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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-23 06:26:23 +00:00
adpcmdec: port to audiodecoder
This commit is contained in:
Mark Nauwelaerts
parent
1a73bf0b79
commit
dbc8bbd755
2 changed files with 96 additions and 180 deletions
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@ -5,8 +5,9 @@ libgstadpcmdec_la_SOURCES = adpcmdec.c
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# flags used to compile this plugin
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# add other _CFLAGS and _LIBS as needed
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libgstadpcmdec_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
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libgstadpcmdec_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
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libgstadpcmdec_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
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libgstadpcmdec_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-@GST_MAJORMINOR@ \
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$(GST_LIBS)
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libgstadpcmdec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstadpcmdec_la_LIBTOOLFLAGS = --tag=disable-static
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@ -28,7 +28,7 @@
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/gstaudiodecoder.h>
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#define GST_TYPE_ADPCM_DEC \
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(adpcmdec_get_type ())
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@ -69,80 +69,29 @@ enum adpcm_layout
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typedef struct _ADPCMDecClass
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{
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GstElementClass parent_class;
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GstAudioDecoderClass parent_class;
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} ADPCMDecClass;
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typedef struct _ADPCMDec
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{
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GstElement parent;
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GstPad *sinkpad;
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GstPad *srcpad;
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GstCaps *output_caps;
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GstAudioDecoder parent;
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enum adpcm_layout layout;
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int rate;
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int channels;
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int blocksize;
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gboolean is_setup;
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GstClockTime timestamp;
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GstClockTime base_timestamp;
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guint64 out_samples;
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GstAdapter *adapter;
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} ADPCMDec;
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GType adpcmdec_get_type (void);
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GST_BOILERPLATE (ADPCMDec, adpcmdec, GstElement, GST_TYPE_ELEMENT);
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static gboolean
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adpcmdec_setup (ADPCMDec * dec)
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{
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dec->output_caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->rate,
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"channels", G_TYPE_INT, dec->channels,
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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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 (dec->output_caps) {
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gst_pad_set_caps (dec->srcpad, dec->output_caps);
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}
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dec->is_setup = TRUE;
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dec->timestamp = GST_CLOCK_TIME_NONE;
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dec->base_timestamp = GST_CLOCK_TIME_NONE;
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dec->adapter = gst_adapter_new ();
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dec->out_samples = 0;
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return TRUE;
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}
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static void
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adpcmdec_teardown (ADPCMDec * dec)
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{
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if (dec->output_caps) {
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gst_caps_unref (dec->output_caps);
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dec->output_caps = NULL;
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}
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if (dec->adapter) {
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g_object_unref (dec->adapter);
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dec->adapter = NULL;
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}
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dec->is_setup = FALSE;
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}
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GST_BOILERPLATE (ADPCMDec, adpcmdec, GstAudioDecoder, GST_TYPE_AUDIO_DECODER);
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static gboolean
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adpcmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
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adpcmdec_set_format (GstAudioDecoder * bdec, GstCaps * in_caps)
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{
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ADPCMDec *dec = (ADPCMDec *) gst_pad_get_parent (pad);
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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ADPCMDec *dec = (ADPCMDec *) (bdec);
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GstStructure *structure = gst_caps_get_structure (in_caps, 0);
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const gchar *layout;
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GstCaps *caps;
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layout = gst_structure_get_string (structure, "layout");
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if (!layout)
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@ -163,9 +112,16 @@ adpcmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
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if (!gst_structure_get_int (structure, "channels", &dec->channels))
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return FALSE;
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if (dec->is_setup)
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adpcmdec_teardown (dec);
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gst_object_unref (dec);
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->rate,
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"channels", G_TYPE_INT, dec->channels,
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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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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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (bdec), caps);
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gst_caps_unref (caps);
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return TRUE;
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}
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@ -377,10 +333,10 @@ adpcmdec_decode_ima_block (ADPCMDec * dec, int n_samples, const guint8 * data,
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return TRUE;
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}
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static GstFlowReturn
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static GstBuffer *
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adpcmdec_decode_block (ADPCMDec * dec, const guint8 * data, int blocksize)
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{
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gboolean res;
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gboolean res = FALSE;
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GstBuffer *outbuf = NULL;
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int outsize;
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int samples;
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@ -390,7 +346,7 @@ adpcmdec_decode_block (ADPCMDec * dec, const guint8 * data, int blocksize)
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give two initial sample values per channel. Then the remainder gives
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two samples per byte */
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if (blocksize < 7 * dec->channels)
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return GST_FLOW_ERROR;
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goto exit;
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samples = (blocksize - 7 * dec->channels) * 2 + 2 * dec->channels;
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outsize = 2 * samples;
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outbuf = gst_buffer_new_and_alloc (outsize);
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@ -401,7 +357,7 @@ adpcmdec_decode_block (ADPCMDec * dec, const guint8 * data, int blocksize)
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/* Each block has a 4 byte header per channel, include an initial sample.
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Then the remainder gives two samples per byte */
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if (blocksize < 4 * dec->channels)
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return GST_FLOW_ERROR;
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goto exit;
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samples = (blocksize - 4 * dec->channels) * 2 + dec->channels;
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outsize = 2 * samples;
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outbuf = gst_buffer_new_and_alloc (outsize);
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@ -410,155 +366,114 @@ adpcmdec_decode_block (ADPCMDec * dec, const guint8 * data, int blocksize)
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(gint16 *) (GST_BUFFER_DATA (outbuf)));
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} else {
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GST_WARNING_OBJECT (dec, "Unknown layout");
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return GST_FLOW_ERROR;
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}
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if (!res) {
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gst_buffer_unref (outbuf);
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if (outbuf)
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gst_buffer_unref (outbuf);
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outbuf = NULL;
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GST_WARNING_OBJECT (dec, "Decode of block failed");
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return GST_FLOW_ERROR;
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}
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gst_buffer_set_caps (outbuf, dec->output_caps);
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GST_BUFFER_TIMESTAMP (outbuf) = dec->timestamp;
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dec->out_samples += samples / dec->channels;
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dec->timestamp = dec->base_timestamp +
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gst_util_uint64_scale_int (dec->out_samples, GST_SECOND, dec->rate);
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GST_BUFFER_DURATION (outbuf) = dec->timestamp - GST_BUFFER_TIMESTAMP (outbuf);
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return gst_pad_push (dec->srcpad, outbuf);
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exit:
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return outbuf;
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}
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static GstFlowReturn
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adpcmdec_chain (GstPad * pad, GstBuffer * buf)
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adpcmdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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ADPCMDec *dec = (ADPCMDec *) gst_pad_get_parent (pad);
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ADPCMDec *dec = (ADPCMDec *) (bdec);
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guint size;
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size = gst_adapter_available (adapter);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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if (dec->blocksize < 0) {
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/* No explicit blocksize; we just process one input buffer at a time */
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*offset = 0;
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*length = size;
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} else {
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if (size >= dec->blocksize) {
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*offset = 0;
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*length = dec->blocksize;
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} else {
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return GST_FLOW_UNEXPECTED;
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}
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}
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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adpcmdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
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{
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ADPCMDec *dec = (ADPCMDec *) (bdec);
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GstFlowReturn ret = GST_FLOW_OK;
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guint8 *data;
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GstBuffer *databuf = NULL;
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GstBuffer *outbuf = NULL;
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if (!dec->is_setup)
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adpcmdec_setup (dec);
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/* no fancy draining */
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if (G_UNLIKELY (!buffer))
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return GST_FLOW_OK;
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if (dec->base_timestamp == GST_CLOCK_TIME_NONE) {
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dec->base_timestamp = GST_BUFFER_TIMESTAMP (buf);
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if (dec->base_timestamp == GST_CLOCK_TIME_NONE)
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dec->base_timestamp = 0;
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dec->timestamp = dec->base_timestamp;
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if (!dec->blocksize)
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return GST_FLOW_NOT_NEGOTIATED;
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data = GST_BUFFER_DATA (buffer);
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outbuf = adpcmdec_decode_block (dec, data, dec->blocksize);
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if (outbuf == NULL) {
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
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("frame decode failed"), ret);
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}
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if (dec->blocksize > 0) {
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gst_adapter_push (dec->adapter, buf);
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while (gst_adapter_available (dec->adapter) >= dec->blocksize) {
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databuf = gst_adapter_take_buffer (dec->adapter, dec->blocksize);
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data = GST_BUFFER_DATA (databuf);
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ret = adpcmdec_decode_block (dec, data, dec->blocksize);
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/* Done with input data, free it */
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gst_buffer_unref (databuf);
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if (ret != GST_FLOW_OK)
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goto done;
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}
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} else {
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/* No explicit blocksize; we just process one input buffer at a time */
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ret = adpcmdec_decode_block (dec, GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf));
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gst_buffer_unref (buf);
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}
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done:
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gst_object_unref (dec);
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if (ret == GST_FLOW_OK)
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ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
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return ret;
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}
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static gboolean
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adpcmdec_sink_event (GstPad * pad, GstEvent * event)
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adpcmdec_start (GstAudioDecoder * bdec)
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{
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ADPCMDec *dec = (ADPCMDec *) gst_pad_get_parent (pad);
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gboolean res;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_STOP:
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dec->out_samples = 0;
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dec->timestamp = GST_CLOCK_TIME_NONE;
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dec->base_timestamp = GST_CLOCK_TIME_NONE;
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gst_adapter_clear (dec->adapter);
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/* Fall through */
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default:
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res = gst_pad_push_event (dec->srcpad, event);
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break;
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}
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gst_object_unref (dec);
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return res;
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ADPCMDec *dec = (ADPCMDec *) bdec;
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GST_DEBUG_OBJECT (dec, "start");
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dec->blocksize = 0;
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dec->rate = 0;
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dec->channels = 0;
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return TRUE;
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}
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static GstStateChangeReturn
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adpcmdec_change_state (GstElement * element, GstStateChange transition)
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static gboolean
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adpcmdec_stop (GstAudioDecoder * dec)
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{
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GstStateChangeReturn ret;
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ADPCMDec *dec = (ADPCMDec *) element;
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GST_DEBUG_OBJECT (dec, "stop");
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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adpcmdec_teardown (dec);
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return ret;
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}
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static void
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adpcmdec_dispose (GObject * obj)
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{
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G_OBJECT_CLASS (parent_class)->dispose (obj);
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return TRUE;
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}
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static void
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adpcmdec_init (ADPCMDec * dec, ADPCMDecClass * klass)
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{
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dec->sinkpad =
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gst_pad_new_from_static_template (&adpcmdec_sink_template, "sink");
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gst_pad_set_setcaps_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (adpcmdec_sink_setcaps));
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gst_pad_set_chain_function (dec->sinkpad, GST_DEBUG_FUNCPTR (adpcmdec_chain));
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gst_pad_set_event_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (adpcmdec_sink_event));
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gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
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dec->srcpad =
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gst_pad_new_from_static_template (&adpcmdec_src_template, "src");
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gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
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}
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static void
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adpcmdec_class_init (ADPCMDecClass * klass)
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{
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GObjectClass *gobjectclass = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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gobjectclass->dispose = adpcmdec_dispose;
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gstelement_class->change_state = adpcmdec_change_state;
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} static void
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GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) klass;
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base_class->start = GST_DEBUG_FUNCPTR (adpcmdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (adpcmdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (adpcmdec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (adpcmdec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmdec_handle_frame);
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}
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static void
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adpcmdec_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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