diff options
Diffstat (limited to 'gst-libs')
| -rw-r--r-- | gst-libs/gst/audio/audio.h | 152 |
1 files changed, 86 insertions, 66 deletions
diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h index a737e468..c22052f5 100644 --- a/gst-libs/gst/audio/audio.h +++ b/gst-libs/gst/audio/audio.h @@ -22,78 +22,97 @@ #include <gst/audio/audioclock.h> +G_BEGIN_DECLS + /* For people that are looking at this source: the purpose of these defines is * to make GstCaps a bit easier, in that you don't have to know all of the * properties that need to be defined. you can just use these macros. currently * (8/01) the only plugins that use these are the passthrough, speed, volume, - * adder, and [de]interleave plugins. - * These are for convenience only, and do not specify the 'limits' of - * GStreamer. you might also use these definitions as a + * adder, and [de]interleave plugins. These are for convenience only, and do not + * specify the 'limits' of GStreamer. you might also use these definitions as a * base for making your own caps, if need be. * - * For example, to make a source pad that can output mono streams of either - * float or int: - - template = gst_pad_template_new - ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - gst_caps_append(gst_caps_new ("sink_int", "audio/raw", - GST_AUDIO_INT_PAD_TEMPLATE_PROPS), - gst_caps_new ("sink_float", "audio/raw", - GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS)), - NULL); - - srcpad = gst_pad_new_from_template(template,"src"); - - * Andy Wingo, 18 August 2001 + * For example, to make a source pad that can output streams of either mono + * float or any channel int: + * + * template = gst_pad_template_new + * ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, + * gst_caps_append(gst_caps_new ("sink_int", "audio/x-raw-int", + * GST_AUDIO_INT_PAD_TEMPLATE_PROPS), + * gst_caps_new ("sink_float", "audio/x-raw-float", + * GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)), + * NULL); + * + * sinkpad = gst_pad_new_from_template(template, "sink"); + * + * Andy Wingo, 18 August 2001 * Thomas, 6 September 2002 */ -/* a few useful defines for arbitrary limits */ -#define GST_AUDIO_MIN_RATE 4000 -#define GST_AUDIO_MAX_RATE 96000 -#define GST_AUDIO_DEF_RATE 44100 +#define GST_AUDIO_DEF_RATE 44100 #define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \ - gst_props_new (\ - "endianness", GST_PROPS_INT (G_BYTE_ORDER),\ - "signed", GST_PROPS_LIST (\ - GST_PROPS_BOOLEAN (TRUE),\ - GST_PROPS_BOOLEAN (FALSE)\ - ),\ - "width", GST_PROPS_LIST (GST_PROPS_INT (8), \ - GST_PROPS_INT (16)), \ - "depth", GST_PROPS_LIST (GST_PROPS_INT (8), \ - GST_PROPS_INT (16)),\ - "rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \ - GST_AUDIO_MAX_RATE),\ - "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\ - NULL) + gst_props_new (\ + "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + "endianness", GST_PROPS_LIST (\ + GST_PROPS_INT (G_LITTLE_ENDIAN),\ + GST_PROPS_INT (G_BIG_ENDIAN)\ + ),\ + "width", GST_PROPS_LIST (\ + GST_PROPS_INT (8),\ + GST_PROPS_INT (16),\ + GST_PROPS_INT (32)\ + ),\ + "depth", GST_PROPS_INT_RANGE (1, 32),\ + "signed", GST_PROPS_LIST (\ + GST_PROPS_BOOLEAN (TRUE),\ + GST_PROPS_BOOLEAN (FALSE)\ + ),\ + NULL) #define GST_AUDIO_INT_MONO_PAD_TEMPLATE_PROPS \ - gst_props_new (\ - "endianness", GST_PROPS_INT (G_BYTE_ORDER),\ - "signed", GST_PROPS_LIST (\ - GST_PROPS_BOOLEAN (TRUE),\ - GST_PROPS_BOOLEAN (FALSE)\ - ),\ - "width", GST_PROPS_LIST (GST_PROPS_INT (8), \ - GST_PROPS_INT (16)),\ - "depth", GST_PROPS_LIST (GST_PROPS_INT (8), \ - GST_PROPS_INT (16)),\ - "rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \ - GST_AUDIO_MAX_RATE),\ - "channels", GST_PROPS_INT (1),\ - NULL) - -#define GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS \ - gst_props_new (\ - "depth", GST_PROPS_INT (32),\ - "endianness", GST_PROPS_INT (G_BYTE_ORDER),\ - "intercept", GST_PROPS_FLOAT (0.0),\ - "slope", GST_PROPS_FLOAT (1.0),\ - "rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \ - GST_AUDIO_MAX_RATE),\ - "channels", GST_PROPS_INT (1),\ - NULL) + gst_props_new (\ + "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + "channels", GST_PROPS_INT (1),\ + "endianness", GST_PROPS_LIST (\ + GST_PROPS_INT (G_LITTLE_ENDIAN),\ + GST_PROPS_INT (G_BIG_ENDIAN)\ + ),\ + "width", GST_PROPS_LIST (\ + GST_PROPS_INT (8),\ + GST_PROPS_INT (16),\ + GST_PROPS_INT (32)\ + ),\ + "depth", GST_PROPS_INT_RANGE (1, 32),\ + "signed", GST_PROPS_LIST (\ + GST_PROPS_BOOLEAN (TRUE),\ + GST_PROPS_BOOLEAN (FALSE)\ + ),\ + NULL) + +#define GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS \ + gst_props_new (\ + "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + "endianness", GST_PROPS_LIST (\ + GST_PROPS_INT (G_LITTLE_ENDIAN),\ + GST_PROPS_INT (G_BIG_ENDIAN)\ + ),\ + "width", GST_PROPS_LIST (\ + GST_PROPS_INT (32),\ + GST_PROPS_INT (64)\ + ),\ + "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + NULL) + +#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS \ + gst_props_new (\ + "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + "channels", GST_PROPS_INT (1),\ + "endianness", GST_PROPS_INT (G_BYTE_ORDER),\ + "width", GST_PROPS_INT (32),\ + "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + NULL) /* * this library defines and implements some helper functions for audio @@ -101,21 +120,22 @@ */ /* get byte size of audio frame (based on caps of pad */ -int gst_audio_frame_byte_size (GstPad* pad); +int gst_audio_frame_byte_size (GstPad* pad); /* get length in frames of buffer */ -long gst_audio_frame_length (GstPad* pad, GstBuffer* buf); +long gst_audio_frame_length (GstPad* pad, GstBuffer* buf); /* get frame rate based on caps */ -long gst_audio_frame_rate (GstPad *pad); +long gst_audio_frame_rate (GstPad *pad); /* calculate length in seconds of audio buffer buf based on caps of pad */ -double gst_audio_length (GstPad* pad, GstBuffer* buf); +double gst_audio_length (GstPad* pad, GstBuffer* buf); /* calculate highest possible sample value based on capabilities of pad */ -long gst_audio_highest_sample_value (GstPad* pad); +long gst_audio_highest_sample_value (GstPad* pad); /* check if the buffer size is a whole multiple of the frame size */ -gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf); +gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf); +G_END_DECLS |