mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 10:11:08 +00:00
wavpackdec: port to audiodecoder
This commit is contained in:
Mark Nauwelaerts
parent
b863df570f
commit
004377b0b5
2 changed files with 110 additions and 198 deletions
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@ -55,8 +55,6 @@
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#include "gstwavpackstreamreader.h"
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#define WAVPACK_DEC_MAX_ERRORS 16
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GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
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#define GST_CAT_DEFAULT gst_wavpack_dec_debug
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@ -80,15 +78,18 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
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);
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static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
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static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
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static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
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static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec);
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static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec,
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GstCaps * caps);
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static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void gst_wavpack_dec_finalize (GObject * object);
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static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
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GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
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GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER);
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static void
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gst_wavpack_dec_base_init (gpointer klass)
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@ -108,11 +109,14 @@ static void
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gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
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gobject_class->finalize = gst_wavpack_dec_finalize;
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base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame);
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}
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static void
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@ -121,33 +125,15 @@ gst_wavpack_dec_reset (GstWavpackDec * dec)
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dec->wv_id.buffer = NULL;
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dec->wv_id.position = dec->wv_id.length = 0;
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dec->error_count = 0;
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dec->channels = 0;
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dec->channel_mask = 0;
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dec->sample_rate = 0;
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dec->depth = 0;
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gst_segment_init (&dec->segment, GST_FORMAT_TIME);
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dec->next_block_index = 0;
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}
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static void
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gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
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{
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dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
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gst_pad_set_chain_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
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gst_pad_set_setcaps_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps));
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gst_pad_set_event_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
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gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
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dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
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gst_pad_use_fixed_caps (dec->srcpad);
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gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
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dec->context = NULL;
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dec->stream_reader = gst_wavpack_stream_reader_new ();
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@ -166,26 +152,73 @@ gst_wavpack_dec_finalize (GObject * object)
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}
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static gboolean
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gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
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gst_wavpack_dec_start (GstAudioDecoder * dec)
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{
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GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
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GST_DEBUG_OBJECT (dec, "start");
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/* never mind a few errors */
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gst_audio_decoder_set_max_errors (dec, 16);
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/* don't bother us with flushing */
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gst_audio_decoder_set_drainable (dec, FALSE);
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return TRUE;
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}
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static gboolean
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gst_wavpack_dec_stop (GstAudioDecoder * dec)
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{
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GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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if (wpdec->context) {
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WavpackCloseFile (wpdec->context);
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wpdec->context = NULL;
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}
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gst_wavpack_dec_reset (wpdec);
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return TRUE;
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}
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static void
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gst_wavpack_dec_negotiate (GstWavpackDec * dec)
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{
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GstCaps *caps;
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->sample_rate,
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"channels", G_TYPE_INT, dec->channels,
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"depth", G_TYPE_INT, dec->depth,
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"width", G_TYPE_INT, 32,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE, NULL);
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/* Only set the channel layout for more than two channels
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* otherwise things break unfortunately */
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if (dec->channel_mask != 0 && dec->channels > 2)
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if (!gst_wavpack_set_channel_layout (caps, dec->channel_mask))
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GST_WARNING_OBJECT (dec, "Failed to set channel layout");
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GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
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/* should always succeed */
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
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gst_caps_unref (caps);
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}
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static gboolean
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gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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GstWavpackDec *dec = GST_WAVPACK_DEC (bdec);
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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/* Check if we can set the caps here already */
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if (gst_structure_get_int (structure, "channels", &dec->channels) &&
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gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
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gst_structure_get_int (structure, "width", &dec->depth)) {
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GstCaps *caps;
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GstAudioChannelPosition *pos;
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->sample_rate,
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"channels", G_TYPE_INT, dec->channels,
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"depth", G_TYPE_INT, dec->depth,
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"width", G_TYPE_INT, 32,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE, NULL);
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/* If we already have the channel layout set from upstream
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* take this */
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if (gst_structure_has_field (structure, "channel-positions")) {
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@ -202,19 +235,13 @@ gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
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g_free (pos);
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}
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GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
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/* should always succeed */
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gst_pad_set_caps (dec->srcpad, caps);
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gst_caps_unref (caps);
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gst_wavpack_dec_negotiate (dec);
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/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
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* is decoded or after the format has changed */
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gst_wavpack_dec_post_tags (dec);
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}
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gst_object_unref (dec);
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return TRUE;
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}
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@ -225,28 +252,26 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec)
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GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
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gint64 duration, size;
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list = gst_tag_list_new ();
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_AUDIO_CODEC, "Wavpack", NULL);
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/* try to estimate the average bitrate */
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if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) &&
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gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
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size > 0 && duration > 0) {
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if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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&format_bytes, &size) &&
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gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
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&format_time, &duration) && size > 0 && duration > 0) {
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guint64 bitrate;
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list = gst_tag_list_new ();
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bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
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(guint) bitrate, NULL);
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}
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gst_element_post_message (GST_ELEMENT (dec),
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gst_message_new_tag (GST_OBJECT (dec), list));
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gst_element_post_message (GST_ELEMENT (dec),
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gst_message_new_tag (GST_OBJECT (dec), list));
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}
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}
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static GstFlowReturn
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gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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{
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GstWavpackDec *dec;
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GstBuffer *outbuf = NULL;
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@ -255,7 +280,9 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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int32_t decoded, unpacked_size;
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gboolean format_changed;
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dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
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dec = GST_WAVPACK_DEC (bdec);
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g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
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/* check input, we only accept framed input with complete chunks */
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if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
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@ -283,21 +310,15 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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dec->context = WavpackOpenFileInputEx (dec->stream_reader,
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&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
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/* expect this to work */
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if (!dec->context) {
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GST_WARNING ("Couldn't decode buffer: %s", error_msg);
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dec->error_count++;
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if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
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goto out; /* just return OK for now */
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} else {
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goto decode_error;
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}
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GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
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goto context_failed;
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}
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}
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g_assert (dec->context != NULL);
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dec->error_count = 0;
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format_changed =
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(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
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(dec->channels != WavpackGetNumChannels (dec->context)) ||
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@ -308,22 +329,13 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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(dec->channel_mask != WavpackGetChannelMask (dec->context));
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#endif
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if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
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GstCaps *caps;
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if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) {
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gint channel_mask;
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dec->sample_rate = WavpackGetSampleRate (dec->context);
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dec->channels = WavpackGetNumChannels (dec->context);
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dec->depth = WavpackGetBitsPerSample (dec->context);
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->sample_rate,
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"channels", G_TYPE_INT, dec->channels,
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"depth", G_TYPE_INT, dec->depth,
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"width", G_TYPE_INT, 32,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE, NULL);
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#ifdef WAVPACK_OLD_API
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channel_mask = dec->context->config.channel_mask;
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#else
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@ -334,17 +346,7 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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dec->channel_mask = channel_mask;
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/* Only set the channel layout for more than two channels
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* otherwise things break unfortunately */
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if (channel_mask != 0 && dec->channels > 2)
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if (!gst_wavpack_set_channel_layout (caps, channel_mask))
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GST_WARNING_OBJECT (dec, "Failed to set channel layout");
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GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
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/* should always succeed */
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gst_pad_set_caps (dec->srcpad, caps);
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gst_caps_unref (caps);
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gst_wavpack_dec_negotiate (dec);
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/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
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* is decoded or after the format has changed */
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@ -353,34 +355,20 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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/* alloc output buffer */
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unpacked_size = 4 * wph.block_samples * dec->channels;
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ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
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unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
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ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec),
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GST_BUFFER_OFFSET (buf), unpacked_size,
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GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
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if (ret != GST_FLOW_OK)
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goto out;
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gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
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/* If we got a DISCONT buffer forward the flag. Nothing else
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* has to be done as libwavpack doesn't store state between
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* Wavpack blocks */
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if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index)
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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dec->next_block_index = wph.block_index + wph.block_samples;
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/* decode */
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decoded = WavpackUnpackSamples (dec->context,
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(int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
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if (decoded != wph.block_samples)
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goto decode_error;
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if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
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dec->sample_rate, 4 * dec->channels))) {
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GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
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ret = gst_pad_push (dec->srcpad, outbuf);
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}
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ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
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out:
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@ -388,23 +376,25 @@ out:
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GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
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}
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gst_buffer_unref (buf);
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return ret;
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/* ERRORS */
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input_not_framed:
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{
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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}
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invalid_header:
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{
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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}
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context_failed:
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{
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GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
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("error creating Wavpack context"), ret);
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goto out;
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}
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decode_error:
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{
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const gchar *reason = "unknown";
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@ -418,88 +408,16 @@ decode_error:
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} else {
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reason = "couldn't create decoder context";
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}
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
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("Failed to decode wavpack stream: %s", reason));
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
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("decoding error: %s", reason), ret);
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if (outbuf)
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gst_buffer_unref (outbuf);
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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if (ret == GST_FLOW_OK)
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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return ret;
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}
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}
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static gboolean
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gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
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{
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GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
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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;
|
||||
}
|
||||
|
||||
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
|
||||
gst_wavpack_dec_plugin_init (GstPlugin * plugin)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -24,6 +24,7 @@
|
|||
#define __GST_WAVPACK_DEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
|
||||
#include <wavpack/wavpack.h>
|
||||
|
||||
|
|
@ -45,31 +46,24 @@ typedef struct _GstWavpackDecClass GstWavpackDecClass;
|
|||
|
||||
struct _GstWavpackDec
|
||||
{
|
||||
GstElement element;
|
||||
GstAudioDecoder element;
|
||||
|
||||
/*< private > */
|
||||
GstPad *sinkpad;
|
||||
GstPad *srcpad;
|
||||
|
||||
WavpackContext *context;
|
||||
WavpackStreamReader *stream_reader;
|
||||
|
||||
read_id wv_id;
|
||||
|
||||
GstSegment segment; /* used for clipping, TIME format */
|
||||
guint32 next_block_index;
|
||||
|
||||
gint sample_rate;
|
||||
gint depth;
|
||||
gint channels;
|
||||
gint channel_mask;
|
||||
|
||||
gint error_count;
|
||||
};
|
||||
|
||||
struct _GstWavpackDecClass
|
||||
{
|
||||
GstElementClass parent;
|
||||
GstAudioDecoderClass parent;
|
||||
};
|
||||
|
||||
GType gst_wavpack_dec_get_type (void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue