static void
gst_audio_parse_class_init (GstAudioParseClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstRawParseClass *rp_class = GST_RAW_PARSE_CLASS (klass);
GstCaps *caps;
gobject_class->set_property = gst_audio_parse_set_property;
gobject_class->get_property = gst_audio_parse_get_property;
gobject_class->finalize = gst_audio_parse_finalize;
rp_class->get_caps = gst_audio_parse_get_caps;
g_object_class_install_property (gobject_class, PROP_FORMAT,
g_param_spec_enum ("format", "Format",
"Format of audio samples in raw stream", GST_AUDIO_PARSE_FORMAT,
GST_AUDIO_PARSE_FORMAT_RAW,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RAW_FORMAT,
g_param_spec_enum ("raw-format", "Raw Format",
"Format of audio samples in raw stream", GST_TYPE_AUDIO_FORMAT,
GST_AUDIO_FORMAT_S16, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RATE,
g_param_spec_int ("rate", "Rate", "Rate of audio samples in raw stream",
1, INT_MAX, 44100, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CHANNELS,
g_param_spec_int ("channels", "Channels",
"Number of channels in raw stream", 1, 64, 2,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_INTERLEAVED,
g_param_spec_boolean ("interleaved", "Interleaved Layout",
"True if audio has interleaved layout", TRUE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS,
g_param_spec_value_array ("channel-positions", "Channel positions",
"Channel positions used on the output",
g_param_spec_enum ("channel-position", "Channel position",
"Channel position of the n-th input",
GST_TYPE_AUDIO_CHANNEL_POSITION,
GST_AUDIO_CHANNEL_POSITION_NONE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_USE_SINK_CAPS,
g_param_spec_boolean ("use-sink-caps", "Use sink caps",
"Use the sink caps for the format, only performing timestamping",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_metadata (gstelement_class, "Audio Parse",
"Filter/Audio",
"Converts stream into audio frames",
"Sebastian Dröge <*****@*****.**>");
caps = gst_caps_from_string (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
", layout = (string) { interleaved, non-interleaved }; "
"audio/x-alaw, rate=(int)[1,MAX], channels=(int)[1,MAX]; "
"audio/x-mulaw, rate=(int)[1,MAX], channels=(int)[1,MAX]");
gst_raw_parse_class_set_src_pad_template (rp_class, caps);
gst_raw_parse_class_set_multiple_frames_per_buffer (rp_class, TRUE);
gst_caps_unref (caps);
GST_DEBUG_CATEGORY_INIT (gst_audio_parse_debug, "audioparse", 0,
"audioparse element");
}
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstaudiosegmentclip.h"
static GstStaticPadTemplate sink_pad_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
static GstStaticPadTemplate src_pad_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
static void gst_audio_segment_clip_reset (GstSegmentClip * self);
static GstFlowReturn gst_audio_segment_clip_clip_buffer (GstSegmentClip * self,
GstBuffer * buffer, GstBuffer ** outbuf);
static gboolean gst_audio_segment_clip_set_caps (GstSegmentClip * self,
GstCaps * caps);
GST_DEBUG_CATEGORY_STATIC (gst_audio_segment_clip_debug);
#define GST_CAT_DEFAULT gst_audio_segment_clip_debug
G_DEFINE_TYPE (GstAudioSegmentClip, gst_audio_segment_clip,
// Capabilities that Audacity can handle
//
// This resolves to: (on little endien)
//
// "audio/x-raw, "
// "format = (string) {S16LE, S24_32LE, F32LE}, "
// "rate = (int) [ 1, max ]",
// "channels = (int) [ 1, max ]"
static GstStaticCaps supportedCaps =
GST_STATIC_CAPS(
GST_AUDIO_CAPS_MAKE(
"{"
GST_AUDIO_NE(S16) ", "
GST_AUDIO_NE(S24_32) ", "
GST_AUDIO_NE(F32)
"}"
)
);
// Context used for private stream data
struct GStreamContext
{
GstElement *mConv; // Audio converter
GstElement *mSink; // Application sink
bool mUse; // True if this stream should be imported
WaveTrack **mChannels; // Array of WaveTrack pointers, one for each channel
gint mNumChannels; // Number of channels
gdouble mSampleRate; // Sample rate
gchar *mType; // Audio type
#include "gstvideoframe-audiolevel.h"
#include <math.h>
#define GST_CAT_DEFAULT gst_videoframe_audiolevel_debug
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
# define FORMATS "{ S8, S16LE, S32LE, F32LE, F64LE }"
#else
# define FORMATS "{ S8, S16BE, S32BE, F32BE, F64BE }"
#endif
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static GstStaticPadTemplate audio_sink_template =
GST_STATIC_PAD_TEMPLATE ("asink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (FORMATS))
);
static GstStaticPadTemplate audio_src_template =
GST_STATIC_PAD_TEMPLATE ("asrc",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (FORMATS))
);
static GstStaticPadTemplate video_sink_template =
GST_STATIC_PAD_TEMPLATE ("vsink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-raw")
);
#include <orc/orc.h>
#else
#define orc_memcpy memcpy
#endif
#define gst_droidadec_parent_class parent_class
G_DEFINE_TYPE (GstDroidADec, gst_droidadec, GST_TYPE_AUDIO_DECODER);
GST_DEBUG_CATEGORY_EXTERN (gst_droid_adec_debug);
#define GST_CAT_DEFAULT gst_droid_adec_debug
static GstStaticPadTemplate gst_droidadec_src_template_factory =
GST_STATIC_PAD_TEMPLATE (GST_AUDIO_DECODER_SRC_NAME,
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_DEF_FORMAT)));
static void gst_droidadec_error (void *data, int err);
static void gst_droidadec_signal_eos (void *data);
static void gst_droidadec_data_available (void *data,
DroidMediaCodecData * encoded);
static GstFlowReturn gst_droidadec_finish (GstAudioDecoder * decoder);
static gboolean
gst_droidadec_create_codec (GstDroidADec * dec, GstBuffer * input)
{
DroidMediaCodecDecoderMetaData md;
const gchar *droid = gst_droid_codec_get_droid_type (dec->codec_type);
GST_INFO_OBJECT (dec, "create codec of type %s. Channels: %d, Rate: %d",
droid, dec->channels, dec->rate);
static GstFlowReturn gst_atdec_handle_frame (GstAudioDecoder * decoder,
GstBuffer * buffer);
static void gst_atdec_flush (GstAudioDecoder * decoder, gboolean hard);
static void gst_atdec_buffer_emptied (void *user_data,
AudioQueueRef queue, AudioQueueBufferRef buffer);
enum
{
PROP_0
};
static GstStaticPadTemplate gst_atdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE ("S16LE") ", layout=interleaved;"
GST_AUDIO_CAPS_MAKE ("F32LE") ", layout=interleaved")
);
static GstStaticPadTemplate gst_atdec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, mpegversion=4, framed=true, channels=[1,max];"
"audio/mpeg, mpegversion=1, layer=3")
);
G_DEFINE_TYPE_WITH_CODE (GstATDec, gst_atdec, GST_TYPE_AUDIO_DECODER,
GST_DEBUG_CATEGORY_INIT (gst_atdec_debug_category, "atdec", 0,
"debug category for atdec element"));
