gstreamer/gst-libs/gst/audio/gstbaseaudioutils.c
2011-08-27 14:47:00 +01:00

316 lines
8.6 KiB
C

/* 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>
#define CHECK_VALUE(var, val) \
G_STMT_START { \
if (!res) \
goto fail; \
if (var != val) \
changed = TRUE; \
var = val; \
} G_STMT_END
/**
* gst_base_audio_parse_caps:
* @caps: a #GstCaps
* @state: a #GstAudioFormatInfo
* @changed: whether @caps introduced a change in current @state
*
* Parses audio format as represented by @caps into a more concise form
* as represented by @state, while checking if for changes to currently
* defined audio format.
*
* Returns: TRUE if parsing succeeded, otherwise FALSE
*/
gboolean
gst_base_audio_parse_caps (GstCaps * caps, GstAudioFormatInfo * state,
gboolean * _changed)
{
gboolean res = TRUE, changed = FALSE;
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);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_has_name (s, "audio/x-raw-int"))
state->is_int = TRUE;
else if (gst_structure_has_name (s, "audio/x-raw-float"))
state->is_int = FALSE;
else
goto fail;
res = gst_structure_get_int (s, "rate", &vi);
CHECK_VALUE (state->rate, vi);
res &= gst_structure_get_int (s, "channels", &vi);
CHECK_VALUE (state->channels, vi);
res &= gst_structure_get_int (s, "width", &vi);
CHECK_VALUE (state->width, vi);
res &= (!state->is_int || gst_structure_get_int (s, "depth", &vi));
CHECK_VALUE (state->depth, vi);
res &= gst_structure_get_int (s, "endianness", &vi);
CHECK_VALUE (state->endian, vi);
res &= (!state->is_int || gst_structure_get_boolean (s, "signed", &vb));
CHECK_VALUE (state->sign, vb);
state->bpf = (state->width / 8) * state->channels;
GST_LOG ("bpf: %d", state->bpf);
if (!state->bpf)
goto fail;
g_free (state->channel_pos);
state->channel_pos = gst_audio_get_channel_positions (s);
if (_changed)
*_changed = changed;
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_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;
}