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baseaudioutils: remove, merged into or superseded by audio.c

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Tim-Philipp Müller 2011-08-27 12:41:28 +01:00
parent 7f0c7e5f82
commit 52ecb383d7
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417
gst-libs/gst/audio/gstbaseaudioutils.c
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@ -1,417 +0,0 @@
/* GStreamer
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* 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.
*/
#include "gstbaseaudioutils.h"
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <string.h>
/**
* gst_base_audio_format_init:
* @info: (transfer none): a #GstAudioFormatInfo instance
*
* Initializes a #GstAudioFormatInfo instance for subsequent use.
* The instance must either have been previously cleared or set to 0
* upon allocation.
*/
void
gst_base_audio_format_info_init (GstAudioFormatInfo * info)
{
g_return_if_fail (info != NULL);
/* no-op for now */
}
/**
* gst_base_audio_format_info_new:
*
* Allocate a new #GstAudioFormatInfo structure and initialize it using
* gst_base_audio_format_info().
*
* Free-function: gst_base_audio_format_info_free
*
* Returns: (transfer full): a new #GstAudioFormatInfo,
* free with gst_base_audio_format_info_free().
*/
GstAudioFormatInfo *
gst_base_audio_format_info_new (void)
{
GstAudioFormatInfo *result;
result = g_slice_new0 (GstAudioFormatInfo);
gst_base_audio_format_info_init (result);
return result;
}
/**
* gst_base_audio_format_clear:
* @info: (transfer none): a #GstAudioFormatInfo instance
*
* Clears a #GstAudioFormatInfo instance.
*/
void
gst_base_audio_format_info_clear (GstAudioFormatInfo * info)
{
g_return_if_fail (info != NULL);
g_free (info->channel_pos);
memset (info, 0, sizeof (*info));
}
/**
* gst_base_audio_format_copy:
* @info: (transfer none): a #GstAudioFormatInfo instance
*
* Create a copy of a given #GstAudioFormatInfo instance.
*
* Returns: (transfer full): a new #GstAudioFormatInfo,
* free with gst_base_audio_format_info_free()
*/
GstAudioFormatInfo *
gst_base_audio_format_info_copy (GstAudioFormatInfo * info)
{
GstAudioFormatInfo *result = NULL;
if (info) {
result = (GstAudioFormatInfo *) g_slice_copy (sizeof (GstAudioFormatInfo),
info);
}
return result;
}
/**
* gst_base_audio_format_free:
* @info: (in) (transfer full): a #GstAudioFormatInfo instance
*
* Frees the allocated @info.
*/
void
gst_base_audio_format_info_free (GstAudioFormatInfo * info)
{
g_slice_free (GstAudioFormatInfo, info);
}
#define CHECK_VALUE(var, val) \
G_STMT_START { \
if (!res) \
goto fail; \
var = val; \
} G_STMT_END
/**
* gst_base_audio_parse_caps:
* @caps: a #GstCaps
* @state: a #GstAudioFormatInfo
*
* Parses audio format as represented by @caps into a more concise form
* as represented by @state.
*
* Returns: TRUE if parsing succeeded, otherwise FALSE
*/
gboolean
gst_base_audio_parse_caps (GstCaps * caps, GstAudioFormatInfo * info)
{
gboolean res = TRUE;
GstStructure *s;
gboolean vb;
gint vi;
g_return_val_if_fail (caps != NULL, FALSE);
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
gst_base_audio_format_info_clear (info);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_has_name (s, "audio/x-raw-int"))
info->is_int = TRUE;
else if (gst_structure_has_name (s, "audio/x-raw-float"))
info->is_int = FALSE;
else
goto fail;
res = gst_structure_get_int (s, "rate", &vi);
CHECK_VALUE (info->rate, vi);
res &= gst_structure_get_int (s, "channels", &vi);
CHECK_VALUE (info->channels, vi);
res &= gst_structure_get_int (s, "width", &vi);
CHECK_VALUE (info->width, vi);
res &= (!info->is_int || gst_structure_get_int (s, "depth", &vi));
CHECK_VALUE (info->depth, vi);
res &= gst_structure_get_int (s, "endianness", &vi);
CHECK_VALUE (info->endian, vi);
res &= (!info->is_int || gst_structure_get_boolean (s, "signed", &vb));
CHECK_VALUE (info->sign, vb);
info->bpf = (info->width / 8) * info->channels;
GST_LOG ("bpf: %d", info->bpf);
if (!info->bpf)
goto fail;
info->channel_pos = gst_audio_get_channel_positions (s);
return res;
/* ERRORS */
fail:
{
/* there should not be caps out there that fail parsing ... */
GST_WARNING ("failed to parse caps %" GST_PTR_FORMAT, caps);
return res;
}
}
/**
* gst_base_audio_compare_format_info:
* @from: a #GstAudioFormatInfo
* @to: a #GstAudioFormatInfo
*
* Checks whether provides #GstAudioFormatInfo instances represent identical
* audio dat.
*
* Returns: TRUE if represent audio formats are identical
*/
gboolean
gst_base_audio_compare_format_info (GstAudioFormatInfo * from,
GstAudioFormatInfo * to)
{
g_return_val_if_fail (from != NULL, FALSE);
g_return_val_if_fail (to != NULL, FALSE);
/* this is a bit silly, so maybe should not need this function at all */
return memcmp (from, to, sizeof (*from));
}
/**
* gst_base_audio_add_streamheader:
* @caps: a #GstCaps
* @buf: header buffers
*
* Adds given buffers to an array of buffers set as streamheader field
* on the given @caps. List of buffer arguments must be NULL-terminated.
*
* Returns: input caps with a streamheader field added, or NULL if some error
*/
GstCaps *
gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...)
{
GstStructure *structure = NULL;
va_list va;
GValue array = { 0 };
GValue value = { 0 };
g_return_val_if_fail (caps != NULL, NULL);
g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
g_value_init (&array, GST_TYPE_ARRAY);
va_start (va, buf);
/* put buffers in a fixed list */
while (buf) {
g_assert (gst_buffer_is_metadata_writable (buf));
/* mark buffer */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
g_value_init (&value, GST_TYPE_BUFFER);
buf = gst_buffer_copy (buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
gst_value_set_buffer (&value, buf);
gst_buffer_unref (buf);
gst_value_array_append_value (&array, &value);
g_value_unset (&value);
buf = va_arg (va, GstBuffer *);
}
gst_structure_set_value (structure, "streamheader", &array);
g_value_unset (&array);
return caps;
}
/**
* gst_base_audio_encoded_audio_convert:
* @fmt: audio format of the encoded audio
* @bytes: number of encoded bytes
* @samples: number of encoded samples
* @src_format: source format
* @src_value: source value
* @dest_format: destination format
* @dest_value: destination format
*
* Helper function to convert @src_value in @src_format to @dest_value in
* @dest_format for encoded audio data. Conversion is possible between
* BYTE and TIME format by using estimated bitrate based on
* @samples and @bytes (and @fmt).
*/
gboolean
gst_base_audio_encoded_audio_convert (GstAudioFormatInfo * fmt,
gint64 bytes, gint64 samples, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = FALSE;
g_return_val_if_fail (dest_format != NULL, FALSE);
g_return_val_if_fail (dest_value != NULL, FALSE);
if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
src_value == -1)) {
if (dest_value)
*dest_value = src_value;
return TRUE;
}
if (samples == 0 || bytes == 0 || fmt->rate == 0) {
GST_DEBUG ("not enough metadata yet to convert");
goto exit;
}
bytes *= fmt->rate;
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale (src_value,
GST_SECOND * samples, bytes);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = gst_util_uint64_scale (src_value, bytes,
samples * GST_SECOND);
res = TRUE;
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
exit:
return res;
}
/**
* gst_base_audio_raw_audio_convert:
* @fmt: audio format of the encoded audio
* @src_format: source format
* @src_value: source value
* @dest_format: destination format
* @dest_value: destination format
*
* Helper function to convert @src_value in @src_format to @dest_value in
* @dest_format for encoded audio data. Conversion is possible between
* BYTE, DEFAULT and TIME format based on audio characteristics provided
* by @fmt.
*/
gboolean
gst_base_audio_raw_audio_convert (GstAudioFormatInfo * fmt,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = FALSE;
guint scale = 1;
gint bytes_per_sample, rate, byterate;
g_return_val_if_fail (dest_format != NULL, FALSE);
g_return_val_if_fail (dest_value != NULL, FALSE);
if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
src_value == -1)) {
if (dest_value)
*dest_value = src_value;
return TRUE;
}
bytes_per_sample = fmt->bpf;
rate = fmt->rate;
byterate = bytes_per_sample * rate;
if (G_UNLIKELY (bytes_per_sample == 0 || rate == 0)) {
GST_DEBUG ("not enough metadata yet to convert");
goto exit;
}
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / bytes_per_sample;
res = TRUE;
break;
case GST_FORMAT_TIME:
*dest_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, byterate);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * bytes_per_sample;
res = TRUE;
break;
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, rate);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
scale = bytes_per_sample;
/* fallthrough */
case GST_FORMAT_DEFAULT:
*dest_value = gst_util_uint64_scale_int (src_value,
scale * rate, GST_SECOND);
res = TRUE;
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
exit:
return res;
}

81
gst-libs/gst/audio/gstbaseaudioutils.h
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@ -1,81 +0,0 @@
/* GStreamer
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* 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.
*/
#ifndef _GST_BASE_AUDIO_UTILS_H_
#define _GST_BASE_AUDIO_UTILS_H_
#ifndef GST_USE_UNSTABLE_API
#warning "Base audio utils provide unstable API and may change in future."
#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
#endif
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
G_BEGIN_DECLS
/**
* GstAudioFormatInfo:
* @is_int: whether sample data is int or float
* @rate: rate of sample data
* @channels: number of channels in sample data
* @width: width (in bits) of sample data
* @depth: used bits in sample data (if integer)
* @sign: sign of sample data (if integer)
* @endian: endianness of sample data
* @bpf: bytes per audio frame
*/
typedef struct _GstAudioFormatInfo {
gboolean is_int;
gint rate;
gint channels;
gint width;
gint depth;
gboolean sign;
gint endian;
GstAudioChannelPosition *channel_pos;
gint bpf;
} GstAudioFormatInfo;
void gst_base_audio_format_info_init (GstAudioFormatInfo * info);
void gst_base_audio_format_info_clear (GstAudioFormatInfo * info);
GstAudioFormatInfo *gst_base_audio_format_info_new (void);
void gst_base_audio_format_info_free (GstAudioFormatInfo * info);
GstAudioFormatInfo *gst_base_audio_format_info_copy (GstAudioFormatInfo * info);
gboolean gst_base_audio_parse_caps (GstCaps * caps, GstAudioFormatInfo * info);
gboolean gst_base_audio_compare_format_info (GstAudioFormatInfo * from,
GstAudioFormatInfo * to);
GstCaps *gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...);
gboolean gst_base_audio_encoded_audio_convert (GstAudioFormatInfo * fmt,
gint64 bytes, gint64 samples, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
gboolean gst_base_audio_raw_audio_convert (GstAudioFormatInfo * fmt, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
G_END_DECLS
#endif