVENT_FLUSH_STOP:
+ /* prepare for fresh start */
+ gst_base_audio_decoder_flush (dec, FALSE);
+ break;
+
+ case GST_EVENT_EOS:
+ gst_base_audio_decoder_drain (dec);
+ break;
- if (codec_class->reset) {
- codec_class->reset (codec);
+ default:
+ break;
}
- return TRUE;
+ return handled;
}
-/**
- * gst_base_audio_decoder_stop:
- * @codec: The #GstBaseAudioDecoder instance.
- *
- * Stop the codec. Normally this will be used for closing resource.
- *
- * This method will also invoke the subclass's stop virtual method
- * if available.
- *
- * Returns: TRUE if the start operation was successful.
- */
-gboolean
-gst_base_audio_decoder_stop (GstBaseAudioDecoder * codec)
+static gboolean
+gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
{
- GstBaseAudioDecoderClass *codec_class;
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean handled = FALSE;
+ gboolean ret = TRUE;
- GST_DEBUG ("gst_base_audio_decoder_stop");
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+ GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
+ GST_EVENT_TYPE_NAME (event));
- gst_base_audio_decoder_reset (codec);
+ if (klass->event)
+ handled = klass->event (dec, event);
- codec->bytes_in = 0;
- codec->bytes_out = 0;
+ if (!handled)
+ handled = gst_base_audio_decoder_sink_eventfunc (dec, event);
- if (codec_class->stop) {
- codec_class->stop (codec);
- }
+ if (!handled)
+ ret = gst_pad_event_default (pad, event);
- codec->started = FALSE;
+ GST_DEBUG_OBJECT (dec, "event handled");
- return TRUE;
+ gst_object_unref (dec);
+ return ret;
}
-/**
- * gst_base_audio_decoder_start:
- * @codec: The #GstBaseAudioDecoder instance.
- *
- * Setup the codec so it can start processing data. Normally
- * this will be used for opening resources needed for operation.
- *
- * This method will also invoke the subclass's start virtual method
- * if available.
- *
- * Returns: TRUE if the start operation was successful.
- */
-gboolean
-gst_base_audio_decoder_start (GstBaseAudioDecoder * codec)
+static gboolean
+gst_base_audio_decoder_src_event (GstPad * pad, GstEvent * event)
{
- GstBaseAudioDecoderClass *codec_class;
+ GstBaseAudioDecoder *dec;
+ GstBaseAudioDecoderClass *klass;
+ gboolean res = FALSE;
- GST_DEBUG ("gst_base_audio_decoder_start");
+ dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+ GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
+ GST_EVENT_TYPE_NAME (event));
- gst_base_audio_decoder_reset (codec);
+ 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 */
+ res = gst_pad_push_event (dec->sinkpad, event);
+
+ /* if upstream fails for a non-time seek, maybe we can help */
+ if (G_LIKELY (format == GST_FORMAT_TIME || res))
+ break;
+
+ /* 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);
- codec->bytes_in = 0;
- codec->bytes_out = 0;
+ return res;
- if (codec_class->start) {
- codec_class->start (codec);
+ /* ERRORS */
+convert_error:
+ {
+ GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
+ goto done;
}
+}
- codec->started = TRUE;
+static gboolean
+gst_base_audio_decoder_sink_query (GstPad * pad, GstQuery * query)
+{
+ gboolean res = TRUE;
+ GstBaseAudioDecoder *dec;
- return TRUE;
+ 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;
}
-/**
- * gst_base_audio_decoder_flush:
- * @codec: The #GstBaseAudioDecoder instance.
- *
- * Flushes the input and output adapters. Subclasses should provide
- * a flush_input implementation to allow flushing the input adapter.
- * For the output adapter subclasses should provide a flush_output
- * implementation. If no flush_output implementation is provided
- * the output adapter will be flushed by pushing a single buffer
- * containing all the data present in the output adapter.
- *
- * It is guaranteed that any data present in the adapters will be cleared
- * after calling this method even if the operation flush
- * operation was not successfull.
- *
- * Returns: TRUE if the flush operation was successful (any data present in
- * the adapters was properly processed).
- */
-gboolean
-gst_base_audio_decoder_flush (GstBaseAudioDecoder * codec)
+static const GstQueryType *
+gst_base_audio_decoder_get_query_types (GstPad * pad)
{
- GstFlowReturn ret_i = GST_FLOW_OK;
- GstFlowReturn ret_o = GST_FLOW_OK;
- guint bytes;
- GstBaseAudioDecoderClass *codec_class;
+ 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;
- GST_DEBUG ("gst_base_audio_decoder_flush");
+ dec = GST_BASE_AUDIO_DECODER (GST_PAD_PARENT (pad));
+ peerpad = gst_pad_get_peer (GST_PAD (dec->sinkpad));
- codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+ GST_LOG_OBJECT (dec, "handling query: %" GST_PTR_FORMAT, query);
- /* Flush input adapter */
- bytes = gst_adapter_available (codec->input_adapter);
- if (bytes > 0) {
- GST_DEBUG ("Flushing input adapter");
- /* If the subclass provides a flush_input implementation, use that.
- Otherwise we will clear the adapter and lose the data */
- if (codec_class->flush_input) {
- ret_i = codec_class->flush_input (codec);
- if (ret_i != GST_FLOW_OK) {
- GST_DEBUG ("failed to flush input");
+ switch (GST_QUERY_TYPE (query)) {
+ 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;
}
- } else {
- GST_DEBUG ("Received EOS but cannot flush input, data will be lost");
- ret_i = GST_FLOW_ERROR;
+
+ /* 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;
}
- gst_adapter_clear (codec->input_adapter);
- }
-
- /* Flush output adapter */
- bytes = gst_adapter_available (codec->output_adapter);
- if (bytes > 0) {
- /* If the subclass provides a flush_output implementation, use that.
- Otherwise just push a single buffer with the adapter contents */
- GST_DEBUG ("Flushing output adapter");
- if (codec_class->flush_output) {
- ret_o = codec_class->flush_output (codec);
- if (ret_o != GST_FLOW_OK) {
- GST_DEBUG ("failed to flush output (flush_output)");
- }
- } else {
- GstBuffer *outbuf =
- gst_adapter_take_buffer (codec->output_adapter, bytes);
- ret_o = gst_base_audio_decoder_push_buffer (codec, outbuf);
- gst_buffer_unref (outbuf);
- if (ret_o != GST_FLOW_OK) {
- GST_DEBUG ("Forced output flush failed");
+ 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;
}
- gst_adapter_clear (codec->output_adapter);
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
}
- return (ret_i == GST_FLOW_OK && ret_o == GST_FLOW_OK);
+ gst_object_unref (peerpad);
+ return res;
}
-/**
- * gst_base_audio_decoder_set_src_caps:
- * @codec: #GstBaseAudioDecoder instance
- * @caps: The caps to set on the source pad of @codec.
- *
- * Attempts to set @caps as the source caps of @codec. If the new caps
- * are accepted on the source pad, this will issue a flush on the adapters
- * to ensure that any data received with the old caps is processed first
- * and a reset of the codec.
- *
- * Returns: TRUE if caps were set successfully.
- */
-gboolean
-gst_base_audio_decoder_set_src_caps (GstBaseAudioDecoder * codec,
- GstCaps * caps)
+static gboolean
+gst_base_audio_decoder_stop (GstBaseAudioDecoder * dec)
{
- gboolean ret;
+ GstBaseAudioDecoderClass *klass;
+ gboolean ret = TRUE;
- GST_DEBUG ("gst_base_audio_decoder_set_src_caps %" GST_PTR_FORMAT, caps);
+ GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_stop");
- /* First, check if the pad accepts the new caps */
- if (!gst_pad_accept_caps (codec->srcpad, caps)) {
- GST_DEBUG ("pad does not accept new caps");
- return FALSE;
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
+
+ if (klass->stop) {
+ ret = klass->stop (dec);
}
- /* If we have data in our adapters we should probably flush first */
- gst_base_audio_decoder_flush (codec);
+ /* clean up */
+ gst_base_audio_decoder_reset (dec, TRUE);
+
+ if (ret)
+ dec->priv->active = FALSE;
- /* Set the caps on the pad */
- ret = gst_pad_set_caps (codec->srcpad, caps);
+ return TRUE;
+}
- /* And update the state of the codec from the caps */
- if (ret) {
- gst_base_audio_decoder_read_state_from_caps (codec, caps);
- codec->caps_set = 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);
}
- return ret;
+ if (ret)
+ dec->priv->active = TRUE;
+
+ return TRUE;
}
-/**
- * gst_base_audio_decoder_push_buffer:
- * @codec: #GstBaseAudioDecoder instance
- * @buffer: a #GstBuffer.
- *
- * Pushes a buffer through the source pad.
- *
- * Returns: a #GstFlowReturn indicating the result of the push operation.
- */
-GstFlowReturn
-gst_base_audio_decoder_push_buffer (GstBaseAudioDecoder * codec,
- GstBuffer * buffer)
+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)
{
- codec->bytes_out += GST_BUFFER_SIZE (buffer);
- return gst_pad_push (codec->srcpad, buffer);
+ 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;
+ }
}
projects / platform / upstream / gst-plugins-base.git / commitdiff