#include "gstwavpackstreamreader.h"
-#define WAVPACK_DEC_MAX_ERRORS 16
-
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
#define GST_CAT_DEFAULT gst_wavpack_dec_debug
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
);
-static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
-static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
-static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
+static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
+static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec);
+static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec,
+ GstCaps * caps);
+static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer);
+
static void gst_wavpack_dec_finalize (GObject * object);
-static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
- GstStateChange transition);
static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
-GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
+GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder,
+ GST_TYPE_AUDIO_DECODER);
static void
gst_wavpack_dec_base_init (gpointer klass)
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
- GstElementClass *gstelement_class = (GstElementClass *) klass;
+ GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
gobject_class->finalize = gst_wavpack_dec_finalize;
+
+ base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame);
}
static void
dec->wv_id.buffer = NULL;
dec->wv_id.position = dec->wv_id.length = 0;
- dec->error_count = 0;
-
dec->channels = 0;
dec->channel_mask = 0;
dec->sample_rate = 0;
dec->depth = 0;
-
- gst_segment_init (&dec->segment, GST_FORMAT_TIME);
- dec->next_block_index = 0;
}
static void
gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
{
- dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
- gst_pad_set_chain_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
- gst_pad_set_setcaps_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps));
- gst_pad_set_event_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
- gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
-
- dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_pad_use_fixed_caps (dec->srcpad);
- gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
-
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
}
static gboolean
-gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
+gst_wavpack_dec_start (GstAudioDecoder * dec)
+{
+ GST_DEBUG_OBJECT (dec, "start");
+
+ /* never mind a few errors */
+ gst_audio_decoder_set_max_errors (dec, 16);
+ /* don't bother us with flushing */
+ gst_audio_decoder_set_drainable (dec, FALSE);
+
+ return TRUE;
+}
+
+static gboolean
+gst_wavpack_dec_stop (GstAudioDecoder * dec)
{
- GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
+ GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
+
+ GST_DEBUG_OBJECT (dec, "stop");
+
+ if (wpdec->context) {
+ WavpackCloseFile (wpdec->context);
+ wpdec->context = NULL;
+ }
+
+ gst_wavpack_dec_reset (wpdec);
+
+ return TRUE;
+}
+
+static void
+gst_wavpack_dec_negotiate (GstWavpackDec * dec)
+{
+ GstCaps *caps;
+
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "rate", G_TYPE_INT, dec->sample_rate,
+ "channels", G_TYPE_INT, dec->channels,
+ "depth", G_TYPE_INT, dec->depth,
+ "width", G_TYPE_INT, 32,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* Only set the channel layout for more than two channels
+ * otherwise things break unfortunately */
+ if (dec->channel_mask != 0 && dec->channels > 2)
+ if (!gst_wavpack_set_channel_layout (caps, dec->channel_mask))
+ GST_WARNING_OBJECT (dec, "Failed to set channel layout");
+
+ GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
+
+ /* should always succeed */
+ gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
+ gst_caps_unref (caps);
+}
+
+static gboolean
+gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
+{
+ GstWavpackDec *dec = GST_WAVPACK_DEC (bdec);
GstStructure *structure = gst_caps_get_structure (caps, 0);
/* Check if we can set the caps here already */
if (gst_structure_get_int (structure, "channels", &dec->channels) &&
gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
gst_structure_get_int (structure, "width", &dec->depth)) {
- GstCaps *caps;
GstAudioChannelPosition *pos;
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, dec->sample_rate,
- "channels", G_TYPE_INT, dec->channels,
- "depth", G_TYPE_INT, dec->depth,
- "width", G_TYPE_INT, 32,
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "signed", G_TYPE_BOOLEAN, TRUE, NULL);
-
/* If we already have the channel layout set from upstream
* take this */
if (gst_structure_has_field (structure, "channel-positions")) {
g_free (pos);
}
- GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
-
- /* should always succeed */
- gst_pad_set_caps (dec->srcpad, caps);
- gst_caps_unref (caps);
+ gst_wavpack_dec_negotiate (dec);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
- gst_object_unref (dec);
-
return TRUE;
}
GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
gint64 duration, size;
- list = gst_tag_list_new ();
-
- gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
- GST_TAG_AUDIO_CODEC, "Wavpack", NULL);
-
/* try to estimate the average bitrate */
- if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) &&
- gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
- size > 0 && duration > 0) {
+ if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
+ &format_bytes, &size) &&
+ gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
+ &format_time, &duration) && size > 0 && duration > 0) {
guint64 bitrate;
+ list = gst_tag_list_new ();
+
bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
(guint) bitrate, NULL);
- }
- gst_element_post_message (GST_ELEMENT (dec),
- gst_message_new_tag (GST_OBJECT (dec), list));
+ gst_element_post_message (GST_ELEMENT (dec),
+ gst_message_new_tag (GST_OBJECT (dec), list));
+ }
}
static GstFlowReturn
-gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
+gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf = NULL;
int32_t decoded, unpacked_size;
gboolean format_changed;
- dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
+ dec = GST_WAVPACK_DEC (bdec);
+
+ g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
+ /* expect this to work */
if (!dec->context) {
- GST_WARNING ("Couldn't decode buffer: %s", error_msg);
- dec->error_count++;
- if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
- goto out; /* just return OK for now */
- } else {
- goto decode_error;
- }
+ GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
+ goto context_failed;
}
}
g_assert (dec->context != NULL);
- dec->error_count = 0;
-
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->channel_mask != WavpackGetChannelMask (dec->context));
#endif
- if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
- GstCaps *caps;
+ if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) {
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBitsPerSample (dec->context);
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, dec->sample_rate,
- "channels", G_TYPE_INT, dec->channels,
- "depth", G_TYPE_INT, dec->depth,
- "width", G_TYPE_INT, 32,
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "signed", G_TYPE_BOOLEAN, TRUE, NULL);
-
#ifdef WAVPACK_OLD_API
channel_mask = dec->context->config.channel_mask;
#else
dec->channel_mask = channel_mask;
- /* Only set the channel layout for more than two channels
- * otherwise things break unfortunately */
- if (channel_mask != 0 && dec->channels > 2)
- if (!gst_wavpack_set_channel_layout (caps, channel_mask))
- GST_WARNING_OBJECT (dec, "Failed to set channel layout");
-
- GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
-
- /* should always succeed */
- gst_pad_set_caps (dec->srcpad, caps);
- gst_caps_unref (caps);
+ gst_wavpack_dec_negotiate (dec);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
/* alloc output buffer */
unpacked_size = 4 * wph.block_samples * dec->channels;
- ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
- unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
+ ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec),
+ GST_BUFFER_OFFSET (buf), unpacked_size,
+ GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
- gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
-
- /* If we got a DISCONT buffer forward the flag. Nothing else
- * has to be done as libwavpack doesn't store state between
- * Wavpack blocks */
- if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index)
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
-
- dec->next_block_index = wph.block_index + wph.block_samples;
-
/* decode */
decoded = WavpackUnpackSamples (dec->context,
(int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
- if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
- dec->sample_rate, 4 * dec->channels))) {
- GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
- ret = gst_pad_push (dec->srcpad, outbuf);
- }
+ ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out:
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
- gst_buffer_unref (buf);
-
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
- gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
- gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
+context_failed:
+ {
+ GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
+ ("error creating Wavpack context"), ret);
+ goto out;
+ }
decode_error:
{
const gchar *reason = "unknown";
} else {
reason = "couldn't create decoder context";
}
- GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
- ("Failed to decode wavpack stream: %s", reason));
+ GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
+ ("decoding error: %s", reason), ret);
if (outbuf)
gst_buffer_unref (outbuf);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
-}
-
-static gboolean
-gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
-{
- GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
-
- GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_NEWSEGMENT:{
- GstFormat fmt;
- gboolean is_update;
- gint64 start, end, base;
- gdouble rate;
-
- gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
- &end, &base);
- if (fmt == GST_FORMAT_TIME) {
- GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
- GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
- GST_TIME_ARGS (end));
- gst_segment_set_newsegment (&dec->segment, is_update, rate, fmt,
- start, end, base);
- } else {
- gst_segment_init (&dec->segment, GST_FORMAT_TIME);
- }
- break;
- }
- default:
- break;
+ if (ret == GST_FLOW_OK)
+ gst_audio_decoder_finish_frame (bdec, NULL, 1);
+ return ret;
}
-
- gst_object_unref (dec);
- return gst_pad_event_default (pad, event);
-}
-
-static GstStateChangeReturn
-gst_wavpack_dec_change_state (GstElement * element, GstStateChange transition)
-{
- GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
- GstWavpackDec *dec = GST_WAVPACK_DEC (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- break;
- default:
- break;
- }
-
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- switch (transition) {
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- if (dec->context) {
- WavpackCloseFile (dec->context);
- dec->context = NULL;
- }
-
- gst_wavpack_dec_reset (dec);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
- }
-
- return ret;
}
gboolean
projects / platform / upstream / gst-plugins-good.git / commitdiff