gstreamer/gst/speexresample/gstspeexresample.c
Julien Moutte 0625160416 configure.ac: Add QuickTime Wrapper plug-in.
Original commit message from CVS:
2007-11-26  Julien Moutte  <julien@fluendo.com>

* configure.ac: Add QuickTime Wrapper plug-in.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
build on Mac OS X Leopard. Incorrect printf format arguments.
* sys/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/qtwrapper/audiodecoders.c:
(qtwrapper_audio_decoder_base_init),
(qtwrapper_audio_decoder_class_init),
(qtwrapper_audio_decoder_init),
(clear_AudioStreamBasicDescription), (fill_indesc_mp3),
(fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
(make_samr_magic_cookie), (open_decoder),
(qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
(qtwrapper_audio_decoder_chain),
(qtwrapper_audio_decoder_sink_event),
(qtwrapper_audio_decoders_register):
* sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
(fourcc_to_caps):
* sys/qtwrapper/codecmapping.h:
* sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
(image_description_for_mp4v), (image_description_from_stsd_buffer),
(image_description_from_codec_data):
* sys/qtwrapper/imagedescription.h:
* sys/qtwrapper/qtutils.c: (get_name_info_from_component),
(get_output_info_from_component), (dump_avcc_atom),
(dump_image_description), (dump_codec_decompress_params),
(addSInt32ToDictionary), (dump_cvpixel_buffer),
(DestroyAudioBufferList), (AllocateAudioBufferList):
* sys/qtwrapper/qtutils.h:
* sys/qtwrapper/qtwrapper.c: (plugin_init):
* sys/qtwrapper/qtwrapper.h:
* sys/qtwrapper/videodecoders.c:
(qtwrapper_video_decoder_base_init),
(qtwrapper_video_decoder_class_init),
(qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
(fill_image_description), (new_image_description), (close_decoder),
(open_decoder), (qtwrapper_video_decoder_sink_setcaps),
(decompressCb), (qtwrapper_video_decoder_chain),
(qtwrapper_video_decoder_sink_event),
(qtwrapper_video_decoders_register): Initial import of QuickTime
wrapper jointly developped by Songbird authors (Pioneers of the
Inevitable) and Fluendo.
2007-11-26 13:19:46 +00:00

994 lines
30 KiB
C

/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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.
*/
/**
* SECTION:element-speexresample
*
* <refsect2>
* speexresample resamples raw audio buffers to different sample rates using
* a configurable windowing function to enhance quality.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! speexresample ! audio/x-raw-int, rate=8000 ! alsasink
* </programlisting>
* Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* </para>
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include "gstspeexresample.h"
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
GST_DEBUG_CATEGORY (speex_resample_debug);
#define GST_CAT_DEFAULT speex_resample_debug
enum
{
PROP_0,
PROP_QUALITY
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS ( \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (boolean) true" \
)
static GstStaticPadTemplate gst_speex_resample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static GstStaticPadTemplate gst_speex_resample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_speex_resample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_speex_resample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
static gboolean gst_speex_resample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, guint * size);
static GstCaps *gst_speex_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps);
static gboolean gst_speex_resample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * incaps, guint insize,
GstCaps * outcaps, guint * outsize);
static gboolean gst_speex_resample_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_speex_resample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean gst_speex_resample_event (GstBaseTransform * base,
GstEvent * event);
static gboolean gst_speex_resample_start (GstBaseTransform * base);
static gboolean gst_speex_resample_stop (GstBaseTransform * base);
static gboolean gst_speex_resample_query (GstPad * pad, GstQuery * query);
static const GstQueryType *gst_speex_resample_query_type (GstPad * pad);
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (speex_resample_debug, "speex_resample", 0, "audio resampling element");
GST_BOILERPLATE_FULL (GstSpeexResample, gst_speex_resample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
static void
gst_speex_resample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_speex_resample_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_speex_resample_sink_template));
gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
"Filter/Converter/Audio", "Resamples audio",
"Sebastian Dröge <slomo@circular-chaos.org>");
}
static void
gst_speex_resample_class_init (GstSpeexResampleClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_speex_resample_set_property;
gobject_class->get_property = gst_speex_resample_get_property;
g_object_class_install_property (gobject_class, PROP_QUALITY,
g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
"the lowest and 10 being the best",
SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX,
SPEEX_RESAMPLER_QUALITY_DEFAULT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
GST_BASE_TRANSFORM_CLASS (klass)->start =
GST_DEBUG_FUNCPTR (gst_speex_resample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
GST_DEBUG_FUNCPTR (gst_speex_resample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
GST_DEBUG_FUNCPTR (gst_speex_resample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
GST_DEBUG_FUNCPTR (gst_speex_resample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
GST_DEBUG_FUNCPTR (gst_speex_resample_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
GST_DEBUG_FUNCPTR (gst_speex_resample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
GST_DEBUG_FUNCPTR (gst_speex_resample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->event =
GST_DEBUG_FUNCPTR (gst_speex_resample_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void
gst_speex_resample_init (GstSpeexResample * resample,
GstSpeexResampleClass * klass)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
resample->need_discont = FALSE;
gst_pad_set_query_function (trans->srcpad, gst_speex_resample_query);
gst_pad_set_query_type_function (trans->srcpad,
gst_speex_resample_query_type);
}
/* vmethods */
static gboolean
gst_speex_resample_start (GstBaseTransform * base)
{
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
resample->ts_offset = -1;
resample->offset = -1;
resample->next_ts = -1;
return TRUE;
}
static gboolean
gst_speex_resample_stop (GstBaseTransform * base)
{
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
if (resample->state) {
resample_resampler_destroy (resample->state);
resample->state = NULL;
}
gst_caps_replace (&resample->sinkcaps, NULL);
gst_caps_replace (&resample->srccaps, NULL);
return TRUE;
}
static gboolean
gst_speex_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size)
{
gint width, channels;
GstStructure *structure;
gboolean ret;
g_return_val_if_fail (size != NULL, FALSE);
/* this works for both float and int */
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "channels", &channels);
g_return_val_if_fail (ret, FALSE);
*size = width * channels / 8;
return TRUE;
}
static GstCaps *
gst_speex_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *res;
GstStructure *structure;
/* transform caps gives one single caps so we can just replace
* the rate property with our range. */
res = gst_caps_copy (caps);
structure = gst_caps_get_structure (res, 0);
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
return res;
}
static SpeexResamplerState *
gst_speex_resample_init_state (guint channels, guint inrate, guint outrate,
guint quality, gboolean fp)
{
SpeexResamplerState *ret = NULL;
gint err = RESAMPLER_ERR_SUCCESS;
if (fp)
ret =
resample_float_resampler_init (channels, inrate, outrate, quality,
&err);
else
ret =
resample_int_resampler_init (channels, inrate, outrate, quality, &err);
if (err != RESAMPLER_ERR_SUCCESS) {
GST_ERROR ("Failed to create resampler state: %s",
resample_resampler_strerror (err));
return NULL;
}
if (fp)
resample_float_resampler_skip_zeros (ret);
else
resample_int_resampler_skip_zeros (ret);
return ret;
}
static gboolean
gst_speex_resample_update_state (GstSpeexResample * resample, gint channels,
gint inrate, gint outrate, gint quality, gboolean fp)
{
gboolean ret = TRUE;
gboolean updated_latency = FALSE;
updated_latency = (resample->inrate != inrate
|| quality != resample->quality) && resample->state != NULL;
if (resample->state == NULL) {
ret = TRUE;
} else if (resample->channels != channels || fp != resample->fp) {
resample_resampler_destroy (resample->state);
resample->state =
gst_speex_resample_init_state (channels, inrate, outrate, quality, fp);
ret = (resample->state != NULL);
} else if (resample->inrate != inrate || resample->outrate != outrate) {
gint err = RESAMPLER_ERR_SUCCESS;
if (fp)
err =
resample_float_resampler_set_rate (resample->state, inrate, outrate);
else
err = resample_int_resampler_set_rate (resample->state, inrate, outrate);
if (err != RESAMPLER_ERR_SUCCESS)
GST_ERROR ("Failed to update rate: %s",
resample_resampler_strerror (err));
ret = (err == RESAMPLER_ERR_SUCCESS);
} else if (quality != resample->quality) {
gint err = RESAMPLER_ERR_SUCCESS;
if (fp)
err = resample_float_resampler_set_quality (resample->state, quality);
else
err = resample_int_resampler_set_quality (resample->state, quality);
if (err != RESAMPLER_ERR_SUCCESS)
GST_ERROR ("Failed to update quality: %s",
resample_resampler_strerror (err));
ret = (err == RESAMPLER_ERR_SUCCESS);
}
resample->channels = channels;
resample->fp = fp;
resample->quality = quality;
resample->inrate = inrate;
resample->outrate = outrate;
if (updated_latency)
gst_element_post_message (GST_ELEMENT (resample),
gst_message_new_latency (GST_OBJECT (resample)));
return ret;
}
static void
gst_speex_resample_reset_state (GstSpeexResample * resample)
{
if (resample->state && resample->fp)
resample_float_resampler_reset_mem (resample->state);
else if (resample->state && !resample->fp)
resample_int_resampler_reset_mem (resample->state);
}
static gboolean
gst_speex_resample_parse_caps (GstCaps * incaps,
GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate,
gboolean * fp)
{
GstStructure *structure;
gboolean ret;
gint myinrate, myoutrate, mychannels;
gboolean myfp;
GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
structure = gst_caps_get_structure (incaps, 0);
if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float"))
myfp = TRUE;
else
myfp = FALSE;
ret = gst_structure_get_int (structure, "rate", &myinrate);
ret &= gst_structure_get_int (structure, "channels", &mychannels);
if (!ret)
goto no_in_rate_channels;
structure = gst_caps_get_structure (outcaps, 0);
ret = gst_structure_get_int (structure, "rate", &myoutrate);
if (!ret)
goto no_out_rate;
if (channels)
*channels = mychannels;
if (inrate)
*inrate = myinrate;
if (outrate)
*outrate = myoutrate;
if (fp)
*fp = myfp;
return TRUE;
/* ERRORS */
no_in_rate_channels:
{
GST_DEBUG ("could not get input rate and channels");
return FALSE;
}
no_out_rate:
{
GST_DEBUG ("could not get output rate");
return FALSE;
}
}
static gboolean
gst_speex_resample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
SpeexResamplerState *state;
GstCaps *srccaps, *sinkcaps;
gboolean use_internal = FALSE; /* whether we use the internal state */
gboolean ret = TRUE;
guint32 ratio_den, ratio_num;
gboolean fp;
GST_LOG ("asked to transform size %d in direction %s",
size, direction == GST_PAD_SINK ? "SINK" : "SRC");
if (direction == GST_PAD_SINK) {
sinkcaps = caps;
srccaps = othercaps;
} else {
sinkcaps = othercaps;
srccaps = caps;
}
/* if the caps are the ones that _set_caps got called with; we can use
* our own state; otherwise we'll have to create a state */
if (resample->state && gst_caps_is_equal (sinkcaps, resample->sinkcaps) &&
gst_caps_is_equal (srccaps, resample->srccaps)) {
use_internal = TRUE;
state = resample->state;
fp = resample->fp;
} else {
gint inrate, outrate, channels;
GST_DEBUG ("Can't use internal state, creating state");
ret =
gst_speex_resample_parse_caps (caps, othercaps, &channels, &inrate,
&outrate, &fp);
if (!ret) {
GST_ERROR ("Wrong caps");
return FALSE;
}
state = gst_speex_resample_init_state (channels, inrate, outrate, 0, TRUE);
if (!state)
return FALSE;
}
if (resample->fp || use_internal)
resample_float_resampler_get_ratio (state, &ratio_num, &ratio_den);
else
resample_int_resampler_get_ratio (state, &ratio_num, &ratio_den);
if (direction == GST_PAD_SINK) {
gint fac = (fp) ? 4 : 2;
/* asked to convert size of an incoming buffer */
size /= fac;
*othersize = (size * ratio_den + (ratio_num >> 1)) / ratio_num;
*othersize *= fac;
size *= fac;
} else {
gint fac = (fp) ? 4 : 2;
/* asked to convert size of an outgoing buffer */
size /= fac;
*othersize = (size * ratio_num + (ratio_den >> 1)) / ratio_den;
*othersize *= fac;
size *= fac;
}
GST_LOG ("transformed size %d to %d", size, *othersize);
if (!use_internal)
resample_resampler_destroy (state);
return ret;
}
static gboolean
gst_speex_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
gint inrate = 0, outrate = 0, channels = 0;
gboolean fp = FALSE;
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
ret = gst_speex_resample_parse_caps (incaps, outcaps,
&channels, &inrate, &outrate, &fp);
g_return_val_if_fail (ret, FALSE);
ret =
gst_speex_resample_update_state (resample, channels, inrate, outrate,
resample->quality, fp);
g_return_val_if_fail (ret, FALSE);
/* save caps so we can short-circuit in the size_transform if the caps
* are the same */
gst_caps_replace (&resample->sinkcaps, incaps);
gst_caps_replace (&resample->srccaps, outcaps);
return TRUE;
}
static void
gst_speex_resample_push_drain (GstSpeexResample * resample)
{
GstBuffer *buf;
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
GstFlowReturn res;
gint outsize;
guint out_len, out_processed;
gint err;
if (!resample->state)
return;
if (resample->fp) {
guint num, den;
resample_float_resampler_get_ratio (resample->state, &num, &den);
out_len = resample_float_resampler_get_input_latency (resample->state);
out_len = out_processed = (out_len * den + (num >> 1)) / num;
outsize = 4 * out_len * resample->channels;
} else {
guint num, den;
resample_int_resampler_get_ratio (resample->state, &num, &den);
out_len = resample_int_resampler_get_input_latency (resample->state);
out_len = out_processed = (out_len * den + (num >> 1)) / num;
outsize = 2 * out_len * resample->channels;
}
res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (trans->srcpad), &buf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING ("failed allocating buffer of %d bytes", outsize);
return;
}
if (resample->fp) {
guint len = resample_float_resampler_get_input_latency (resample->state);
err =
resample_float_resampler_process_interleaved_float (resample->state,
NULL, &len, (gfloat *) GST_BUFFER_DATA (buf), &out_processed);
} else {
guint len = resample_int_resampler_get_input_latency (resample->state);
err =
resample_int_resampler_process_interleaved_int (resample->state, NULL,
&len, (gint16 *) GST_BUFFER_DATA (buf), &out_processed);
}
if (err != RESAMPLER_ERR_SUCCESS) {
GST_WARNING ("Failed to process drain: %s",
resample_resampler_strerror (err));
gst_buffer_unref (buf);
return;
}
if (out_processed == 0) {
GST_WARNING ("Failed to get drain, dropping buffer");
gst_buffer_unref (buf);
return;
}
GST_BUFFER_OFFSET (buf) = resample->offset;
GST_BUFFER_TIMESTAMP (buf) = resample->next_ts;
GST_BUFFER_SIZE (buf) =
out_processed * resample->channels * ((resample->fp) ? 4 : 2);
if (resample->ts_offset != -1) {
resample->offset += out_processed;
resample->ts_offset += out_processed;
resample->next_ts =
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
GST_BUFFER_OFFSET_END (buf) = resample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (buf) = resample->next_ts - GST_BUFFER_TIMESTAMP (buf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (buf) =
GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
}
GST_LOG ("Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT
" duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT
" offset_end %" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
res = gst_pad_push (trans->srcpad, buf);
if (res != GST_FLOW_OK)
GST_WARNING ("Failed to push drain");
return;
}
static gboolean
gst_speex_resample_event (GstBaseTransform * base, GstEvent * event)
{
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
gst_speex_resample_reset_state (resample);
resample->ts_offset = -1;
resample->next_ts = -1;
resample->offset = -1;
case GST_EVENT_NEWSEGMENT:
gst_speex_resample_push_drain (resample);
gst_speex_resample_reset_state (resample);
resample->ts_offset = -1;
resample->next_ts = -1;
resample->offset = -1;
break;
case GST_EVENT_EOS:{
gst_speex_resample_push_drain (resample);
gst_speex_resample_reset_state (resample);
break;
}
default:
break;
}
parent_class->event (base, event);
return TRUE;
}
static gboolean
gst_speex_resample_check_discont (GstSpeexResample * resample,
GstClockTime timestamp)
{
if (timestamp != GST_CLOCK_TIME_NONE &&
resample->prev_ts != GST_CLOCK_TIME_NONE &&
resample->prev_duration != GST_CLOCK_TIME_NONE &&
timestamp != resample->prev_ts + resample->prev_duration) {
/* Potentially a discontinuous buffer. However, it turns out that many
* elements generate imperfect streams due to rounding errors, so we permit
* a small error (up to one sample) without triggering a filter
* flush/restart (if triggered incorrectly, this will be audible) */
GstClockTimeDiff diff = timestamp -
(resample->prev_ts + resample->prev_duration);
if (ABS (diff) > GST_SECOND / resample->inrate) {
GST_WARNING ("encountered timestamp discontinuity of %" G_GINT64_FORMAT,
diff);
return TRUE;
}
}
return FALSE;
}
static void
gst_speex_fix_output_buffer (GstSpeexResample * resample, GstBuffer * outbuf,
guint diff)
{
GstClockTime timediff = GST_FRAMES_TO_CLOCK_TIME (diff, resample->outrate);
GST_LOG ("Adjusting buffer by %d samples", diff);
GST_BUFFER_DURATION (outbuf) -= timediff;
GST_BUFFER_SIZE (outbuf) -=
diff * ((resample->fp) ? 4 : 2) * resample->channels;
if (resample->ts_offset != -1) {
GST_BUFFER_OFFSET_END (outbuf) -= diff;
resample->offset -= diff;
resample->ts_offset -= diff;
resample->next_ts =
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
}
}
static GstFlowReturn
gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
GstBuffer * outbuf)
{
guint32 in_len, in_processed;
guint32 out_len, out_processed;
gint err = RESAMPLER_ERR_SUCCESS;
in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
if (resample->fp) {
in_len /= 4;
out_len /= 4;
} else {
in_len /= 2;
out_len /= 2;
}
in_processed = in_len;
out_processed = out_len;
if (resample->fp)
err = resample_float_resampler_process_interleaved_float (resample->state,
(const gfloat *) GST_BUFFER_DATA (inbuf), &in_processed,
(gfloat *) GST_BUFFER_DATA (outbuf), &out_processed);
else
err = resample_int_resampler_process_interleaved_int (resample->state,
(const gint16 *) GST_BUFFER_DATA (inbuf), &in_processed,
(gint16 *) GST_BUFFER_DATA (outbuf), &out_processed);
if (in_len != in_processed)
GST_WARNING ("Converted %d of %d input samples", in_processed, in_len);
if (out_len != out_processed) {
/* One sample difference is allowed as this will happen
* because of rounding errors */
if (out_processed == 0) {
GST_DEBUG ("Converted to 0 samples, buffer dropped");
if (resample->ts_offset != -1) {
GST_BUFFER_OFFSET_END (outbuf) -= out_len;
resample->offset -= out_len;
resample->ts_offset -= out_len;
resample->next_ts =
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
}
return GST_BASE_TRANSFORM_FLOW_DROPPED;
} else if (out_len - out_processed != 1)
GST_WARNING ("Converted to %d instead of %d output samples",
out_processed, out_len);
if (out_len > out_processed) {
gst_speex_fix_output_buffer (resample, outbuf, out_len - out_processed);
} else {
GST_ERROR ("Wrote more output than allocated!");
return GST_FLOW_ERROR;
}
}
if (err != RESAMPLER_ERR_SUCCESS) {
GST_ERROR ("Failed to convert data: %s", resample_resampler_strerror (err));
return GST_FLOW_ERROR;
} else {
GST_LOG ("Converted to buffer of %u bytes with timestamp %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
", offset_end %" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
return GST_FLOW_OK;
}
}
static GstFlowReturn
gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
guint8 *data;
gulong size;
GstClockTime timestamp;
gint outsamples;
if (resample->state == NULL)
if (!(resample->state = gst_speex_resample_init_state (resample->channels,
resample->inrate, resample->outrate, resample->quality,
resample->fp)))
return GST_FLOW_ERROR;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
GST_LOG ("transforming buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
size, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
/* check for timestamp discontinuities and flush/reset if needed */
if (G_UNLIKELY (gst_speex_resample_check_discont (resample, timestamp)
|| GST_BUFFER_IS_DISCONT (inbuf))) {
/* Flush internal samples */
gst_speex_resample_reset_state (resample);
/* Inform downstream element about discontinuity */
resample->need_discont = TRUE;
/* We want to recalculate the offset */
resample->ts_offset = -1;
}
outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels;
outsamples /= (resample->fp) ? 4 : 2;
if (resample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
* input timestamp. */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
GstClockTime stime;
/* offset used to calculate the timestamps. We use the sample offset for
* this to make it more accurate. We want the first buffer to have the
* same timestamp as the incoming timestamp. */
resample->next_ts = timestamp;
resample->ts_offset =
GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
/* offset used to set as the buffer offset, this offset is always
* relative to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
resample->offset = GST_CLOCK_TIME_TO_FRAMES (stime, resample->outrate);
}
}
resample->prev_ts = timestamp;
resample->prev_duration = GST_BUFFER_DURATION (inbuf);
GST_BUFFER_OFFSET (outbuf) = resample->offset;
GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
if (resample->ts_offset != -1) {
resample->offset += outsamples;
resample->ts_offset += outsamples;
resample->next_ts =
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
GST_BUFFER_OFFSET_END (outbuf) = resample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (outbuf) = resample->next_ts -
GST_BUFFER_TIMESTAMP (outbuf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (outbuf) =
GST_FRAMES_TO_CLOCK_TIME (outsamples, resample->outrate);
}
if (G_UNLIKELY (resample->need_discont)) {
GST_DEBUG ("marking this buffer with the DISCONT flag");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
resample->need_discont = FALSE;
}
return gst_speex_resample_process (resample, inbuf, outbuf);
}
static gboolean
gst_speex_resample_query (GstPad * pad, GstQuery * query)
{
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (gst_pad_get_parent (pad));
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
GstPad *peer;
gint rate = resample->inrate;
gint resampler_latency;
if (resample->state && resample->fp)
resampler_latency =
resample_float_resampler_get_input_latency (resample->state);
else if (resample->state && !resample->fp)
resampler_latency =
resample_int_resampler_get_input_latency (resample->state);
else
resampler_latency = 0;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
if ((peer = gst_pad_get_peer (trans->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG ("Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
if (rate != 0 && resampler_latency != 0)
latency =
gst_util_uint64_scale (resampler_latency, GST_SECOND, rate);
else
latency = 0;
GST_DEBUG ("Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
GST_DEBUG ("Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (resample);
return res;
}
static const GstQueryType *
gst_speex_resample_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
0
};
return types;
}
static void
gst_speex_resample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSpeexResample *resample;
resample = GST_SPEEX_RESAMPLE (object);
switch (prop_id) {
case PROP_QUALITY:
resample->quality = g_value_get_int (value);
GST_DEBUG ("new quality %d", resample->quality);
gst_speex_resample_update_state (resample, resample->channels,
resample->inrate, resample->outrate, resample->quality, resample->fp);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_speex_resample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSpeexResample *resample;
resample = GST_SPEEX_RESAMPLE (object);
switch (prop_id) {
case PROP_QUALITY:
g_value_set_int (value, resample->quality);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "speexresample", GST_RANK_NONE,
GST_TYPE_SPEEX_RESAMPLE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"speexresample",
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);