--- /dev/null
+/* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral Quiroga <itoral@igalia.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 codec elements
+ * @see_also: #GstBaseTransform, #GstBaseSource, #GstBaseSink
+ *
+ * #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.
+ *
+ * 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.
+ *
+ * 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.
+ *
+ * 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.
+ *
+ * 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().
+ *
+ * #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.
+ *
+ * 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.
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstbaseaudiodecoder.h"
+#include <gst/audio/audio.h>
+#include <string.h>
+
+/*
+ * FIXME: maybe we need more work with the segments (see ac3 decoder)
+ */
+
+GST_DEBUG_CATEGORY (baseaudiodecoder_debug);
+#define GST_CAT_DEFAULT baseaudiodecoder_debug
+
+/* ----- Signals and properties ----- */
+enum
+{
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_INPUT_BUFFER_SIZE,
+ PROP_OUTPUT_BUFFER_SIZE
+};
+
+/* ----- Function prototypes ----- */
+
+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 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_sink_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);
+
+/* ----- 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
+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);
+
+ 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));
+}
+
+static void
+gst_base_audio_decoder_init (GstBaseAudioDecoder * codec,
+ GstBaseAudioDecoderClass * klass)
+{
+ GstPadTemplate *pad_template;
+
+ GST_DEBUG ("gst_base_audio_decoder_init");
+
+ /* 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);
+
+ /* 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;
+}
+
+static void
+gst_base_audio_decoder_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstBaseAudioDecoder *codec;
+
+ codec = GST_BASE_AUDIO_DECODER (object);
+
+ 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;
+
+ codec = GST_BASE_AUDIO_DECODER (object);
+
+ 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;
+ }
+}
+
+static void
+gst_base_audio_decoder_finalize (GObject * object)
+{
+ GstBaseAudioDecoder *codec;
+
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (object));
+ codec = GST_BASE_AUDIO_DECODER (object);
+
+ if (codec->input_adapter) {
+ g_object_unref (codec->input_adapter);
+ }
+ if (codec->output_adapter) {
+ g_object_unref (codec->output_adapter);
+ }
+ if (codec->codec_data) {
+ gst_buffer_unref (codec->codec_data);
+ }
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* ----- Private element implementation ----- */
+
+static void
+gst_base_audio_decoder_read_state_from_caps (GstBaseAudioDecoder * codec,
+ GstCaps * caps)
+{
+ GstStructure *structure;
+ const GValue *codec_data;
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ if (codec->codec_data) {
+ gst_buffer_unref (codec->codec_data);
+ codec->codec_data = NULL;
+ }
+
+ 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);
+ }
+
+ 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;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstBaseAudioDecoder *codec;
+ GstBaseAudioDecoderClass *codec_class;
+ gboolean ret = FALSE;
+
+ codec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+
+ 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_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+ &start, &stop, &time);
+
+ if (format != GST_FORMAT_TIME)
+ goto newseg_wrong_format;
+
+ if (rate <= 0.0)
+ goto newseg_wrong_rate;
+
+ 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;
+ }
+ default:
+ ret = gst_pad_push_event (codec->srcpad, event);
+ break;
+ }
+
+ /* Let the subclass see the event too */
+ if (codec_class->event) {
+ if (!codec_class->event (codec, event)) {
+ ret = FALSE;
+ goto subclass_event_error;
+ }
+ }
+
+done:
+ gst_object_unref (codec);
+ return ret;
+
+newseg_wrong_format:
+ GST_DEBUG ("received non TIME newsegment");
+ gst_event_unref (event);
+ goto done;
+
+newseg_wrong_rate:
+ GST_DEBUG ("negative rates not supported");
+ gst_event_unref (event);
+ goto done;
+
+subclass_event_error:
+ GST_DEBUG ("codec implementation failed to proces event");
+ gst_event_unref (event);
+ goto done;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstBaseAudioDecoder *codec;
+ GstBaseAudioDecoderClass *codec_class;
+
+ 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;
+ }
+ 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);
+
+ g_object_unref (codec);
+
+ return TRUE;
+}
+
+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:
+ if (!gst_base_audio_decoder_start (codec)) {
+ goto start_failed;
+ }
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ if (!gst_base_audio_decoder_reset (codec)) {
+ goto reset_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;
+ }
+reset_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;
+ }
+}
+
+static void
+gst_base_audio_decoder_handle_discont (GstBaseAudioDecoder * codec,
+ GstBuffer * buffer)
+{
+ GstBaseAudioDecoderClass *codec_class;
+
+ codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+
+ /* Reset codec on discont */
+ if (codec->started) {
+ gst_base_audio_decoder_reset (codec);
+ }
+
+ codec->discont = TRUE;
+
+ /* Let the subclass do its stuff too if that is needed */
+ if (codec_class->handle_discont) {
+ codec_class->handle_discont (codec, buffer);
+ }
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buf)
+{
+ 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);
+
+ GST_DEBUG ("gst_base_audio_decoder_chain");
+
+ /* 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;
+ }
+ }
+
+ /* 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;
+ }
+ 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);
+ }
+
+ 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;
+ }
+ }
+
+ 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)));
+
+ /* 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);
+ }
+
+ /* 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");
+ }
+
+ ret = gst_base_audio_decoder_push_buffer (codec, outbuf);
+ }
+
+ /* 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");
+ }
+
+ GST_DEBUG ("chain-done");
+
+ 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)
+{
+ GstBaseAudioDecoderClass *codec_class;
+
+ GST_DEBUG ("gst_base_audio_decoder_reset");
+
+ codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+
+ gst_adapter_clear (codec->input_adapter);
+ gst_adapter_clear (codec->output_adapter);
+
+ /* FIXME: is this needed? */
+ gst_segment_init (&codec->state.segment, GST_FORMAT_TIME);
+
+ codec->first_ts = -1;
+ codec->last_ts = -1;
+
+ if (codec_class->reset) {
+ codec_class->reset (codec);
+ }
+
+ return TRUE;
+}
+
+/**
+ * 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)
+{
+ GstBaseAudioDecoderClass *codec_class;
+
+ GST_DEBUG ("gst_base_audio_decoder_stop");
+
+ codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+
+ gst_base_audio_decoder_reset (codec);
+
+ codec->bytes_in = 0;
+ codec->bytes_out = 0;
+
+ if (codec_class->stop) {
+ codec_class->stop (codec);
+ }
+
+ codec->started = FALSE;
+
+ return TRUE;
+}
+
+/**
+ * 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)
+{
+ GstBaseAudioDecoderClass *codec_class;
+
+ GST_DEBUG ("gst_base_audio_decoder_start");
+
+ codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+
+ gst_base_audio_decoder_reset (codec);
+
+ codec->bytes_in = 0;
+ codec->bytes_out = 0;
+
+ if (codec_class->start) {
+ codec_class->start (codec);
+ }
+
+ codec->started = TRUE;
+
+ return TRUE;
+}
+
+/**
+ * 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)
+{
+ GstFlowReturn ret_i = GST_FLOW_OK;
+ GstFlowReturn ret_o = GST_FLOW_OK;
+ guint bytes;
+ GstBaseAudioDecoderClass *codec_class;
+
+ GST_DEBUG ("gst_base_audio_decoder_flush");
+
+ codec_class = GST_BASE_AUDIO_DECODER_GET_CLASS (codec);
+
+ /* 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");
+ }
+ } else {
+ GST_DEBUG ("Received EOS but cannot flush input, data will be lost");
+ ret_i = GST_FLOW_ERROR;
+ }
+ 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");
+ }
+ }
+ gst_adapter_clear (codec->output_adapter);
+ }
+
+ return (ret_i == GST_FLOW_OK && ret_o == GST_FLOW_OK);
+}
+
+/**
+ * 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)
+{
+ gboolean ret;
+
+ GST_DEBUG ("gst_base_audio_decoder_set_src_caps %" GST_PTR_FORMAT, caps);
+
+ /* 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;
+ }
+
+ /* If we have data in our adapters we should probably flush first */
+ gst_base_audio_decoder_flush (codec);
+
+ /* Set the caps on the pad */
+ ret = gst_pad_set_caps (codec->srcpad, caps);
+
+ /* 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;
+ }
+
+ return ret;
+}
+
+/**
+ * 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)
+{
+ codec->bytes_out += GST_BUFFER_SIZE (buffer);
+ return gst_pad_push (codec->srcpad, buffer);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral Quiroga <itoral@igalia.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/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)
+
+/**
+ * GST_BASE_AUDIO_DECODER_INPUT_ADAPTER:
+ * @obj: base audio codec instance
+ *
+ * Gives the pointer to the input #GstAdapter object of the element.
+ */
+#define GST_BASE_AUDIO_DECODER_INPUT_ADAPTER(obj) (((GstBaseAudioDecoder *) (obj))->input_adapter)
+
+/**
+ * GST_BASE_AUDIO_DECODER_OUTPUT_ADAPTER:
+ * @obj: base audio codec instance
+ *
+ * Gives the pointer to the output #GstAdapter object of the element.
+ */
+#define GST_BASE_AUDIO_DECODER_OUTPUT_ADAPTER(obj) (((GstBaseAudioDecoder *) (obj))->output_adapter)
+
+typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
+typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
+typedef struct _GstAudioState GstAudioState;
+
+struct _GstAudioState
+{
+ gint channels;
+ gint rate;
+ gint bytes_per_sample;
+ gint sample_depth;
+ gint frame_size;
+ GstSegment segment;
+};
+
+/**
+ * GstBaseAudioDecoder:
+ * @element: the parent element.
+ * @caps_set: whether caps have been set on the codec's source pad.
+ * @sinkpad: the sink pad.
+ * @srcpad: the source pad.
+ * @input_adapter: the input adapter that will be filled with the input buffers.
+ * @output_adapter: the output adapter. Subclasses will read from the input
+ * adapter, process the data and fill the output adapter with the result.
+ * @input_buffer_size: The minimum amount of data that should be present on the
+ * input adapter for the codec to process it.
+ * @output_buffer_size: The minimum amount of data that should be present on the
+ * output adapter for the codec to push buffers out.
+ * @bytes_in: total bytes that have been received.
+ * @bytes_out: total bytes that have been pushed out.
+ * @discont: whether the next buffer to push represents a discontinuity in the
+ * stream.
+ * @state: Audio stream information. See #GstAudioState for details.
+ * @codec_data: The codec data.
+ * @started: Whether the codec has been started and is ready to process data
+ * or not.
+ * @first_ts: timestamp of the first buffer in the input adapter.
+ * @last_ts: timestamp of the last buffer in the input adapter.
+ *
+ * The opaque #GstBaseAudioDecoder data structure.
+ */
+struct _GstBaseAudioDecoder
+{
+ GstElement element;
+
+ /*< private >*/
+ gboolean caps_set;
+
+ /*< protected >*/
+ GstPad *sinkpad;
+ GstPad *srcpad;
+ GstAdapter *input_adapter;
+ GstAdapter *output_adapter;
+ guint input_buffer_size;
+ guint output_buffer_size;
+ guint64 bytes_in;
+ guint64 bytes_out;
+ gboolean discont;
+ GstAudioState state;
+ GstBuffer *codec_data;
+ gboolean started;
+
+ guint64 first_ts;
+ guint64 last_ts;
+};
+
+/**
+ * GstBaseAudioDecoderClass:
+ * @parent_class: Element parent class
+ * @start: Start processing. Ideal for opening resources in the subclass
+ * @stop: Stop processing. Subclasses should use this to close resources.
+ * @reset: Resets the codec. Called on discontinuities, etc.
+ * @event: Override this to handle events arriving on the sink pad.
+ * @handle_discont: Override to be notified on discontinuities.
+ * @flush_input: Subclasses may implement this to flush the input adapter,
+ * processing any data present in it and filling the output adapter with the
+ * result. This could be necessary if it is possible for the codec to
+ * receive an end-of-stream event before all the data in the input
+ * adapter has been processed.
+ * @flush_output: Subclasses may implement this to flush the output adapter,
+ * pushing buffers out through the codec's source pad when the end-of-stream
+ * event is received and there is data waiting to be processed in the
+ * adapters.
+ * @process_data: Subclasses must implement this. They should read from the
+ * input adapter, encode/decode the data present in it and fill the
+ * output adapter with the result.
+ * @push_data: Normally, #GstBaseAudioDecoder will handle pushing buffers out.
+ * However, it is possible for developers to take control of when and how
+ * buffers are pushed out by overriding this method. If subclasses provide
+ * an implementation, #GstBaseAudioDecoder will not push any buffers,
+ * instead, whenever there is data on the output adapter, it will call this
+ * method on the subclass, which would be the sole responsible for
+ * pushing the buffers out when appropriate.
+ * @negotiate_src_caps: Subclasses can implement this method to provide
+ * appropriate caps to be set on the codec's source pad. If they don't
+ * provide this, they will be responsible for calling
+ * gst_base_audio_decoder_set_src_caps when appropriate.
+ */
+struct _GstBaseAudioDecoderClass
+{
+ GstElementClass parent_class;
+
+ gboolean (*start) (GstBaseAudioDecoder *codec);
+ gboolean (*stop) (GstBaseAudioDecoder *codec);
+ gboolean (*reset) (GstBaseAudioDecoder *codec);
+
+ GstFlowReturn (*event) (GstBaseAudioDecoder *codec, GstEvent *event);
+ void (*handle_discont) (GstBaseAudioDecoder *codec, GstBuffer *buffer);
+ gboolean (*flush_input) (GstBaseAudioDecoder *codec);
+ gboolean (*flush_output) (GstBaseAudioDecoder *codec);
+ GstFlowReturn (*process_data) (GstBaseAudioDecoder *codec);
+ GstFlowReturn (*push_data) (GstBaseAudioDecoder *codec);
+ GstCaps * (*negotiate_src_caps) (GstBaseAudioDecoder *codec,
+ GstCaps *sink_caps);
+};
+
+GType gst_base_audio_decoder_get_type (void);
+gboolean gst_base_audio_decoder_reset (GstBaseAudioDecoder *codec);
+gboolean gst_base_audio_decoder_stop (GstBaseAudioDecoder *codec);
+gboolean gst_base_audio_decoder_start (GstBaseAudioDecoder *codec);
+gboolean gst_base_audio_decoder_flush (GstBaseAudioDecoder *codec);
+gboolean gst_base_audio_decoder_set_src_caps (GstBaseAudioDecoder *codec,
+ GstCaps *caps);
+GstFlowReturn gst_base_audio_decoder_push_buffer (GstBaseAudioDecoder *codec,
+ GstBuffer *buffer);
+
+G_END_DECLS
+
+#endif
+
projects / platform / upstream / gstreamer.git / commitdiff