--- /dev/null
+/* GStreamer
+ * Copyright (C) 2009 Igalia S.L.
+ * Author: Iago Toral <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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstbaseaudiodecoder.h"
+
+#include <string.h>
+
+GST_DEBUG_CATEGORY_EXTERN (baseaudio_debug);
+#define GST_CAT_DEFAULT baseaudio_debug
+
+static void gst_base_audio_decoder_finalize (GObject * object);
+
+static gboolean gst_base_audio_decoder_sink_setcaps (GstPad * pad,
+ GstCaps * caps);
+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 GstFlowReturn gst_base_audio_decoder_chain (GstPad * pad,
+ GstBuffer * buf);
+static gboolean gst_base_audio_decoder_sink_query (GstPad * pad,
+ GstQuery * query);
+static GstStateChangeReturn gst_base_audio_decoder_change_state (GstElement *
+ element, GstStateChange transition);
+static const GstQueryType *gst_base_audio_decoder_get_query_types (GstPad *
+ pad);
+static gboolean gst_base_audio_decoder_src_query (GstPad * pad,
+ GstQuery * query);
+static gboolean gst_base_audio_decoder_src_convert (GstPad * pad,
+ GstFormat src_format, gint64 src_value, GstFormat * dest_format,
+ gint64 * dest_value);
+static void gst_base_audio_decoder_reset (GstBaseAudioDecoder *
+ base_audio_decoder);
+
+static guint64
+gst_base_audio_decoder_get_timestamp (GstBaseAudioDecoder * base_audio_decoder,
+ int picture_number);
+static guint64
+gst_base_audio_decoder_get_field_timestamp (GstBaseAudioDecoder *
+ base_audio_decoder, int field_offset);
+static GstAudioFrame *gst_base_audio_decoder_new_frame (GstBaseAudioDecoder *
+ base_audio_decoder);
+static void gst_base_audio_decoder_free_frame (GstAudioFrame * frame);
+
+GST_BOILERPLATE (GstBaseAudioDecoder, gst_base_audio_decoder,
+ GstBaseAudioCodec, GST_TYPE_BASE_AUDIO_CODEC);
+
+static void
+gst_base_audio_decoder_base_init (gpointer g_class)
+{
+
+}
+
+static void
+gst_base_audio_decoder_class_init (GstBaseAudioDecoderClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ gstelement_class = GST_ELEMENT_CLASS (klass);
+
+ gobject_class->finalize = gst_base_audio_decoder_finalize;
+
+ gstelement_class->change_state = gst_base_audio_decoder_change_state;
+
+ parent_class = g_type_class_peek_parent (klass);
+}
+
+static void
+gst_base_audio_decoder_init (GstBaseAudioDecoder * base_audio_decoder,
+ GstBaseAudioDecoderClass * klass)
+{
+ GstPad *pad;
+
+ GST_DEBUG ("gst_base_audio_decoder_init");
+
+ pad = GST_BASE_AUDIO_CODEC_SINK_PAD (base_audio_decoder);
+
+ gst_pad_set_chain_function (pad, gst_base_audio_decoder_chain);
+ gst_pad_set_event_function (pad, gst_base_audio_decoder_sink_event);
+ gst_pad_set_setcaps_function (pad, gst_base_audio_decoder_sink_setcaps);
+ gst_pad_set_query_function (pad, gst_base_audio_decoder_sink_query);
+
+ pad = GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder);
+
+ gst_pad_set_event_function (pad, gst_base_audio_decoder_src_event);
+ gst_pad_set_query_type_function (pad, gst_base_audio_decoder_get_query_types);
+ gst_pad_set_query_function (pad, gst_base_audio_decoder_src_query);
+
+ base_audio_decoder->input_adapter = gst_adapter_new ();
+ base_audio_decoder->output_adapter = gst_adapter_new ();
+
+ gst_segment_init (&base_audio_decoder->state.segment, GST_FORMAT_TIME);
+ gst_base_audio_decoder_reset (base_audio_decoder);
+
+ base_audio_decoder->current_frame =
+ gst_base_audio_decoder_new_frame (base_audio_decoder);
+
+ base_audio_decoder->sink_clipping = TRUE;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstBaseAudioDecoder *base_audio_decoder;
+ GstBaseAudioDecoderClass *base_audio_decoder_class;
+ GstStructure *structure;
+ const GValue *codec_data;
+
+ base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ base_audio_decoder_class =
+ GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+ GST_DEBUG ("setcaps %" GST_PTR_FORMAT, caps);
+
+ if (base_audio_decoder->codec_data) {
+ gst_buffer_unref (base_audio_decoder->codec_data);
+ base_audio_decoder->codec_data = NULL;
+ }
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ codec_data = gst_structure_get_value (structure, "codec_data");
+ if (codec_data && G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
+ base_audio_decoder->codec_data = gst_value_get_buffer (codec_data);
+ }
+
+ if (base_audio_decoder_class->start) {
+ base_audio_decoder_class->start (base_audio_decoder);
+ }
+
+ g_object_unref (base_audio_decoder);
+
+ return TRUE;
+}
+
+static void
+gst_base_audio_decoder_finalize (GObject * object)
+{
+ GstBaseAudioDecoder *base_audio_decoder;
+ GstBaseAudioDecoderClass *base_audio_decoder_class;
+
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (object));
+ base_audio_decoder = GST_BASE_AUDIO_DECODER (object);
+ base_audio_decoder_class = GST_BASE_AUDIO_DECODER_GET_CLASS (object);
+
+ gst_base_audio_decoder_reset (base_audio_decoder);
+
+ GST_DEBUG_OBJECT (object, "finalize");
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstBaseAudioDecoder *base_audio_decoder;
+ GstBaseAudioDecoderClass *base_audio_decoder_class;
+ gboolean ret = FALSE;
+
+ base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ base_audio_decoder_class =
+ GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_EOS:
+ {
+ GstAudioFrame *frame;
+
+ frame = g_malloc0 (sizeof (GstAudioFrame));
+ frame->presentation_frame_number =
+ base_audio_decoder->presentation_frame_number;
+ frame->presentation_duration = 0;
+ base_audio_decoder->presentation_frame_number++;
+
+ base_audio_decoder->frames =
+ g_list_append (base_audio_decoder->frames, frame);
+ if (base_audio_decoder_class->finish) {
+ base_audio_decoder_class->finish (base_audio_decoder, frame);
+ }
+
+ ret =
+ gst_pad_push_event (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+ event);
+ }
+ break;
+ case GST_EVENT_NEWSEGMENT:
+ {
+ gboolean update;
+ double rate;
+ double applied_rate;
+ GstFormat format;
+ gint64 start;
+ gint64 stop;
+ gint64 position;
+
+ gst_event_parse_new_segment_full (event, &update, &rate,
+ &applied_rate, &format, &start, &stop, &position);
+
+ if (format != GST_FORMAT_TIME)
+ goto newseg_wrong_format;
+
+ GST_DEBUG ("new segment %lld %lld", start, position);
+
+ gst_segment_set_newsegment_full (&base_audio_decoder->state.segment,
+ update, rate, applied_rate, format, start, stop, position);
+
+ ret =
+ gst_pad_push_event (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+ event);
+ }
+ break;
+ default:
+ /* FIXME this changes the order of events */
+ ret =
+ gst_pad_push_event (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+ event);
+ break;
+ }
+
+done:
+ gst_object_unref (base_audio_decoder);
+ return ret;
+
+newseg_wrong_format:
+ {
+ GST_DEBUG_OBJECT (base_audio_decoder, "received non TIME newsegment");
+ gst_event_unref (event);
+ goto done;
+ }
+}
+
+static gboolean
+gst_base_audio_decoder_src_event (GstPad * pad, GstEvent * event)
+{
+ GstBaseAudioDecoder *base_audio_decoder;
+ gboolean res = FALSE;
+
+ base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEEK:
+ {
+ GstFormat format, tformat;
+ gdouble rate;
+ GstEvent *real_seek;
+ GstSeekFlags flags;
+ GstSeekType cur_type, stop_type;
+ gint64 cur, stop;
+ gint64 tcur, tstop;
+
+ gst_event_parse_seek (event, &rate, &format, &flags, &cur_type,
+ &cur, &stop_type, &stop);
+ gst_event_unref (event);
+
+ tformat = GST_FORMAT_TIME;
+ res =
+ gst_base_audio_decoder_src_convert (pad, format, cur, &tformat,
+ &tcur);
+ if (!res)
+ goto convert_error;
+ res =
+ gst_base_audio_decoder_src_convert (pad, format, stop, &tformat,
+ &tstop);
+ if (!res)
+ goto convert_error;
+
+ real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
+ flags, cur_type, tcur, stop_type, tstop);
+
+ res =
+ gst_pad_push_event (GST_BASE_AUDIO_CODEC_SINK_PAD
+ (base_audio_decoder), real_seek);
+
+ break;
+ }
+ case GST_EVENT_QOS:
+ {
+ gdouble proportion;
+ GstClockTimeDiff diff;
+ GstClockTime timestamp;
+
+ gst_event_parse_qos (event, &proportion, &diff, ×tamp);
+
+ GST_OBJECT_LOCK (base_audio_decoder);
+ base_audio_decoder->proportion = proportion;
+ base_audio_decoder->earliest_time = timestamp + diff;
+ GST_OBJECT_UNLOCK (base_audio_decoder);
+
+ GST_DEBUG_OBJECT (base_audio_decoder,
+ "got QoS %" GST_TIME_FORMAT ", %" G_GINT64_FORMAT ", %g",
+ GST_TIME_ARGS (timestamp), diff, proportion);
+
+ res =
+ gst_pad_push_event (GST_BASE_AUDIO_CODEC_SINK_PAD
+ (base_audio_decoder), event);
+ break;
+ }
+ default:
+ res =
+ gst_pad_push_event (GST_BASE_AUDIO_CODEC_SINK_PAD
+ (base_audio_decoder), event);
+ break;
+ }
+done:
+ gst_object_unref (base_audio_decoder);
+ return res;
+
+convert_error:
+ GST_DEBUG_OBJECT (base_audio_decoder, "could not convert format");
+ goto done;
+}
+
+
+#if 0
+static gboolean
+gst_base_audio_decoder_sink_convert (GstPad * pad,
+ GstFormat src_format, gint64 src_value,
+ GstFormat * dest_format, gint64 * dest_value)
+{
+ gboolean res = TRUE;
+ GstBaseAudioDecoder *enc;
+
+ if (src_format == *dest_format) {
+ *dest_value = src_value;
+ return TRUE;
+ }
+
+ enc = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ /* FIXME: check if we are in a decoding state */
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+#if 0
+ case GST_FORMAT_DEFAULT:
+ *dest_value = gst_util_uint64_scale_int (src_value, 1,
+ enc->bytes_per_picture);
+ break;
+#endif
+ case GST_FORMAT_TIME:
+ /* seems like a rather silly conversion, implement me if you like */
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_DEFAULT:
+ switch (*dest_format) {
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale (src_value,
+ GST_SECOND * enc->fps_d, enc->fps_n);
+ break;
+#if 0
+ case GST_FORMAT_BYTES:
+ *dest_value = gst_util_uint64_scale_int (src_value,
+ enc->bytes_per_picture, 1);
+ break;
+#endif
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+}
+#endif
+
+static gboolean
+gst_base_audio_decoder_src_convert (GstPad * pad,
+ GstFormat src_format, gint64 src_value,
+ GstFormat * dest_format, gint64 * dest_value)
+{
+ gboolean res = TRUE;
+ GstBaseAudioDecoder *enc;
+
+ if (src_format == *dest_format) {
+ *dest_value = src_value;
+ return TRUE;
+ }
+
+ enc = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ /* FIXME: check if we are in a encoding state */
+
+ GST_DEBUG ("src convert");
+ switch (src_format) {
+#if 0
+ case GST_FORMAT_DEFAULT:
+ switch (*dest_format) {
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale (granulepos_to_frame (src_value),
+ enc->fps_d * GST_SECOND, enc->fps_n);
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_DEFAULT:
+ {
+ *dest_value = gst_util_uint64_scale (src_value,
+ enc->fps_n, enc->fps_d * GST_SECOND);
+ break;
+ }
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+#endif
+ default:
+ res = FALSE;
+ break;
+ }
+
+ gst_object_unref (enc);
+
+ return res;
+}
+
+static const GstQueryType *
+gst_base_audio_decoder_get_query_types (GstPad * pad)
+{
+ static const GstQueryType query_types[] = {
+ GST_QUERY_CONVERT,
+ 0
+ };
+
+ return query_types;
+}
+
+static gboolean
+gst_base_audio_decoder_src_query (GstPad * pad, GstQuery * query)
+{
+ GstBaseAudioDecoder *enc;
+ gboolean res;
+
+ enc = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ switch GST_QUERY_TYPE
+ (query) {
+ 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);
+ res =
+ gst_base_audio_decoder_src_convert (pad, src_fmt, src_val, &dest_fmt,
+ &dest_val);
+ if (!res)
+ goto error;
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ }
+ gst_object_unref (enc);
+ return res;
+
+error:
+ GST_DEBUG_OBJECT (enc, "query failed");
+ gst_object_unref (enc);
+ return res;
+}
+
+static gboolean
+gst_base_audio_decoder_sink_query (GstPad * pad, GstQuery * query)
+{
+ GstBaseAudioDecoder *base_audio_decoder;
+ gboolean res = FALSE;
+
+ base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+
+ GST_DEBUG_OBJECT (base_audio_decoder, "sink query fps=%d/%d",
+ base_audio_decoder->state.fps_n, base_audio_decoder->state.fps_d);
+ switch (GST_QUERY_TYPE (query)) {
+ 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);
+ res = gst_base_audio_rawaudio_convert (&base_audio_decoder->state,
+ src_fmt, src_val, &dest_fmt, &dest_val);
+ if (!res)
+ 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;
+ }
+done:
+ gst_object_unref (base_audio_decoder);
+
+ return res;
+error:
+ GST_DEBUG_OBJECT (base_audio_decoder, "query failed");
+ goto done;
+}
+
+
+#if 0
+static gboolean
+gst_pad_is_negotiated (GstPad * pad)
+{
+ GstCaps *caps;
+
+ g_return_val_if_fail (pad != NULL, FALSE);
+
+ caps = gst_pad_get_negotiated_caps (pad);
+ if (caps) {
+ gst_caps_unref (caps);
+ return TRUE;
+ }
+
+ return FALSE;
+}
+#endif
+
+static void
+gst_base_audio_decoder_reset (GstBaseAudioDecoder * base_audio_decoder)
+{
+ GstBaseAudioDecoderClass *base_audio_decoder_class;
+ GList *g;
+
+ base_audio_decoder_class =
+ GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+ GST_DEBUG ("reset");
+
+ base_audio_decoder->started = FALSE;
+
+ base_audio_decoder->discont = TRUE;
+ base_audio_decoder->have_sync = FALSE;
+
+ base_audio_decoder->timestamp_offset = GST_CLOCK_TIME_NONE;
+ base_audio_decoder->system_frame_number = 0;
+ base_audio_decoder->presentation_frame_number = 0;
+ base_audio_decoder->last_sink_timestamp = GST_CLOCK_TIME_NONE;
+ base_audio_decoder->last_sink_offset_end = GST_CLOCK_TIME_NONE;
+ base_audio_decoder->base_picture_number = 0;
+ base_audio_decoder->last_timestamp = GST_CLOCK_TIME_NONE;
+
+ base_audio_decoder->offset = 0;
+
+ if (base_audio_decoder->caps) {
+ gst_caps_unref (base_audio_decoder->caps);
+ base_audio_decoder->caps = NULL;
+ }
+
+ if (base_audio_decoder->current_frame) {
+ gst_base_audio_decoder_free_frame (base_audio_decoder->current_frame);
+ base_audio_decoder->current_frame = NULL;
+ }
+
+ base_audio_decoder->have_src_caps = FALSE;
+
+ for (g = g_list_first (base_audio_decoder->frames); g; g = g_list_next (g)) {
+ GstAudioFrame *frame = g->data;
+ gst_base_audio_decoder_free_frame (frame);
+ }
+ g_list_free (base_audio_decoder->frames);
+ base_audio_decoder->frames = NULL;
+
+ if (base_audio_decoder_class->reset) {
+ base_audio_decoder_class->reset (base_audio_decoder);
+ }
+}
+
+static GstBuffer *
+gst_adapter_get_buffer (GstAdapter * adapter)
+{
+ return gst_buffer_ref (GST_BUFFER (adapter->buflist->data));
+
+}
+
+static GstFlowReturn
+gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstBaseAudioDecoder *base_audio_decoder;
+ GstBaseAudioDecoderClass *klass;
+ GstBuffer *buffer;
+ GstFlowReturn ret;
+
+ GST_DEBUG ("chain %lld", GST_BUFFER_TIMESTAMP (buf));
+
+#if 0
+ /* requiring the pad to be negotiated makes it impossible to use
+ * oggdemux or filesrc ! decoder */
+ if (!gst_pad_is_negotiated (pad)) {
+ GST_DEBUG ("not negotiated");
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+#endif
+
+ base_audio_decoder = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
+ klass = GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+ GST_DEBUG_OBJECT (base_audio_decoder, "chain");
+
+ if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
+ GST_DEBUG_OBJECT (base_audio_decoder, "received DISCONT buffer");
+ if (base_audio_decoder->started) {
+ gst_base_audio_decoder_reset (base_audio_decoder);
+ }
+ }
+
+ if (!base_audio_decoder->started) {
+ klass->start (base_audio_decoder);
+ base_audio_decoder->started = TRUE;
+ }
+
+ if (GST_BUFFER_TIMESTAMP (buf) != GST_CLOCK_TIME_NONE) {
+ GST_DEBUG ("timestamp %lld offset %lld", GST_BUFFER_TIMESTAMP (buf),
+ base_audio_decoder->offset);
+ base_audio_decoder->last_sink_timestamp = GST_BUFFER_TIMESTAMP (buf);
+ }
+ if (GST_BUFFER_OFFSET_END (buf) != -1) {
+ GST_DEBUG ("gp %lld", GST_BUFFER_OFFSET_END (buf));
+ base_audio_decoder->last_sink_offset_end = GST_BUFFER_OFFSET_END (buf);
+ }
+ base_audio_decoder->offset += GST_BUFFER_SIZE (buf);
+
+#if 0
+ if (base_audio_decoder->timestamp_offset == GST_CLOCK_TIME_NONE &&
+ GST_BUFFER_TIMESTAMP (buf) != GST_CLOCK_TIME_NONE) {
+ GST_DEBUG ("got new offset %lld", GST_BUFFER_TIMESTAMP (buf));
+ base_audio_decoder->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
+ }
+#endif
+
+ if (base_audio_decoder->current_frame == NULL) {
+ base_audio_decoder->current_frame =
+ gst_base_audio_decoder_new_frame (base_audio_decoder);
+ }
+
+ gst_adapter_push (base_audio_decoder->input_adapter, buf);
+
+ if (!base_audio_decoder->have_sync) {
+ int n, m;
+
+ GST_DEBUG ("no sync, scanning");
+
+ n = gst_adapter_available (base_audio_decoder->input_adapter);
+ m = klass->scan_for_sync (base_audio_decoder, FALSE, 0, n);
+
+ if (m >= n) {
+ g_warning ("subclass scanned past end %d >= %d", m, n);
+ }
+
+ gst_adapter_flush (base_audio_decoder->input_adapter, m);
+
+ if (m < n) {
+ GST_DEBUG ("found possible sync after %d bytes (of %d)", m, n);
+
+ /* this is only "maybe" sync */
+ base_audio_decoder->have_sync = TRUE;
+ }
+
+ if (!base_audio_decoder->have_sync) {
+ gst_object_unref (base_audio_decoder);
+ return GST_FLOW_OK;
+ }
+ }
+
+ /* FIXME: use gst_adapter_prev_timestamp() here instead? */
+ buffer = gst_adapter_get_buffer (base_audio_decoder->input_adapter);
+
+ base_audio_decoder->buffer_timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ gst_buffer_unref (buffer);
+
+ do {
+ ret = klass->parse_data (base_audio_decoder, FALSE);
+ } while (ret == GST_FLOW_OK);
+
+ if (ret == GST_BASE_AUDIO_DECODER_FLOW_NEED_DATA) {
+ gst_object_unref (base_audio_decoder);
+ return GST_FLOW_OK;
+ }
+
+ gst_object_unref (base_audio_decoder);
+ return ret;
+}
+
+static GstStateChangeReturn
+gst_base_audio_decoder_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstBaseAudioDecoder *base_audio_decoder;
+ GstBaseAudioDecoderClass *base_audio_decoder_class;
+ GstStateChangeReturn ret;
+
+ base_audio_decoder = GST_BASE_AUDIO_DECODER (element);
+ base_audio_decoder_class = GST_BASE_AUDIO_DECODER_GET_CLASS (element);
+
+ switch (transition) {
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ if (base_audio_decoder_class->stop) {
+ base_audio_decoder_class->stop (base_audio_decoder);
+ }
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+static void
+gst_base_audio_decoder_free_frame (GstAudioFrame * frame)
+{
+ g_return_if_fail (frame != NULL);
+
+ if (frame->sink_buffer) {
+ gst_buffer_unref (frame->sink_buffer);
+ }
+#if 0
+ if (frame->src_buffer) {
+ gst_buffer_unref (frame->src_buffer);
+ }
+#endif
+
+ g_free (frame);
+}
+
+static GstAudioFrame *
+gst_base_audio_decoder_new_frame (GstBaseAudioDecoder * base_audio_decoder)
+{
+ GstAudioFrame *frame;
+
+ frame = g_malloc0 (sizeof (GstAudioFrame));
+
+ frame->system_frame_number = base_audio_decoder->system_frame_number;
+ base_audio_decoder->system_frame_number++;
+
+ frame->decode_frame_number = frame->system_frame_number -
+ base_audio_decoder->reorder_depth;
+
+ frame->decode_timestamp = -1;
+ frame->presentation_timestamp = -1;
+ frame->presentation_duration = -1;
+ frame->n_fields = 2;
+
+ return frame;
+}
+
+GstFlowReturn
+gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * base_audio_decoder,
+ GstAudioFrame * frame)
+{
+ GstBaseAudioDecoderClass *base_audio_decoder_class;
+ GstBuffer *src_buffer;
+
+ GST_DEBUG ("finish frame");
+
+ base_audio_decoder_class =
+ GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+ GST_DEBUG ("finish frame sync=%d pts=%lld", frame->is_sync_point,
+ frame->presentation_timestamp);
+
+ if (frame->is_sync_point) {
+ if (GST_CLOCK_TIME_IS_VALID (frame->presentation_timestamp)) {
+ if (frame->presentation_timestamp != base_audio_decoder->timestamp_offset) {
+ GST_DEBUG ("sync timestamp %lld diff %lld",
+ frame->presentation_timestamp,
+ frame->presentation_timestamp -
+ base_audio_decoder->state.segment.start);
+ base_audio_decoder->timestamp_offset = frame->presentation_timestamp;
+ base_audio_decoder->field_index = 0;
+ } else {
+ /* This case is for one initial timestamp and no others, e.g.,
+ * filesrc ! decoder ! xvimagesink */
+ GST_WARNING ("sync timestamp didn't change, ignoring");
+ frame->presentation_timestamp = GST_CLOCK_TIME_NONE;
+ }
+ } else {
+ GST_WARNING ("sync point doesn't have timestamp");
+ if (GST_CLOCK_TIME_IS_VALID (base_audio_decoder->timestamp_offset)) {
+ GST_ERROR ("No base timestamp. Assuming frames start at 0");
+ base_audio_decoder->timestamp_offset = 0;
+ base_audio_decoder->field_index = 0;
+ }
+ }
+ }
+ frame->field_index = base_audio_decoder->field_index;
+ base_audio_decoder->field_index += frame->n_fields;
+
+ if (frame->presentation_timestamp == GST_CLOCK_TIME_NONE) {
+ frame->presentation_timestamp =
+ gst_base_audio_decoder_get_field_timestamp (base_audio_decoder,
+ frame->field_index);
+ frame->presentation_duration = GST_CLOCK_TIME_NONE;
+ frame->decode_timestamp =
+ gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+ frame->decode_frame_number);
+ }
+ if (frame->presentation_duration == GST_CLOCK_TIME_NONE) {
+ frame->presentation_duration =
+ gst_base_audio_decoder_get_field_timestamp (base_audio_decoder,
+ frame->field_index + frame->n_fields) - frame->presentation_timestamp;
+ }
+
+ if (GST_CLOCK_TIME_IS_VALID (base_audio_decoder->last_timestamp)) {
+ if (frame->presentation_timestamp < base_audio_decoder->last_timestamp) {
+ GST_WARNING ("decreasing timestamp (%lld < %lld)",
+ frame->presentation_timestamp, base_audio_decoder->last_timestamp);
+ }
+ }
+ base_audio_decoder->last_timestamp = frame->presentation_timestamp;
+
+ GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_BUFFER_FLAG_DELTA_UNIT);
+ if (base_audio_decoder->state.interlaced) {
+#ifndef GST_AUDIO_BUFFER_TFF
+#define GST_AUDIO_BUFFER_TFF (GST_MINI_OBJECT_FLAG_LAST << 5)
+#endif
+#ifndef GST_AUDIO_BUFFER_RFF
+#define GST_AUDIO_BUFFER_RFF (GST_MINI_OBJECT_FLAG_LAST << 6)
+#endif
+#ifndef GST_AUDIO_BUFFER_ONEFIELD
+#define GST_AUDIO_BUFFER_ONEFIELD (GST_MINI_OBJECT_FLAG_LAST << 7)
+#endif
+ int tff = base_audio_decoder->state.top_field_first;
+
+ if (frame->field_index & 1) {
+ tff ^= 1;
+ }
+ if (tff) {
+ GST_BUFFER_FLAG_SET (frame->src_buffer, GST_AUDIO_BUFFER_TFF);
+ } else {
+ GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_TFF);
+ }
+ GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_RFF);
+ GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_ONEFIELD);
+ if (frame->n_fields == 3) {
+ GST_BUFFER_FLAG_SET (frame->src_buffer, GST_AUDIO_BUFFER_RFF);
+ } else if (frame->n_fields == 1) {
+ GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_AUDIO_BUFFER_ONEFIELD);
+ }
+ }
+
+ GST_BUFFER_TIMESTAMP (frame->src_buffer) = frame->presentation_timestamp;
+ GST_BUFFER_DURATION (frame->src_buffer) = frame->presentation_duration;
+ GST_BUFFER_OFFSET (frame->src_buffer) = -1;
+ GST_BUFFER_OFFSET_END (frame->src_buffer) = -1;
+
+ GST_DEBUG ("pushing frame %lld", frame->presentation_timestamp);
+
+ base_audio_decoder->frames =
+ g_list_remove (base_audio_decoder->frames, frame);
+
+ gst_base_audio_decoder_set_src_caps (base_audio_decoder);
+
+ src_buffer = frame->src_buffer;
+ frame->src_buffer = NULL;
+
+ gst_base_audio_decoder_free_frame (frame);
+
+ if (base_audio_decoder->sink_clipping) {
+ gint64 start = GST_BUFFER_TIMESTAMP (src_buffer);
+ gint64 stop = GST_BUFFER_TIMESTAMP (src_buffer) +
+ GST_BUFFER_DURATION (src_buffer);
+
+ if (gst_segment_clip (&base_audio_decoder->state.segment, GST_FORMAT_TIME,
+ start, stop, &start, &stop)) {
+ GST_BUFFER_TIMESTAMP (src_buffer) = start;
+ GST_BUFFER_DURATION (src_buffer) = stop - start;
+ } else {
+ GST_DEBUG ("dropping buffer outside segment");
+ gst_buffer_unref (src_buffer);
+ return GST_FLOW_OK;
+ }
+ }
+
+ return gst_pad_push (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+ src_buffer);
+}
+
+int
+gst_base_audio_decoder_get_height (GstBaseAudioDecoder * base_audio_decoder)
+{
+ return base_audio_decoder->state.height;
+}
+
+int
+gst_base_audio_decoder_get_width (GstBaseAudioDecoder * base_audio_decoder)
+{
+ return base_audio_decoder->state.width;
+}
+
+GstFlowReturn
+gst_base_audio_decoder_end_of_stream (GstBaseAudioDecoder * base_audio_decoder,
+ GstBuffer * buffer)
+{
+
+ if (base_audio_decoder->frames) {
+ GST_DEBUG ("EOS with frames left over");
+ }
+
+ return gst_pad_push (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder),
+ buffer);
+}
+
+void
+gst_base_audio_decoder_add_to_frame (GstBaseAudioDecoder * base_audio_decoder,
+ int n_bytes)
+{
+ GstBuffer *buf;
+
+ GST_DEBUG ("add to frame");
+
+#if 0
+ if (gst_adapter_available (base_audio_decoder->output_adapter) == 0) {
+ GstBuffer *buffer;
+
+ buffer =
+ gst_adapter_get_orig_buffer_at_offset
+ (base_audio_decoder->input_adapter, 0);
+ if (buffer) {
+ base_audio_decoder->current_frame->presentation_timestamp =
+ GST_BUFFER_TIMESTAMP (buffer);
+ gst_buffer_unref (buffer);
+ }
+ }
+#endif
+
+ if (n_bytes == 0)
+ return;
+
+ buf = gst_adapter_take_buffer (base_audio_decoder->input_adapter, n_bytes);
+
+ gst_adapter_push (base_audio_decoder->output_adapter, buf);
+}
+
+static guint64
+gst_base_audio_decoder_get_timestamp (GstBaseAudioDecoder * base_audio_decoder,
+ int picture_number)
+{
+ if (base_audio_decoder->state.fps_d == 0) {
+ return -1;
+ }
+ if (picture_number < base_audio_decoder->base_picture_number) {
+ return base_audio_decoder->timestamp_offset -
+ (gint64) gst_util_uint64_scale (base_audio_decoder->base_picture_number
+ - picture_number, base_audio_decoder->state.fps_d * GST_SECOND,
+ base_audio_decoder->state.fps_n);
+ } else {
+ return base_audio_decoder->timestamp_offset +
+ gst_util_uint64_scale (picture_number -
+ base_audio_decoder->base_picture_number,
+ base_audio_decoder->state.fps_d * GST_SECOND,
+ base_audio_decoder->state.fps_n);
+ }
+}
+
+static guint64
+gst_base_audio_decoder_get_field_timestamp (GstBaseAudioDecoder *
+ base_audio_decoder, int field_offset)
+{
+ if (base_audio_decoder->state.fps_d == 0) {
+ return GST_CLOCK_TIME_NONE;
+ }
+ if (field_offset < 0) {
+ GST_WARNING ("field offset < 0");
+ return GST_CLOCK_TIME_NONE;
+ }
+ return base_audio_decoder->timestamp_offset +
+ gst_util_uint64_scale (field_offset,
+ base_audio_decoder->state.fps_d * GST_SECOND,
+ base_audio_decoder->state.fps_n * 2);
+}
+
+
+GstFlowReturn
+gst_base_audio_decoder_have_frame (GstBaseAudioDecoder * base_audio_decoder)
+{
+ GstAudioFrame *frame = base_audio_decoder->current_frame;
+ GstBuffer *buffer;
+ GstBaseAudioDecoderClass *base_audio_decoder_class;
+ GstFlowReturn ret = GST_FLOW_OK;
+ int n_available;
+
+ GST_DEBUG ("have_frame");
+
+ base_audio_decoder_class =
+ GST_BASE_AUDIO_DECODER_GET_CLASS (base_audio_decoder);
+
+ n_available = gst_adapter_available (base_audio_decoder->output_adapter);
+ if (n_available) {
+ buffer = gst_adapter_take_buffer (base_audio_decoder->output_adapter,
+ n_available);
+ } else {
+ buffer = gst_buffer_new_and_alloc (0);
+ }
+
+ frame->distance_from_sync = base_audio_decoder->distance_from_sync;
+ base_audio_decoder->distance_from_sync++;
+
+#if 0
+ if (frame->presentation_timestamp == GST_CLOCK_TIME_NONE) {
+ frame->presentation_timestamp =
+ gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+ frame->presentation_frame_number);
+ frame->presentation_duration =
+ gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+ frame->presentation_frame_number + 1) - frame->presentation_timestamp;
+ frame->decode_timestamp =
+ gst_base_audio_decoder_get_timestamp (base_audio_decoder,
+ frame->decode_frame_number);
+ }
+#endif
+
+#if 0
+ GST_BUFFER_TIMESTAMP (buffer) = frame->presentation_timestamp;
+ GST_BUFFER_DURATION (buffer) = frame->presentation_duration;
+ if (frame->decode_frame_number < 0) {
+ GST_BUFFER_OFFSET (buffer) = 0;
+ } else {
+ GST_BUFFER_OFFSET (buffer) = frame->decode_timestamp;
+ }
+ GST_BUFFER_OFFSET_END (buffer) = GST_CLOCK_TIME_NONE;
+#endif
+
+ GST_DEBUG ("pts %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (frame->presentation_timestamp));
+ GST_DEBUG ("dts %" GST_TIME_FORMAT, GST_TIME_ARGS (frame->decode_timestamp));
+ GST_DEBUG ("dist %d", frame->distance_from_sync);
+
+ if (frame->is_sync_point) {
+ GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DELTA_UNIT);
+ } else {
+ GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT);
+ }
+ if (base_audio_decoder->discont) {
+ GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
+ base_audio_decoder->discont = FALSE;
+ }
+
+ frame->sink_buffer = buffer;
+
+ base_audio_decoder->frames = g_list_append (base_audio_decoder->frames,
+ frame);
+
+ /* do something with frame */
+ ret = base_audio_decoder_class->handle_frame (base_audio_decoder, frame);
+ if (!GST_FLOW_IS_SUCCESS (ret)) {
+ GST_DEBUG ("flow error!");
+ }
+
+ /* create new frame */
+ base_audio_decoder->current_frame =
+ gst_base_audio_decoder_new_frame (base_audio_decoder);
+
+ return ret;
+}
+
+GstAudioState *
+gst_base_audio_decoder_get_state (GstBaseAudioDecoder * base_audio_decoder)
+{
+ return &base_audio_decoder->state;
+
+}
+
+void
+gst_base_audio_decoder_set_state (GstBaseAudioDecoder * base_audio_decoder,
+ GstAudioState * state)
+{
+ memcpy (&base_audio_decoder->state, state, sizeof (*state));
+
+}
+
+void
+gst_base_audio_decoder_lost_sync (GstBaseAudioDecoder * base_audio_decoder)
+{
+ g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (base_audio_decoder));
+
+ GST_DEBUG ("lost_sync");
+
+ if (gst_adapter_available (base_audio_decoder->input_adapter) >= 1) {
+ gst_adapter_flush (base_audio_decoder->input_adapter, 1);
+ }
+
+ base_audio_decoder->have_sync = FALSE;
+}
+
+void
+gst_base_audio_decoder_set_sync_point (GstBaseAudioDecoder * base_audio_decoder)
+{
+ GST_DEBUG ("set_sync_point");
+
+ base_audio_decoder->current_frame->is_sync_point = TRUE;
+ base_audio_decoder->distance_from_sync = 0;
+
+ base_audio_decoder->current_frame->presentation_timestamp =
+ base_audio_decoder->last_sink_timestamp;
+
+
+}
+
+GstAudioFrame *
+gst_base_audio_decoder_get_frame (GstBaseAudioDecoder * base_audio_decoder,
+ int frame_number)
+{
+ GList *g;
+
+ for (g = g_list_first (base_audio_decoder->frames); g; g = g_list_next (g)) {
+ GstAudioFrame *frame = g->data;
+
+ if (frame->system_frame_number == frame_number) {
+ return frame;
+ }
+ }
+
+ return NULL;
+}
+
+void
+gst_base_audio_decoder_set_src_caps (GstBaseAudioDecoder * base_audio_decoder)
+{
+ GstCaps *caps;
+ GstAudioState *state = &base_audio_decoder->state;
+
+ if (base_audio_decoder->have_src_caps)
+ return;
+
+ caps = gst_audio_format_new_caps (state->format,
+ state->width, state->height,
+ state->fps_n, state->fps_d, state->par_n, state->par_d);
+ gst_caps_set_simple (caps, "interlaced",
+ G_TYPE_BOOLEAN, state->interlaced, NULL);
+
+ GST_DEBUG ("setting caps %" GST_PTR_FORMAT, caps);
+
+ gst_pad_set_caps (GST_BASE_AUDIO_CODEC_SRC_PAD (base_audio_decoder), caps);
+
+ base_audio_decoder->have_src_caps = TRUE;
+}
projects / platform / upstream / gstreamer.git / commitdiff