diff options
Diffstat (limited to 'gst/audioresample/gstaudioresample.c')
| -rw-r--r-- | gst/audioresample/gstaudioresample.c | 860 |
1 files changed, 0 insertions, 860 deletions
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c deleted file mode 100644 index 4f6f85e0..00000000 --- a/gst/audioresample/gstaudioresample.c +++ /dev/null @@ -1,860 +0,0 @@ -/* GStreamer - * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu> - * Copyright (C) 2003,2004 David A. Schleef <ds@schleef.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. - */ -/* Element-Checklist-Version: 5 */ - -/** - * SECTION:element-legacyresample - * - * legacyresample resamples raw audio buffers to different sample rates using - * a configurable windowing function to enhance quality. - * - * <refsect2> - * <title>Example launch line</title> - * |[ - * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! legacyresample ! audio/x-raw-int, rate=8000 ! alsasink - * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa. - * To create the Ogg/Vorbis file refer to the documentation of vorbisenc. - * </refsect2> - * - * Last reviewed on 2006-03-02 (0.10.4) - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <string.h> -#include <math.h> - -/*#define DEBUG_ENABLED */ -#include "gstaudioresample.h" -#include <gst/audio/audio.h> -#include <gst/base/gstbasetransform.h> - -GST_DEBUG_CATEGORY_STATIC (audioresample_debug); -#define GST_CAT_DEFAULT audioresample_debug - -/* elementfactory information */ -static const GstElementDetails gst_audioresample_details = -GST_ELEMENT_DETAILS ("Audio scaler", - "Filter/Converter/Audio", - "Resample audio", - "David Schleef <ds@schleef.org>"); - -#define DEFAULT_FILTERLEN 16 - -enum -{ - PROP_0, - PROP_FILTERLEN -}; - -#define SUPPORTED_CAPS \ -GST_STATIC_CAPS ( \ - "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;" \ - "audio/x-raw-int, " \ - "rate = (int) [ 1, MAX ], " \ - "channels = (int) [ 1, MAX ], " \ - "endianness = (int) BYTE_ORDER, " \ - "width = (int) 32, " \ - "depth = (int) 32, " \ - "signed = (boolean) true;" \ - "audio/x-raw-float, " \ - "rate = (int) [ 1, MAX ], " \ - "channels = (int) [ 1, MAX ], " \ - "endianness = (int) BYTE_ORDER, " \ - "width = (int) 32; " \ - "audio/x-raw-float, " \ - "rate = (int) [ 1, MAX ], " \ - "channels = (int) [ 1, MAX ], " \ - "endianness = (int) BYTE_ORDER, " \ - "width = (int) 64" \ -) - -static GstStaticPadTemplate gst_audioresample_sink_template = -GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS); - -static GstStaticPadTemplate gst_audioresample_src_template = -GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS); - -static void gst_audioresample_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_audioresample_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec); - -/* vmethods */ -static gboolean audioresample_get_unit_size (GstBaseTransform * base, - GstCaps * caps, guint * size); -static GstCaps *audioresample_transform_caps (GstBaseTransform * base, - GstPadDirection direction, GstCaps * caps); -static void audioresample_fixate_caps (GstBaseTransform * base, - GstPadDirection direction, GstCaps * caps, GstCaps * othercaps); -static gboolean audioresample_transform_size (GstBaseTransform * trans, - GstPadDirection direction, GstCaps * incaps, guint insize, - GstCaps * outcaps, guint * outsize); -static gboolean audioresample_set_caps (GstBaseTransform * base, - GstCaps * incaps, GstCaps * outcaps); -static GstFlowReturn audioresample_pushthrough (GstAudioresample * - audioresample); -static GstFlowReturn audioresample_transform (GstBaseTransform * base, - GstBuffer * inbuf, GstBuffer * outbuf); -static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event); -static gboolean audioresample_start (GstBaseTransform * base); -static gboolean audioresample_stop (GstBaseTransform * base); - -static gboolean audioresample_query (GstPad * pad, GstQuery * query); -static const GstQueryType *audioresample_query_type (GstPad * pad); - -#define DEBUG_INIT(bla) \ - GST_DEBUG_CATEGORY_INIT (audioresample_debug, "legacyresample", 0, "audio resampling element"); - -GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform, - GST_TYPE_BASE_TRANSFORM, DEBUG_INIT); - -static void -gst_audioresample_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_audioresample_src_template)); - gst_element_class_add_pad_template (gstelement_class, - gst_static_pad_template_get (&gst_audioresample_sink_template)); - - gst_element_class_set_details (gstelement_class, &gst_audioresample_details); -} - -static void -gst_audioresample_class_init (GstAudioresampleClass * klass) -{ - GObjectClass *gobject_class; - - gobject_class = (GObjectClass *) klass; - - gobject_class->set_property = gst_audioresample_set_property; - gobject_class->get_property = gst_audioresample_get_property; - - g_object_class_install_property (gobject_class, PROP_FILTERLEN, - g_param_spec_int ("filter-length", "filter length", - "Length of the resample filter", 0, G_MAXINT, DEFAULT_FILTERLEN, - G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); - - GST_BASE_TRANSFORM_CLASS (klass)->start = - GST_DEBUG_FUNCPTR (audioresample_start); - GST_BASE_TRANSFORM_CLASS (klass)->stop = - GST_DEBUG_FUNCPTR (audioresample_stop); - GST_BASE_TRANSFORM_CLASS (klass)->transform_size = - GST_DEBUG_FUNCPTR (audioresample_transform_size); - GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size = - GST_DEBUG_FUNCPTR (audioresample_get_unit_size); - GST_BASE_TRANSFORM_CLASS (klass)->transform_caps = - GST_DEBUG_FUNCPTR (audioresample_transform_caps); - GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps = - GST_DEBUG_FUNCPTR (audioresample_fixate_caps); - GST_BASE_TRANSFORM_CLASS (klass)->set_caps = - GST_DEBUG_FUNCPTR (audioresample_set_caps); - GST_BASE_TRANSFORM_CLASS (klass)->transform = - GST_DEBUG_FUNCPTR (audioresample_transform); - GST_BASE_TRANSFORM_CLASS (klass)->event = - GST_DEBUG_FUNCPTR (audioresample_event); - - GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE; -} - -static void -gst_audioresample_init (GstAudioresample * audioresample, - GstAudioresampleClass * klass) -{ - GstBaseTransform *trans; - - trans = GST_BASE_TRANSFORM (audioresample); - - /* buffer alloc passthrough is too impossible. FIXME, it - * is trivial in the passthrough case. */ - gst_pad_set_bufferalloc_function (trans->sinkpad, NULL); - - audioresample->filter_length = DEFAULT_FILTERLEN; - - audioresample->need_discont = FALSE; - - gst_pad_set_query_function (trans->srcpad, audioresample_query); - gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type); -} - -/* vmethods */ -static gboolean -audioresample_start (GstBaseTransform * base) -{ - GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); - - audioresample->resample = resample_new (); - audioresample->ts_offset = -1; - audioresample->offset = -1; - audioresample->next_ts = -1; - - resample_set_filter_length (audioresample->resample, - audioresample->filter_length); - - return TRUE; -} - -static gboolean -audioresample_stop (GstBaseTransform * base) -{ - GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); - - if (audioresample->resample) { - resample_free (audioresample->resample); - audioresample->resample = NULL; - } - - gst_caps_replace (&audioresample->sinkcaps, NULL); - gst_caps_replace (&audioresample->srccaps, NULL); - - return TRUE; -} - -static gboolean -audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps, - guint * size) -{ - gint width, channels; - GstStructure *structure; - gboolean ret; - - g_assert (size); - - /* 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 * -audioresample_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; -} - -/* Fixate rate to the allowed rate that has the smallest difference */ -static void -audioresample_fixate_caps (GstBaseTransform * base, - GstPadDirection direction, GstCaps * caps, GstCaps * othercaps) -{ - GstStructure *s; - gint rate; - - s = gst_caps_get_structure (caps, 0); - if (!gst_structure_get_int (s, "rate", &rate)) - return; - - s = gst_caps_get_structure (othercaps, 0); - gst_structure_fixate_field_nearest_int (s, "rate", rate); -} - -static gboolean -resample_set_state_from_caps (ResampleState * state, GstCaps * incaps, - GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate) -{ - GstStructure *structure; - gboolean ret; - gint myinrate, myoutrate; - int mychannels; - gint width, depth; - ResampleFormat format; - - GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %" - GST_PTR_FORMAT, incaps, outcaps); - - structure = gst_caps_get_structure (incaps, 0); - - /* get width */ - ret = gst_structure_get_int (structure, "width", &width); - if (!ret) - goto no_width; - - /* figure out the format */ - if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) { - if (width == 32) - format = RESAMPLE_FORMAT_F32; - else if (width == 64) - format = RESAMPLE_FORMAT_F64; - else - goto wrong_depth; - } else { - /* for int, depth and width must be the same */ - ret = gst_structure_get_int (structure, "depth", &depth); - if (!ret || width != depth) - goto not_equal; - - if (width == 16) - format = RESAMPLE_FORMAT_S16; - else if (width == 32) - format = RESAMPLE_FORMAT_S32; - else - goto wrong_depth; - } - 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; - - resample_set_format (state, format); - resample_set_n_channels (state, mychannels); - resample_set_input_rate (state, myinrate); - resample_set_output_rate (state, myoutrate); - - return TRUE; - - /* ERRORS */ -no_width: - { - GST_DEBUG ("failed to get width from caps"); - return FALSE; - } -not_equal: - { - GST_DEBUG ("width %d and depth %d must be the same", width, depth); - return FALSE; - } -wrong_depth: - { - GST_DEBUG ("unknown depth %d found", depth); - return FALSE; - } -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 -audioresample_transform_size (GstBaseTransform * base, - GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps, - guint * othersize) -{ - GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); - ResampleState *state; - GstCaps *srccaps, *sinkcaps; - gboolean use_internal = FALSE; /* whether we use the internal state */ - gboolean ret = TRUE; - - GST_LOG_OBJECT (base, "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 (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) && - gst_caps_is_equal (srccaps, audioresample->srccaps)) { - use_internal = TRUE; - state = audioresample->resample; - } else { - GST_DEBUG_OBJECT (audioresample, - "caps are not the set caps, creating state"); - state = resample_new (); - resample_set_filter_length (state, audioresample->filter_length); - resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL); - } - - if (direction == GST_PAD_SINK) { - /* asked to convert size of an incoming buffer */ - *othersize = resample_get_output_size_for_input (state, size); - } else { - /* asked to convert size of an outgoing buffer */ - *othersize = resample_get_input_size_for_output (state, size); - } - g_assert (*othersize % state->sample_size == 0); - - /* we make room for one extra sample, given that the resampling filter - * can output an extra one for non-integral i_rate/o_rate */ - GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize); - - if (!use_internal) { - resample_free (state); - } - - return ret; -} - -static gboolean -audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps, - GstCaps * outcaps) -{ - gboolean ret; - gint inrate, outrate; - int channels; - GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); - - GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %" - GST_PTR_FORMAT, incaps, outcaps); - - ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps, - &channels, &inrate, &outrate); - - g_return_val_if_fail (ret, FALSE); - - audioresample->channels = channels; - GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels); - audioresample->i_rate = inrate; - GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate); - audioresample->o_rate = outrate; - GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate); - - /* save caps so we can short-circuit in the size_transform if the caps - * are the same */ - gst_caps_replace (&audioresample->sinkcaps, incaps); - gst_caps_replace (&audioresample->srccaps, outcaps); - - return TRUE; -} - -static gboolean -audioresample_event (GstBaseTransform * base, GstEvent * event) -{ - GstAudioresample *audioresample; - - audioresample = GST_AUDIORESAMPLE (base); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_FLUSH_START: - break; - case GST_EVENT_FLUSH_STOP: - if (audioresample->resample) - resample_input_flush (audioresample->resample); - audioresample->ts_offset = -1; - audioresample->next_ts = -1; - audioresample->offset = -1; - break; - case GST_EVENT_NEWSEGMENT: - resample_input_pushthrough (audioresample->resample); - audioresample_pushthrough (audioresample); - audioresample->ts_offset = -1; - audioresample->next_ts = -1; - audioresample->offset = -1; - break; - case GST_EVENT_EOS: - resample_input_eos (audioresample->resample); - audioresample_pushthrough (audioresample); - break; - default: - break; - } - return parent_class->event (base, event); -} - -static GstFlowReturn -audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf) -{ - int outsize; - int outsamples; - ResampleState *r; - - r = audioresample->resample; - - outsize = resample_get_output_size (r); - GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize); - - /* protect against mem corruption */ - if (outsize > GST_BUFFER_SIZE (outbuf)) { - GST_WARNING_OBJECT (audioresample, - "overriding audioresample's outsize %d with outbuffer's size %d", - outsize, GST_BUFFER_SIZE (outbuf)); - outsize = GST_BUFFER_SIZE (outbuf); - } - /* catch possibly wrong size differences */ - if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) { - GST_WARNING_OBJECT (audioresample, - "audioresample's outsize %d too far from outbuffer's size %d", - outsize, GST_BUFFER_SIZE (outbuf)); - } - - outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize); - outsamples = outsize / r->sample_size; - GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples", - outsize, outsamples); - - GST_BUFFER_OFFSET (outbuf) = audioresample->offset; - GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts; - - if (audioresample->ts_offset != -1) { - audioresample->offset += outsamples; - audioresample->ts_offset += outsamples; - audioresample->next_ts = - gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND, - audioresample->o_rate); - GST_BUFFER_OFFSET_END (outbuf) = audioresample->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) = audioresample->next_ts - - GST_BUFFER_TIMESTAMP (outbuf); - } else { - /* no valid offset know, we can still sortof calculate the duration though */ - GST_BUFFER_DURATION (outbuf) = - gst_util_uint64_scale_int (outsamples, GST_SECOND, - audioresample->o_rate); - } - - /* check for possible mem corruption */ - if (outsize > GST_BUFFER_SIZE (outbuf)) { - /* this is an error that when it happens, would need fixing in the - * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf), - * and it gave us more ! */ - GST_WARNING_OBJECT (audioresample, - "audioresample, you memory corrupting bastard. " - "you gave me outsize %d while my buffer was size %d", - outsize, GST_BUFFER_SIZE (outbuf)); - return GST_FLOW_ERROR; - } - /* catch possibly wrong size differences */ - if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) { - GST_WARNING_OBJECT (audioresample, - "audioresample's written outsize %d too far from outbuffer's size %d", - outsize, GST_BUFFER_SIZE (outbuf)); - } - GST_BUFFER_SIZE (outbuf) = outsize; - - if (G_UNLIKELY (audioresample->need_discont)) { - GST_DEBUG_OBJECT (audioresample, - "marking this buffer with the DISCONT flag"); - GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); - audioresample->need_discont = FALSE; - } - - GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %" - GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" - G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT, - outsize, 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 gboolean -audioresample_check_discont (GstAudioresample * audioresample, - GstClockTime timestamp) -{ - if (timestamp != GST_CLOCK_TIME_NONE && - audioresample->prev_ts != GST_CLOCK_TIME_NONE && - audioresample->prev_duration != GST_CLOCK_TIME_NONE && - timestamp != audioresample->prev_ts + audioresample->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 - - (audioresample->prev_ts + audioresample->prev_duration); - - if (ABS (diff) > GST_SECOND / audioresample->i_rate) { - GST_WARNING_OBJECT (audioresample, - "encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff); - return TRUE; - } - } - - return FALSE; -} - -static GstFlowReturn -audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf, - GstBuffer * outbuf) -{ - GstAudioresample *audioresample; - ResampleState *r; - guchar *data, *datacopy; - gulong size; - GstClockTime timestamp; - - audioresample = GST_AUDIORESAMPLE (base); - r = audioresample->resample; - - data = GST_BUFFER_DATA (inbuf); - size = GST_BUFFER_SIZE (inbuf); - timestamp = GST_BUFFER_TIMESTAMP (inbuf); - - GST_LOG_OBJECT (audioresample, "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 (audioresample_check_discont (audioresample, timestamp))) { - /* Flush internal samples */ - audioresample_pushthrough (audioresample); - /* Inform downstream element about discontinuity */ - audioresample->need_discont = TRUE; - /* We want to recalculate the offset */ - audioresample->ts_offset = -1; - } - - if (audioresample->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. */ - audioresample->next_ts = timestamp; - audioresample->ts_offset = - gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND); - /* 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; - audioresample->offset = - gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND); - } - } - audioresample->prev_ts = timestamp; - audioresample->prev_duration = GST_BUFFER_DURATION (inbuf); - - /* need to memdup, resample takes ownership. */ - datacopy = g_memdup (data, size); - resample_add_input_data (r, datacopy, size, g_free, datacopy); - - return audioresample_do_output (audioresample, outbuf); -} - -/* push remaining data in the buffers out */ -static GstFlowReturn -audioresample_pushthrough (GstAudioresample * audioresample) -{ - int outsize; - ResampleState *r; - GstBuffer *outbuf; - GstFlowReturn res = GST_FLOW_OK; - GstBaseTransform *trans; - - r = audioresample->resample; - - outsize = resample_get_output_size (r); - if (outsize == 0) { - GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush"); - goto done; - } - - trans = GST_BASE_TRANSFORM (audioresample); - - res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize, - GST_PAD_CAPS (trans->srcpad), &outbuf); - if (G_UNLIKELY (res != GST_FLOW_OK)) { - GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes", - outsize); - goto done; - } - - res = audioresample_do_output (audioresample, outbuf); - if (G_UNLIKELY (res != GST_FLOW_OK)) - goto done; - - res = gst_pad_push (trans->srcpad, outbuf); - -done: - return res; -} - -static gboolean -audioresample_query (GstPad * pad, GstQuery * query) -{ - GstAudioresample *audioresample = - GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); - GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample); - gboolean res = TRUE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_LATENCY: - { - GstClockTime min, max; - gboolean live; - guint64 latency; - GstPad *peer; - gint rate = audioresample->i_rate; - gint resampler_latency = audioresample->filter_length / 2; - - 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 (audioresample); - return res; -} - -static const GstQueryType * -audioresample_query_type (GstPad * pad) -{ - static const GstQueryType types[] = { - GST_QUERY_LATENCY, - 0 - }; - - return types; -} - -static void -gst_audioresample_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstAudioresample *audioresample; - - audioresample = GST_AUDIORESAMPLE (object); - - switch (prop_id) { - case PROP_FILTERLEN: - audioresample->filter_length = g_value_get_int (value); - GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d", - audioresample->filter_length); - if (audioresample->resample) { - resample_set_filter_length (audioresample->resample, - audioresample->filter_length); - gst_element_post_message (GST_ELEMENT (audioresample), - gst_message_new_latency (GST_OBJECT (audioresample))); - } - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_audioresample_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioresample *audioresample; - - audioresample = GST_AUDIORESAMPLE (object); - - switch (prop_id) { - case PROP_FILTERLEN: - g_value_set_int (value, audioresample->filter_length); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - - -static gboolean -plugin_init (GstPlugin * plugin) -{ - resample_init (); - - if (!gst_element_register (plugin, "legacyresample", GST_RANK_MARGINAL, - GST_TYPE_AUDIORESAMPLE)) { - return FALSE; - } - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "legacyresample", - "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, - GST_PACKAGE_ORIGIN); |