#include <gst/gststructure.h>
-#define SINT (GST_AUDIO_FORMAT_FLAG_INT | GST_AUDIO_FORMAT_FLAG_SIGNED)
-#define UINT (GST_AUDIO_FORMAT_FLAG_INT)
+#define SINT (GST_AUDIO_FORMAT_FLAG_INTEGER | GST_AUDIO_FORMAT_FLAG_SIGNED)
+#define UINT (GST_AUDIO_FORMAT_FLAG_INTEGER)
#define MAKE_FORMAT(str,flags,end,width,depth,silent) \
{ GST_AUDIO_FORMAT_ ##str, G_STRINGIFY(str), flags, end, width, depth, silent }
* exists with the given parameters.
*/
GstAudioFormat
-gst_audio_format_build_int (gboolean sign, gint endianness,
+gst_audio_format_build_integer (gboolean sign, gint endianness,
gint width, gint depth)
{
gint i, e;
GstAudioFormatInfo *finfo = &formats[i];
/* must be int */
- if (!GST_AUDIO_FORMAT_INFO_IS_INT (finfo))
+ if (!GST_AUDIO_FORMAT_INFO_IS_INTEGER (finfo))
continue;
/* width and depth must match */
if (width != GST_AUDIO_FORMAT_INFO_WIDTH (finfo))
* gst_audio_format_fill_silence:
* @info: a #GstAudioFormatInfo
* @dest: a destination to fill
- * @lenfth: the length to fill
+ * @length: the length to fill
*
* Fill @length bytes in @dest with silence samples for @info.
*/
*
* Convert the values of @info into a #GstCaps.
*
- * Returns: a new #GstCaps containing the info of @info.
+ * Returns: (transfer full): the new #GstCaps containing the
+ * info of @info.
*/
GstCaps *
gst_audio_info_to_caps (GstAudioInfo * info)
return caps;
}
-
-/**
- * gst_audio_frame_byte_size:
- * @pad: the #GstPad to get the caps from
- *
- * Calculate byte size of an audio frame.
- *
- * Returns: the byte size, or 0 if there was an error
- */
-int
-gst_audio_frame_byte_size (GstPad * pad)
-{
- /* FIXME: this should be moved closer to the gstreamer core
- * and be implemented for every mime type IMO
- */
-
- int width = 0;
- int channels = 0;
- GstCaps *caps;
- GstStructure *structure;
-
- /* get caps of pad */
- caps = gst_pad_get_current_caps (pad);
-
- if (caps == NULL)
- goto no_caps;
-
- structure = gst_caps_get_structure (caps, 0);
-
- gst_structure_get_int (structure, "width", &width);
- gst_structure_get_int (structure, "channels", &channels);
- gst_caps_unref (caps);
-
- return (width / 8) * channels;
-
- /* ERRORS */
-no_caps:
- {
- /* ERROR: could not get caps of pad */
- g_warning ("gstaudio: could not get caps of pad %s:%s\n",
- GST_DEBUG_PAD_NAME (pad));
- return 0;
- }
-}
-
-/**
- * gst_audio_frame_length:
- * @pad: the #GstPad to get the caps from
- * @buf: the #GstBuffer
- *
- * Calculate length of buffer in frames.
- *
- * Returns: 0 if there's an error, or the number of frames if everything's ok
- */
-long
-gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
-{
- /* FIXME: this should be moved closer to the gstreamer core
- * and be implemented for every mime type IMO
- */
- int frame_byte_size = 0;
-
- frame_byte_size = gst_audio_frame_byte_size (pad);
- if (frame_byte_size == 0)
- /* error */
- return 0;
- /* FIXME: this function assumes the buffer size to be a whole multiple
- * of the frame byte size
- */
- return gst_buffer_get_size (buf) / frame_byte_size;
-}
-
-/**
- * gst_audio_duration_from_pad_buffer:
- * @pad: the #GstPad to get the caps from
- * @buf: the #GstBuffer
- *
- * Calculate length in nanoseconds of audio buffer @buf based on capabilities of
- * @pad.
- *
- * Returns: the length.
- */
-GstClockTime
-gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf)
-{
- long bytes = 0;
- int width = 0;
- int channels = 0;
- int rate = 0;
- GstCaps *caps;
- GstStructure *structure;
-
- g_assert (GST_IS_BUFFER (buf));
-
- /* get caps of pad */
- caps = gst_pad_get_current_caps (pad);
- if (caps == NULL)
- goto no_caps;
-
- structure = gst_caps_get_structure (caps, 0);
- bytes = gst_buffer_get_size (buf);
- gst_structure_get_int (structure, "width", &width);
- gst_structure_get_int (structure, "channels", &channels);
- gst_structure_get_int (structure, "rate", &rate);
- gst_caps_unref (caps);
-
- g_assert (bytes != 0);
- g_assert (width != 0);
- g_assert (channels != 0);
- g_assert (rate != 0);
-
- return (bytes * 8 * GST_SECOND) / (rate * channels * width);
-
- /* ERRORS */
-no_caps:
- {
- /* ERROR: could not get caps of pad */
- g_warning ("gstaudio: could not get caps of pad %s:%s\n",
- GST_DEBUG_PAD_NAME (pad));
- return GST_CLOCK_TIME_NONE;
- }
-}
-
-/**
- * gst_audio_is_buffer_framed:
- * @pad: the #GstPad to get the caps from
- * @buf: the #GstBuffer
- *
- * Check if the buffer size is a whole multiple of the frame size.
- *
- * Returns: %TRUE if buffer size is multiple.
- */
-gboolean
-gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
-{
- if (gst_buffer_get_size (buf) % gst_audio_frame_byte_size (pad) == 0)
- return TRUE;
- else
- return FALSE;
-}
-
/**
* gst_audio_buffer_clip:
* @buffer: The buffer to clip.
- * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which the buffer should be clipped.
+ * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
+ * the buffer should be clipped.
* @rate: sample rate.
- * @frame_size: size of one audio frame in bytes.
+ * @bpf: size of one audio frame in bytes. This is the size of one sample
+ * * channels.
*
* Clip the the buffer to the given %GstSegment.
*
- * After calling this function the caller does not own a reference to
+ * After calling this function the caller does not own a reference to
* @buffer anymore.
*
* Returns: %NULL if the buffer is completely outside the configured segment,
* otherwise the clipped buffer is returned.
*
* If the buffer has no timestamp, it is assumed to be inside the segment and
- * is not clipped
+ * is not clipped
*
* Since: 0.10.14
*/
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
- gint frame_size)
+ gint bpf)
{
GstBuffer *ret;
GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
duration = GST_BUFFER_DURATION (buffer);
} else {
change_duration = FALSE;
- duration = gst_util_uint64_scale (size / frame_size, GST_SECOND, rate);
+ duration = gst_util_uint64_scale (size / bpf, GST_SECOND, rate);
}
if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
offset_end = GST_BUFFER_OFFSET_END (buffer);
} else {
change_offset_end = FALSE;
- offset_end = offset + size / frame_size;
+ offset_end = offset + size / bpf;
}
if (segment->format == GST_FORMAT_TIME) {
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset)
offset += diff;
- trim += diff * frame_size;
- size -= diff * frame_size;
+ trim += diff * bpf;
+ size -= diff * bpf;
}
diff = stop - cstop;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset_end)
offset_end -= diff;
- size -= diff * frame_size;
+ size -= diff * bpf;
}
} else {
gst_buffer_unref (buffer);
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
- trim += diff * frame_size;
- size -= diff * frame_size;
+ trim += diff * bpf;
+ size -= diff * bpf;
}
diff = stop - cstop;
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
- size -= diff * frame_size;
+ size -= diff * bpf;
}
} else {
gst_buffer_unref (buffer);
/**
* GstAudioFormatFlags:
- * @GST_AUDIO_FORMAT_FLAG_INT: int samples
+ * @GST_AUDIO_FORMAT_FLAG_INTEGER: integer samples
* @GST_AUDIO_FORMAT_FLAG_FLOAT: float samples
* @GST_AUDIO_FORMAT_FLAG_SIGNED: signed samples
* @GST_AUDIO_FORMAT_FLAG_COMPLEX: complex layout
*/
typedef enum
{
- GST_AUDIO_FORMAT_FLAG_INT = (1 << 0),
+ GST_AUDIO_FORMAT_FLAG_INTEGER = (1 << 0),
GST_AUDIO_FORMAT_FLAG_FLOAT = (1 << 1),
GST_AUDIO_FORMAT_FLAG_SIGNED = (1 << 2),
GST_AUDIO_FORMAT_FLAG_COMPLEX = (1 << 4)
#define GST_AUDIO_FORMAT_INFO_NAME(info) ((info)->name)
#define GST_AUDIO_FORMAT_INFO_FLAGS(info) ((info)->flags)
-#define GST_AUDIO_FORMAT_INFO_IS_INT(info) ((info)->flags & GST_AUDIO_FORMAT_FLAG_INT)
+#define GST_AUDIO_FORMAT_INFO_IS_INTEGER(info) ((info)->flags & GST_AUDIO_FORMAT_FLAG_INTEGER)
#define GST_AUDIO_FORMAT_INFO_IS_FLOAT(info) ((info)->flags & GST_AUDIO_FORMAT_FLAG_FLOAT)
#define GST_AUDIO_FORMAT_INFO_IS_SIGNED(info) ((info)->flags & GST_AUDIO_FORMAT_FLAG_SIGNED)
#define GST_AUDIO_FORMAT_INFO_DEPTH(info) ((info)->depth)
-GstAudioFormat gst_audio_format_build_int (gboolean sign, gint endianness,
- gint width, gint depth) G_GNUC_CONST;
+GstAudioFormat gst_audio_format_build_integer (gboolean sign, gint endianness,
+ gint width, gint depth) G_GNUC_CONST;
+
+GstAudioFormat gst_audio_format_from_string (const gchar *format) G_GNUC_CONST;
+const gchar * gst_audio_format_to_string (GstAudioFormat format) G_GNUC_CONST;
-GstAudioFormat gst_audio_format_from_string (const gchar *format) G_GNUC_CONST;
-const gchar * gst_audio_format_to_string (GstAudioFormat format) G_GNUC_CONST;
const GstAudioFormatInfo *
- gst_audio_format_get_info (GstAudioFormat format) G_GNUC_CONST;
+ gst_audio_format_get_info (GstAudioFormat format) G_GNUC_CONST;
-void gst_audio_format_fill_silence (const GstAudioFormatInfo *info,
- gpointer dest, gsize length);
+void gst_audio_format_fill_silence (const GstAudioFormatInfo *info,
+ gpointer dest, gsize length);
/**
* GstAudioFlags:
* @GST_AUDIO_FLAG_NONE: no valid flag
* handling
*/
-/* get byte size of audio frame (based on caps of pad */
-int gst_audio_frame_byte_size (GstPad* pad);
-
-/* get length in frames of buffer */
-long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
-
-GstClockTime gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf);
-
-/* check if the buffer size is a whole multiple of the frame size */
-gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
-
-GstBuffer *gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment, gint rate, gint frame_size);
+GstBuffer * gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment,
+ gint rate, gint bpf);
G_END_DECLS
};
#define DOUBLE_INTERMEDIATE_FORMAT(ctx) \
- ((!GST_AUDIO_FORMAT_INFO_IS_INT (ctx->in.finfo) && \
- !GST_AUDIO_FORMAT_INFO_IS_INT (ctx->out.finfo)) || \
+ ((!GST_AUDIO_FORMAT_INFO_IS_INTEGER (ctx->in.finfo) && \
+ !GST_AUDIO_FORMAT_INFO_IS_INTEGER (ctx->out.finfo)) || \
(ctx->ns != NOISE_SHAPING_NONE))
static gint
{
gint index = 0;
- if (GST_AUDIO_FORMAT_INFO_IS_INT (fmt)) {
+ if (GST_AUDIO_FORMAT_INFO_IS_INTEGER (fmt)) {
index += (GST_AUDIO_FORMAT_INFO_WIDTH (fmt) / 8 - 1) * 4;
index += GST_AUDIO_FORMAT_INFO_IS_LE (fmt) ? 0 : 2;
index += GST_AUDIO_FORMAT_INFO_IS_SIGNED (fmt) ? 1 : 0;
* as DA converters only can do a SNR up to 20 bits in reality.
* Also don't dither or apply noise shaping if target depth is larger than
* source depth. */
- if (out_depth <= 20 && (!GST_AUDIO_FORMAT_INFO_IS_INT (in->finfo)
+ if (out_depth <= 20 && (!GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo)
|| in_depth >= out_depth)) {
ctx->dither = dither;
ctx->ns = ns;
ctx->in_default, ctx->mix_passthrough, ctx->out_default);
ctx->in_scale =
- GST_AUDIO_FORMAT_INFO_IS_INT (in->finfo) ? (32 - in_depth) : 0;
+ GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo) ? (32 - in_depth) : 0;
ctx->out_scale =
- GST_AUDIO_FORMAT_INFO_IS_INT (out->finfo) ? (32 - out_depth) : 0;
+ GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo) ? (32 - out_depth) : 0;
gst_audio_quantize_setup (ctx);
}
/* we only need to quantize if output format is int */
- if (GST_AUDIO_FORMAT_INFO_IS_INT (ctx->out.finfo)) {
+ if (GST_AUDIO_FORMAT_INFO_IS_INTEGER (ctx->out.finfo)) {
if (ctx->out_default)
outbuf = dst;
else
projects / platform / upstream / gstreamer.git / commitdiff