Commit 004377b0 authored by Mark Nauwelaerts's avatar Mark Nauwelaerts
Browse files

wavpackdec: port to audiodecoder

parent b863df57
......@@ -55,8 +55,6 @@
#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
......@@ -80,15 +78,18 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
"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)
......@@ -108,11 +109,14 @@ static void
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
......@@ -121,33 +125,15 @@ gst_wavpack_dec_reset (GstWavpackDec * dec)
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 ();
......@@ -166,26 +152,73 @@ gst_wavpack_dec_finalize (GObject * object)
}
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")) {
......@@ -202,19 +235,13 @@ gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
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;
}
......@@ -225,28 +252,26 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec)
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;
......@@ -255,7 +280,9 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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))
......@@ -283,21 +310,15 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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)) ||
......@@ -308,22 +329,13 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
(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
......@@ -334,17 +346,7 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
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 */
......@@ -353,34 +355,20 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
/* 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:
......@@ -388,23 +376,25 @@ 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";
......@@ -418,86 +408,14 @@ decode_error:
} 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
......
......@@ -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);
......
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