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baseaudio: add audioutils for caps and query handling helper utils

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This commit is contained in:
Mark Nauwelaerts 2011-03-15 17:27:42 +01:00 committed by Tim-Philipp Müller
parent cb04eaaa8f
commit 8c61685554
4 changed files with 407 additions and 249 deletions
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237
gst-libs/gst/audio/gstbaseaudioencoder.c
View file

@ -150,12 +150,13 @@
# include "config.h"
#endif
#include "gstbaseaudioencoder.h"
#include <gst/base/gstadapter.h>
#include <gst/audio/audio.h>
#include <stdlib.h>
#include <string.h>
#include "gstbaseaudioencoder.h"
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (gst_base_audio_encoder_debug);
#define GST_CAT_DEFAULT gst_base_audio_encoder_debug
@ -318,7 +319,7 @@ gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass * klass)
DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TOLERANCE,
g_param_spec_int64 ("tolerance", "Tolerance",
"Consider discontinuity if timestamp jitter/imperfection exceeds tolerance",
"Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
0, G_MAXINT64, DEFAULT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
@ -887,13 +888,6 @@ wrong_buffer:
}
}
#define CHECK_VALUE(res, var, val) \
if (!res) \
goto refuse_caps; \
if (var != val) \
changed = TRUE; \
var = val;
static gboolean
gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
{
@ -902,9 +896,6 @@ gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
GstBaseAudioEncoderContext *ctx;
GstAudioState *state;
gboolean res = TRUE, changed = FALSE;
GstStructure *s;
gboolean vb;
gint vi;
enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
@ -920,36 +911,9 @@ gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
if (!gst_caps_is_fixed (caps))
goto refuse_caps;
s = gst_caps_get_structure (caps, 0);
/* parse caps here to save subclass the trouble */
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
if (!gst_base_audio_parse_caps (caps, state, &changed))
goto refuse_caps;
res = gst_structure_get_int (s, "rate", &vi);
CHECK_VALUE (res, state->rate, vi);
res &= gst_structure_get_int (s, "channels", &vi);
CHECK_VALUE (res, state->channels, vi);
res &= gst_structure_get_int (s, "width", &vi);
CHECK_VALUE (res, state->width, vi);
res &= (!state->is_int || gst_structure_get_int (s, "depth", &vi));
CHECK_VALUE (res, state->depth, vi);
res &= gst_structure_get_int (s, "endianness", &vi);
CHECK_VALUE (res, state->endian, vi);
res &= (!state->is_int || gst_structure_get_boolean (s, "signed", &vb));
CHECK_VALUE (res, state->sign, vb);
state->bpf = (state->width / 8) * state->channels;
GST_LOG_OBJECT (enc, "bpf: %d", state->bpf);
if (!state->bpf)
goto refuse_caps;
g_free (state->channel_pos);
state->channel_pos = gst_audio_get_channel_positions (s);
if (changed) {
GstClockTime old_min_latency;
GstClockTime old_max_latency;
@ -1185,83 +1149,13 @@ gst_base_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
return ret;
}
static gboolean
gst_base_audio_encoder_convert_sink (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
GstBaseAudioEncoder *enc;
gboolean res = FALSE;
guint scale = 1;
gint bytes_per_sample, rate, byterate;
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
bytes_per_sample = enc->ctx->state.bpf;
rate = enc->ctx->state.rate;
byterate = bytes_per_sample * rate;
if (G_UNLIKELY (bytes_per_sample == 0 || rate == 0)) {
GST_DEBUG_OBJECT (enc, "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:
gst_object_unref (enc);
return res;
}
static gboolean
gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
GstBaseAudioEncoder *enc;
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_FORMATS:
@ -1277,9 +1171,8 @@ gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res =
gst_base_audio_encoder_convert_sink (pad, src_fmt, src_val,
&dest_fmt, &dest_val)))
if (!(res = gst_base_audio_raw_audio_convert (&enc->ctx->state,
src_fmt, src_val, &dest_fmt, &dest_val)))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
@ -1290,53 +1183,6 @@ gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
}
error:
return res;
}
static gboolean
gst_base_audio_encoder_convert_src (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
GstBaseAudioEncoder *enc;
gboolean res = FALSE;
gint64 avg;
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
if (enc->priv->samples_in == 0 ||
enc->priv->bytes_out == 0 || enc->ctx->state.rate == 0) {
GST_DEBUG_OBJECT (enc, "not enough metadata yet to convert");
goto exit;
}
avg = (enc->priv->bytes_out * enc->ctx->state.rate) / (enc->priv->samples_in);
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, avg);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = gst_util_uint64_scale_int (src_value, avg, GST_SECOND);
res = TRUE;
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
exit:
gst_object_unref (enc);
return res;
}
@ -1377,7 +1223,7 @@ gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
GstFormat fmt, req_fmt;
gint64 pos, val;
if ((res = gst_pad_peer_query (pad, query))) {
if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
GST_LOG_OBJECT (enc, "returning peer response");
break;
}
@ -1402,7 +1248,7 @@ gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
GstFormat fmt, req_fmt;
gint64 dur, val;
if ((res = gst_pad_peer_query (pad, query))) {
if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
GST_LOG_OBJECT (enc, "returning peer response");
break;
}
@ -1434,7 +1280,8 @@ gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res = gst_base_audio_encoder_convert_src (pad, src_fmt, src_val,
if (!(res = gst_base_audio_encoded_audio_convert (&enc->ctx->state,
enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
&dest_fmt, &dest_val)))
break;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
@ -1442,7 +1289,7 @@ gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
}
case GST_QUERY_LATENCY:
{
if ((res = gst_pad_peer_query (pad, query))) {
if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
gboolean live;
GstClockTime min_latency, max_latency;
@ -1461,6 +1308,7 @@ gst_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
@ -1577,54 +1425,3 @@ gst_base_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
gst_object_unref (enc);
return result;
}
/**
* gst_base_audio_encoder_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_encoder_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;
}

30
gst-libs/gst/audio/gstbaseaudioencoder.h
View file

@ -28,8 +28,7 @@
#endif
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/audio/multichannel.h>
#include <gst/audio/gstbaseaudioutils.h>
G_BEGIN_DECLS
@ -91,30 +90,6 @@ typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass;
typedef struct _GstBaseAudioEncoderPrivate GstBaseAudioEncoderPrivate;
typedef struct _GstBaseAudioEncoderContext GstBaseAudioEncoderContext;
/**
* GstAudioState:
* @xint: 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: rate of sample data (if integer)
* @endian: endianness of sample data
* @bpf: bytes per audio frame
*/
typedef struct _GstAudioState {
gboolean is_int;
gint rate;
gint channels;
gint width;
gint depth;
gboolean sign;
gint endian;
GstAudioChannelPosition *channel_pos;
gint bpf;
} GstAudioState;
/**
* GstBaseAudioEncoderContext:
* @state: a #GstAudioState describing input audio format
@ -244,9 +219,6 @@ GstFlowReturn gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc,
GstCaps * gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc,
GstCaps * caps);
GstCaps * gst_base_audio_encoder_add_streamheader (GstCaps * caps,
GstBuffer * buf, ...);
G_END_DECLS
#endif /* __GST_BASE_AUDIO_ENCODER_H__ */

315
gst-libs/gst/audio/gstbaseaudioutils.c Normal file
View file

@ -0,0 +1,315 @@
/* 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 #GstAudioState
* @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, GstAudioState * 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 (GstAudioState * 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 (GstAudioState * 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;
}

74
gst-libs/gst/audio/gstbaseaudioutils.h Normal file
View file

@ -0,0 +1,74 @@
/* 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
/**
* GstAudioState:
* @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 _GstAudioState {
gboolean is_int;
gint rate;
gint channels;
gint width;
gint depth;
gboolean sign;
gint endian;
GstAudioChannelPosition *channel_pos;
gint bpf;
} GstAudioState;
gboolean gst_base_audio_parse_caps (GstCaps * caps,
GstAudioState * state, gboolean * changed);
GstCaps *gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...);
gboolean gst_base_audio_encoded_audio_convert (GstAudioState * fmt,
gint64 bytes, gint64 samples, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
gboolean gst_base_audio_raw_audio_convert (GstAudioState * fmt, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
G_END_DECLS
#endif