gstreamer/gst-libs/gst/audio/audio.c

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "audio.h"
2004-11-25 20:36:29 +00:00
#include "multichannel-enumtypes.h"
#include <gst/gststructure.h>
/**
* SECTION:gstaudio
* @short_description: Support library for audio elements
*
* This library contains some helper functions for audio elements.
*/
/**
* 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;
const GstCaps *caps = NULL;
GstStructure *structure;
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL) {
/* 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;
}
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "width", &width);
gst_structure_get_int (structure, "channels", &channels);
return (width / 8) * channels;
}
/**
* 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_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;
GstClockTime length;
const GstCaps *caps = NULL;
GstStructure *structure;
g_assert (GST_IS_BUFFER (buf));
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL) {
/* ERROR: could not get caps of pad */
g_warning ("gstaudio: could not get caps of pad %s:%s\n",
GST_DEBUG_PAD_NAME (pad));
length = GST_CLOCK_TIME_NONE;
} else {
structure = gst_caps_get_structure (caps, 0);
bytes = GST_BUFFER_SIZE (buf);
gst_structure_get_int (structure, "width", &width);
gst_structure_get_int (structure, "channels", &channels);
gst_structure_get_int (structure, "rate", &rate);
g_assert (bytes != 0);
g_assert (width != 0);
g_assert (channels != 0);
g_assert (rate != 0);
length = (bytes * 8 * GST_SECOND) / (rate * channels * width);
}
return length;
}
/**
* 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_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
return TRUE;
else
return FALSE;
}
/* _getcaps helper functions
* sets structure fields to default for audio type
* flag determines which structure fields to set to default
* keep these functions in sync with the templates in audio.h
*/
/* private helper function
* sets a list on the structure
* pass in structure, fieldname for the list, type of the list values,
* number of list values, and each of the values, terminating with NULL
*/
static void
_gst_audio_structure_set_list (GstStructure * structure,
const gchar * fieldname, GType type, int number, ...)
{
va_list varargs;
GValue value = { 0 };
GArray *array;
int j;
g_return_if_fail (structure != NULL);
g_value_init (&value, GST_TYPE_LIST);
array = g_value_peek_pointer (&value);
va_start (varargs, number);
for (j = 0; j < number; ++j) {
int i;
gboolean b;
GValue list_value = { 0 };
switch (type) {
case G_TYPE_INT:
i = va_arg (varargs, int);
g_value_init (&list_value, G_TYPE_INT);
g_value_set_int (&list_value, i);
break;
case G_TYPE_BOOLEAN:
b = va_arg (varargs, gboolean);
g_value_init (&list_value, G_TYPE_BOOLEAN);
g_value_set_boolean (&list_value, b);
break;
default:
g_warning
("_gst_audio_structure_set_list: LIST of given type not implemented.");
}
g_array_append_val (array, list_value);
}
gst_structure_set_value (structure, fieldname, &value);
va_end (varargs);
}
/**
* gst_audio_structure_set_int:
* @structure: a #GstStructure
* @flag: a set of #GstAudioFieldFlag
*
* Do not use anymore.
*
* Deprecated: use gst_structure_set()
*/
#ifndef GST_REMOVE_DEPRECATED
#ifdef GST_DISABLE_DEPRECATED
typedef enum
{
GST_AUDIO_FIELD_RATE = (1 << 0),
GST_AUDIO_FIELD_CHANNELS = (1 << 1),
GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
GST_AUDIO_FIELD_WIDTH = (1 << 3),
GST_AUDIO_FIELD_DEPTH = (1 << 4),
GST_AUDIO_FIELD_SIGNED = (1 << 5),
} GstAudioFieldFlag;
#endif /* GST_DISABLE_DEPRECATED */
Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static... Original commit message from CVS: * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/alsa/gstalsasrc.c: (set_hwparams): * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri): * ext/ogg/gstoggmux.h: * ext/ogg/gstogmparse.c: * gst-libs/gst/audio/audio.c: * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc): * gst-libs/gst/pbutils/missing-plugins.c: (gst_missing_uri_sink_message_new), (gst_missing_element_message_new), (gst_missing_decoder_message_new), (gst_missing_encoder_message_new): * gst-libs/gst/rtp/gstbasertppayload.c: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_bye_get_reason): * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/ffmpegcolorspace/imgconvert.c: * gst/playback/test.c: (gen_video_element), (gen_audio_element): * gst/typefind/gsttypefindfunctions.c: * gst/videoscale/vs_4tap.c: * gst/videoscale/vs_4tap.h: * sys/v4l/gstv4lelement.c: * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps): * sys/v4l/v4l_calls.c: * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init), (gst_v4lsrc_try_capture): * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/audioconvert.c: * tests/check/elements/audioresample.c: (fail_unless_perfect_stream): * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc): * tests/check/elements/decodebin.c: * tests/check/elements/gdpdepay.c: (setup_gdpdepay), (setup_gdpdepay_streamheader): * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST), (setup_gdppay_streamheader): * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink): * tests/check/elements/multifdsink.c: (setup_multifdsink): * tests/check/elements/textoverlay.c: * tests/check/elements/videorate.c: (setup_videorate): * tests/check/elements/videotestsrc.c: (setup_videotestsrc): * tests/check/elements/volume.c: (setup_volume): * tests/check/elements/vorbisdec.c: (setup_vorbisdec): * tests/check/elements/vorbistag.c: * tests/check/generic/clock-selection.c: * tests/check/generic/states.c: (setup), (teardown): * tests/check/libs/cddabasesrc.c: * tests/check/libs/video.c: * tests/check/pipelines/gio.c: * tests/check/pipelines/oggmux.c: * tests/check/pipelines/simple-launch-lines.c: (simple_launch_lines_suite): * tests/check/pipelines/streamheader.c: * tests/check/pipelines/theoraenc.c: * tests/check/pipelines/vorbisdec.c: * tests/check/pipelines/vorbisenc.c: * tests/examples/seek/scrubby.c: * tests/examples/seek/seek.c: (query_positions_elems), (query_positions_pads): * tests/icles/stress-xoverlay.c: (myclock): Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static, using NULL instead of 0 for pointers and using "foo (void)" instead of "foo ()" for declarations. * win32/common/libgstrtp.def: Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-03 06:04:31 +00:00
void
gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
{
/* was added here:
* http://webcvs.freedesktop.org/gstreamer/gst-plugins-base/gst-libs/gst/audio/audio.c?r1=1.16&r2=1.17
* but it is not used
*/
if (flag & GST_AUDIO_FIELD_RATE)
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
NULL);
if (flag & GST_AUDIO_FIELD_CHANNELS)
gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,
NULL);
if (flag & GST_AUDIO_FIELD_ENDIANNESS)
_gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,
G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
if (flag & GST_AUDIO_FIELD_WIDTH)
_gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,
NULL);
if (flag & GST_AUDIO_FIELD_DEPTH)
gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
if (flag & GST_AUDIO_FIELD_SIGNED)
_gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,
FALSE, NULL);
}
#endif /* GST_REMOVE_DEPRECATED */
/**
* 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.
* @rate: sample rate.
* @frame_size: size of one audio frame in bytes.
*
* Clip the the buffer to the given %GstSegment.
*
* 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
*
* Since: 0.10.14
*/
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
gint frame_size)
{
GstBuffer *ret;
GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
guint8 *data;
guint size;
gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =
TRUE;
g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||
segment->format == GST_FORMAT_DEFAULT, buffer);
g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
/* No timestamp - assume the buffer is completely in the segment */
return buffer;
/* Get copies of the buffer metadata to change later.
* Calculate the missing values for the calculations,
* they won't be changed later though. */
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
duration = GST_BUFFER_DURATION (buffer);
} else {
change_duration = FALSE;
duration = gst_util_uint64_scale (size / frame_size, GST_SECOND, rate);
}
if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
offset = GST_BUFFER_OFFSET (buffer);
} else {
change_offset = FALSE;
offset = 0;
}
if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {
offset_end = GST_BUFFER_OFFSET_END (buffer);
} else {
change_offset_end = FALSE;
offset_end = offset + size / frame_size;
}
if (segment->format == GST_FORMAT_TIME) {
/* Handle clipping for GST_FORMAT_TIME */
gint64 start, stop, cstart, cstop, diff;
start = timestamp;
stop = timestamp + duration;
if (gst_segment_clip (segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
timestamp = cstart;
if (change_duration)
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset)
offset += diff;
data += diff * frame_size;
size -= diff * frame_size;
}
diff = stop - cstop;
if (diff > 0) {
/* duration is always valid if stop is valid */
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset_end)
offset_end -= diff;
size -= diff * frame_size;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
} else {
/* Handle clipping for GST_FORMAT_DEFAULT */
gint64 start, stop, cstart, cstop, diff;
g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);
start = offset;
stop = offset_end;
if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
offset = cstart;
timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
data += diff * frame_size;
size -= diff * frame_size;
}
diff = stop - cstop;
if (diff > 0) {
offset_end = cstop;
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
size -= diff * frame_size;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
}
/* Get a metadata writable buffer and apply all changes */
ret = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP (ret) = timestamp;
GST_BUFFER_SIZE (ret) = size;
GST_BUFFER_DATA (ret) = data;
if (change_duration)
GST_BUFFER_DURATION (ret) = duration;
if (change_offset)
GST_BUFFER_OFFSET (ret) = offset;
if (change_offset_end)
GST_BUFFER_OFFSET_END (ret) = offset_end;
return ret;
}