gstreamer/ext/ffmpeg/gstffmpegaudioresample.c

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* This file:
* Copyright (C) 2005 Luca Ognibene <luogni@tin.it>
* Copyright (C) 2006 Martin Zlomek <martin.zlomek@itonis.tv>
*
* 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
#ifdef HAVE_FFMPEG_UNINSTALLED
#include <avcodec.h>
#else
#include <libavcodec/avcodec.h>
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/video/video.h>
#include "gstffmpeg.h"
#include "gstffmpegcodecmap.h"
typedef struct _GstFFMpegAudioResample
{
GstBaseTransform element;
GstPad *sinkpad, *srcpad;
gint in_rate, out_rate;
gint in_channels, out_channels;
ReSampleContext *res;
} GstFFMpegAudioResample;
typedef struct _GstFFMpegAudioResampleClass
{
GstBaseTransformClass parent_class;
} GstFFMpegAudioResampleClass;
#define GST_TYPE_FFMPEGAUDIORESAMPLE \
(gst_ffmpegaudioresample_get_type())
#define GST_FFMPEGAUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_FFMPEGAUDIORESAMPLE,GstFFMpegAudioResample))
#define GST_FFMPEGAUDIORESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_FFMPEGAUDIORESAMPLE,GstFFMpegAudioResampleClass))
#define GST_IS_FFMPEGAUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_FFMPEGAUDIORESAMPLE))
#define GST_IS_FFMPEGAUDIORESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_FFMPEGAUDIORESAMPLE))
GType gst_ffmpegaudioresample_get_type (void);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS
("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) { 1 , 2 }, rate = (int) [1, MAX ]")
);
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS
("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) [ 1 , 6 ], rate = (int) [1, MAX ]")
);
GST_BOILERPLATE (GstFFMpegAudioResample, gst_ffmpegaudioresample,
GstBaseTransform, GST_TYPE_BASE_TRANSFORM);
static void gst_ffmpegaudioresample_finalize (GObject * object);
static GstCaps *gst_ffmpegaudioresample_transform_caps (GstBaseTransform *
trans, GstPadDirection direction, GstCaps * caps);
static gboolean gst_ffmpegaudioresample_transform_size (GstBaseTransform *
trans, GstPadDirection direction, GstCaps * caps, guint size,
GstCaps * othercaps, guint * othersize);
static gboolean gst_ffmpegaudioresample_get_unit_size (GstBaseTransform * trans,
GstCaps * caps, guint * size);
static gboolean gst_ffmpegaudioresample_set_caps (GstBaseTransform * trans,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_ffmpegaudioresample_transform (GstBaseTransform *
trans, GstBuffer * inbuf, GstBuffer * outbuf);
static void
gst_ffmpegaudioresample_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details_simple (element_class,
"FFMPEG Audio resampling element", "Filter/Converter/Audio",
"Converts audio from one samplerate to another",
"Edward Hervey <bilboed@bilboed.com>");
}
static void
gst_ffmpegaudioresample_class_init (GstFFMpegAudioResampleClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
gobject_class->finalize = gst_ffmpegaudioresample_finalize;
trans_class->transform_caps =
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform_caps);
trans_class->get_unit_size =
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_get_unit_size);
trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_set_caps);
trans_class->transform =
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform);
trans_class->transform_size =
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform_size);
trans_class->passthrough_on_same_caps = TRUE;
}
static void
gst_ffmpegaudioresample_init (GstFFMpegAudioResample * resample,
GstFFMpegAudioResampleClass * klass)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
resample->res = NULL;
}
static void
gst_ffmpegaudioresample_finalize (GObject * object)
{
GstFFMpegAudioResample *resample = GST_FFMPEGAUDIORESAMPLE (object);
if (resample->res != NULL)
audio_resample_close (resample->res);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstCaps *
gst_ffmpegaudioresample_transform_caps (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *retcaps;
GstStructure *struc;
retcaps = gst_caps_copy (caps);
struc = gst_caps_get_structure (retcaps, 0);
gst_structure_set (struc, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
GST_LOG_OBJECT (trans, "returning caps %" GST_PTR_FORMAT, retcaps);
return retcaps;
}
static gboolean
gst_ffmpegaudioresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
gint inrate, outrate;
gint inchanns, outchanns;
GstStructure *ins, *outs;
gboolean ret;
guint64 conv;
ins = gst_caps_get_structure (caps, 0);
outs = gst_caps_get_structure (othercaps, 0);
/* Get input/output sample rate and channels */
ret = gst_structure_get_int (ins, "rate", &inrate);
ret &= gst_structure_get_int (ins, "channels", &inchanns);
ret &= gst_structure_get_int (outs, "rate", &outrate);
ret &= gst_structure_get_int (outs, "channels", &outchanns);
if (!ret)
return FALSE;
conv = gst_util_uint64_scale (size, outrate * outchanns, inrate * inchanns);
/* Adding padding to the output buffer size, since audio_resample's internal
* methods might write a bit further. */
*othersize = (guint) conv + 64;
GST_DEBUG_OBJECT (trans, "Transformed size from %d to %d", size, *othersize);
return TRUE;
}
static gboolean
gst_ffmpegaudioresample_get_unit_size (GstBaseTransform * trans, GstCaps * caps,
guint * size)
{
gint channels;
GstStructure *structure;
gboolean ret;
g_assert (size);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "channels", &channels);
g_return_val_if_fail (ret, FALSE);
*size = 2 * channels;
return TRUE;
}
static gboolean
gst_ffmpegaudioresample_set_caps (GstBaseTransform * trans, GstCaps * incaps,
GstCaps * outcaps)
{
GstFFMpegAudioResample *resample = GST_FFMPEGAUDIORESAMPLE (trans);
GstStructure *instructure = gst_caps_get_structure (incaps, 0);
GstStructure *outstructure = gst_caps_get_structure (outcaps, 0);
GST_LOG_OBJECT (resample, "incaps:%" GST_PTR_FORMAT, incaps);
GST_LOG_OBJECT (resample, "outcaps:%" GST_PTR_FORMAT, outcaps);
if (!gst_structure_get_int (instructure, "channels", &resample->in_channels))
return FALSE;
if (!gst_structure_get_int (instructure, "rate", &resample->in_rate))
return FALSE;
if (!gst_structure_get_int (outstructure, "channels",
&resample->out_channels))
return FALSE;
if (!gst_structure_get_int (outstructure, "rate", &resample->out_rate))
return FALSE;
/* FIXME : Allow configuring the various resampling properties */
#define TAPS 16
resample->res =
av_audio_resample_init (resample->out_channels, resample->in_channels,
resample->out_rate, resample->in_rate,
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16, TAPS, 10, 0, 0.8);
if (resample->res == NULL)
return FALSE;
return TRUE;
}
static GstFlowReturn
gst_ffmpegaudioresample_transform (GstBaseTransform * trans, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFFMpegAudioResample *resample = GST_FFMPEGAUDIORESAMPLE (trans);
gint nbsamples;
gint ret;
gst_buffer_copy_metadata (outbuf, inbuf, GST_BUFFER_COPY_TIMESTAMPS);
nbsamples = GST_BUFFER_SIZE (inbuf) / (2 * resample->in_channels);
GST_LOG_OBJECT (resample, "input buffer duration:%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
GST_DEBUG_OBJECT (resample,
"audio_resample(ctx, output:%p [size:%d], input:%p [size:%d], nbsamples:%d",
GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf),
GST_BUFFER_DATA (inbuf), GST_BUFFER_SIZE (inbuf), nbsamples);
ret = audio_resample (resample->res, (short *) GST_BUFFER_DATA (outbuf),
(short *) GST_BUFFER_DATA (inbuf), nbsamples);
GST_DEBUG_OBJECT (resample, "audio_resample returned %d", ret);
GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (ret, GST_SECOND,
resample->out_rate);
GST_BUFFER_SIZE (outbuf) = ret * 2 * resample->out_channels;
GST_LOG_OBJECT (resample, "Output buffer duration:%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
return GST_FLOW_OK;
}
gboolean
gst_ffmpegaudioresample_register (GstPlugin * plugin)
{
return gst_element_register (plugin, "ffaudioresample",
GST_RANK_NONE, GST_TYPE_FFMPEGAUDIORESAMPLE);
}