From d9e22cf81808e1bc9f2c7d3359fc2436ec0cf136 Mon Sep 17 00:00:00 2001 From: Zaheer Abbas Merali Date: Tue, 17 Aug 2004 21:27:30 +0000 Subject: [PATCH] gst/audioscale/gstaudioscale.*: made audioscale resample from any sample rate to any sample rate Original commit message from CVS: 2004-08-17 Zaheer Abbas Merali * gst/audioscale/gstaudioscale.c: * gst/audioscale/gstaudioscale.h: made audioscale resample from any sample rate to any sample rate --- ChangeLog | 6 + gst/audioscale/gstaudioscale.c | 240 ++++++++++++++++++++++++++++----- gst/audioscale/gstaudioscale.h | 12 +- 3 files changed, 224 insertions(+), 34 deletions(-) diff --git a/ChangeLog b/ChangeLog index a8ec16b451..6ba6bc5dbd 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,9 @@ +2004-08-17 Zaheer Abbas Merali + + * gst/audioscale/gstaudioscale.c: + * gst/audioscale/gstaudioscale.h: + made audioscale resample from any sample rate to any sample rate + 2004-08-17 Thomas Vander Stichele * ext/libpng/gstpngdec.c: diff --git a/gst/audioscale/gstaudioscale.c b/gst/audioscale/gstaudioscale.c index 199a609a69..5a635688d0 100644 --- a/gst/audioscale/gstaudioscale.c +++ b/gst/audioscale/gstaudioscale.c @@ -30,6 +30,9 @@ #include #include +GST_DEBUG_CATEGORY_STATIC (audioscale_debug); +#define GST_CAT_DEFAULT audioscale_debug + /* elementfactory information */ static GstElementDetails gst_audioscale_details = GST_ELEMENT_DETAILS ("Audio scaler", @@ -112,6 +115,8 @@ static void gst_audioscale_set_property (GObject * object, guint prop_id, static void gst_audioscale_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); +static void *gst_audioscale_get_buffer (void *priv, unsigned int size); + static GstElementClass *parent_class = NULL; /*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */ @@ -173,6 +178,9 @@ static void gst_audioscale_class_init (AudioscaleClass * klass) G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + GST_DEBUG_CATEGORY_INIT (audioscale_debug, "audioscale", 0, + "audioscale element"); } static void gst_audioscale_expand_value (GValue * dest, const GValue * src) @@ -191,14 +199,8 @@ static void gst_audioscale_expand_value (GValue * dest, const GValue * src) rate_max = gst_value_get_int_range_max (src); } - rate_min = (rate_min + 1) / 2; - if (rate_min < 1) - rate_min = 1; - if (rate_max < G_MAXINT / 2) { - rate_max *= 2; - } else { - rate_max = G_MAXINT; - } + rate_min = 1; + rate_max = G_MAXINT; g_value_init (dest, GST_TYPE_INT_RANGE); gst_value_set_int_range (dest, rate_min, rate_max); @@ -262,6 +264,7 @@ static void gst_audioscale_expand_caps (GstCaps * caps) /* we do this hack, because the audioscale lib doesn't handle * rate conversions larger than a factor of 2 */ + /* UPDATE: allowed for n iterations so can handle any factor */ for (i = 0; i < gst_caps_get_size (caps); i++) { GstStructure *structure = gst_caps_get_structure (caps, i); const GValue *value; @@ -308,7 +311,7 @@ static GstCaps *gst_audioscale_fixate (GstPad * pad, const GstCaps * caps) GstStructure *structure; audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); - r = audioscale->gst_resample; + r = &(audioscale->gst_resample_template); if (pad == audioscale->srcpad) { otherpad = audioscale->sinkpad; @@ -337,6 +340,8 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps) gst_resample_t *r; GstStructure *structure; double *rate, *otherrate; + double temprate; + int temp; gboolean ret; GstPadLinkReturn link_ret; @@ -344,7 +349,7 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps) GstCaps *copy; audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); - r = audioscale->gst_resample; + r = &(audioscale->gst_resample_template); if (pad == audioscale->srcpad) { @@ -362,12 +367,11 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps) ret = gst_structure_get_int (structure, "rate", &temp); ret &= gst_structure_get_int (structure, "channels", &r->channels); g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED); - *rate = temp; + *rate = (double) temp; copy = gst_caps_copy (caps); gst_audioscale_expand_caps (copy); link_ret = gst_pad_try_set_caps_nonfixed (otherpad, copy); - if (GST_PAD_LINK_FAILED (link_ret)) return link_ret; @@ -376,7 +380,7 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps) structure = gst_caps_get_structure (caps, 0); ret = gst_structure_get_int (structure, "rate", &temp); g_return_val_if_fail (ret, GST_PAD_LINK_REFUSED); - *otherrate = temp; + *otherrate = (double) temp; if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) { r->format = GST_RESAMPLE_FLOAT; } else { @@ -384,7 +388,67 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps) } audioscale->passthru = (r->i_rate == r->o_rate); - gst_resample_reinit (r); + audioscale->increase = (r->o_rate >= r->i_rate); + /* now create audioscale iterations */ + audioscale->num_iterations = 0; + + temprate = r->i_rate; + while (TRUE) { + if (r->o_rate > r->i_rate) { + if (temprate >= r->o_rate) + break; + temprate *= 2; + } else { + if (temprate <= r->o_rate) + break; + temprate /= 2; + } + audioscale->num_iterations++; + } + + + + + if (audioscale->num_iterations > 0) { + audioscale->offsets = g_new0 (gint64, audioscale->num_iterations); + audioscale->gst_resample = g_new0 (gst_resample_t, 1); + audioscale->gst_resample->priv = audioscale; + audioscale->gst_resample->get_buffer = gst_audioscale_get_buffer; + audioscale->gst_resample->method = r->method; + audioscale->gst_resample->channels = r->channels; + audioscale->gst_resample->filter_length = r->filter_length; + audioscale->gst_resample->format = r->format; + if (audioscale->increase) { + temprate = r->o_rate; + + while (temprate / 2 >= r->i_rate) { + temprate = temprate / 2; + } + /* now temprate is output rate of gstresample */ + GST_DEBUG ("gstresample will increase rate from %f to %f", r->i_rate, + temprate); + audioscale->gst_resample->o_rate = temprate; + audioscale->gst_resample->i_rate = r->i_rate; + } else { + temprate = r->i_rate; + + while (temprate / 2 >= r->o_rate) { + temprate = temprate / 2; + } + /* now temprate is input rate of gstresample */ + GST_DEBUG ("gstresample will decrease rate from %f to %f", temprate, + r->o_rate); + audioscale->gst_resample->o_rate = r->o_rate; + audioscale->gst_resample->i_rate = temprate; + } + audioscale->passthru = + (audioscale->gst_resample->i_rate == audioscale->gst_resample->o_rate); + if (!audioscale->passthru) + audioscale->num_iterations--; + GST_DEBUG ("Number of iterations: %d", audioscale->num_iterations); + + gst_resample_init (audioscale->gst_resample); + } return link_ret; } @@ -393,21 +457,101 @@ static void *gst_audioscale_get_buffer (void *priv, unsigned int size) { Audioscale *audioscale = priv; + GST_DEBUG ("size requested: %u irate: %f orate: %f", size, + audioscale->gst_resample->i_rate, audioscale->gst_resample->o_rate); audioscale->outbuf = gst_buffer_new (); GST_BUFFER_SIZE (audioscale->outbuf) = size; GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size); GST_BUFFER_TIMESTAMP (audioscale->outbuf) = - audioscale->offset * GST_SECOND / audioscale->gst_resample->o_rate; - audioscale->offset += + audioscale->gst_resample_offset * GST_SECOND / + audioscale->gst_resample->o_rate; + audioscale->gst_resample_offset += size / sizeof (gint16) / audioscale->gst_resample->channels; return GST_BUFFER_DATA (audioscale->outbuf); } +/* reduces rate by factor of 2 */ +GstBuffer *gst_audioscale_decrease_rate (Audioscale * audioscale, + GstBuffer * buf, double outrate, int cur_iteration) +{ + gint i, j, curoffset; + GstBuffer *outbuf = gst_buffer_new (); + gint16 *outdata; + gint16 *indata; + + GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (buf) / 2; + outdata = g_malloc (GST_BUFFER_SIZE (outbuf)); + indata = (gint16 *) GST_BUFFER_DATA (buf); + + GST_DEBUG + ("iteration = %d channels = %d in size = %d out size = %d outrate = %f", + cur_iteration, audioscale->gst_resample_template.channels, + GST_BUFFER_SIZE (buf), GST_BUFFER_SIZE (outbuf), outrate); + curoffset = 0; + for (i = 0; i < GST_BUFFER_SIZE (buf) / (sizeof (gint16)); + i += 2 * audioscale->gst_resample_template.channels) { + for (j = 0; j < audioscale->gst_resample_template.channels; j++) { + outdata[curoffset + j] = + (indata[i + j] + indata[i + j + + audioscale->gst_resample_template.channels]) / 2; + } + curoffset += audioscale->gst_resample_template.channels; + } + + GST_BUFFER_DATA (outbuf) = (gpointer) outdata; + GST_BUFFER_TIMESTAMP (outbuf) = + audioscale->offsets[cur_iteration] * GST_SECOND / outrate; + audioscale->offsets[cur_iteration] += + GST_BUFFER_SIZE (outbuf) / sizeof (gint16) / + audioscale->gst_resample->channels; + return outbuf; +} + +/* increases rate by factor of 2 */ +GstBuffer *gst_audioscale_increase_rate (Audioscale * audioscale, + GstBuffer * buf, double outrate, int cur_iteration) +{ + gint i, j, curoffset; + GstBuffer *outbuf = gst_buffer_new (); + gint16 *outdata; + gint16 *indata; + + GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (buf) * 2; + outdata = g_malloc (GST_BUFFER_SIZE (outbuf)); + indata = (gint16 *) GST_BUFFER_DATA (buf); + + GST_DEBUG + ("iteration = %d channels = %d in size = %d out size = %d out rate = %f", + cur_iteration, audioscale->gst_resample_template.channels, + GST_BUFFER_SIZE (buf), GST_BUFFER_SIZE (outbuf), outrate); + curoffset = 0; + for (i = 0; i < GST_BUFFER_SIZE (buf) / (sizeof (gint16)); + i += audioscale->gst_resample_template.channels) { + for (j = 0; j < audioscale->gst_resample_template.channels; j++) { + outdata[curoffset] = indata[i + j]; + outdata[curoffset + audioscale->gst_resample_template.channels] = + indata[i + j]; + curoffset++; + } + curoffset += audioscale->gst_resample_template.channels; + } + + GST_BUFFER_DATA (outbuf) = (gpointer) outdata; + GST_BUFFER_TIMESTAMP (outbuf) = + audioscale->offsets[cur_iteration] * GST_SECOND / outrate; + audioscale->offsets[cur_iteration] += + GST_BUFFER_SIZE (outbuf) / sizeof (gint16) / + audioscale->gst_resample->channels; + return outbuf; +} + static void gst_audioscale_init (Audioscale * audioscale) { gst_resample_t *r; + audioscale->num_iterations = 1; + audioscale->sinkpad = gst_pad_new_from_template (gst_static_pad_template_get (&gst_audioscale_sink_template), "sink"); @@ -426,8 +570,7 @@ static void gst_audioscale_init (Audioscale * audioscale) gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps); gst_pad_set_fixate_function (audioscale->srcpad, gst_audioscale_fixate); - r = g_new0 (gst_resample_t, 1); - audioscale->gst_resample = r; + r = &(audioscale->gst_resample_template); r->priv = audioscale; r->get_buffer = gst_audioscale_get_buffer; @@ -439,35 +582,42 @@ static void gst_audioscale_init (Audioscale * audioscale) r->format = GST_RESAMPLE_S16; /*r->verbose = 1; */ - gst_resample_init (r); - - /* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */ + audioscale->gst_resample = NULL; + audioscale->outbuf = NULL; + audioscale->offsets = NULL; + audioscale->gst_resample_offset = 0; + audioscale->increase = FALSE; } static void gst_audioscale_dispose (GObject * object) { Audioscale *audioscale = GST_AUDIOSCALE (object); - if (audioscale->gst_resample) + if (audioscale->gst_resample) { g_free (audioscale->gst_resample); - audioscale->gst_resample = NULL; - + } + if (audioscale->offsets) + g_free (audioscale->offsets); G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_audioscale_chain (GstPad * pad, GstData * _data) { GstBuffer *buf = GST_BUFFER (_data); + GstBuffer *tempbuf, *tempbuf2; + Audioscale *audioscale; guchar *data; gulong size; + gint i; + double outrate; g_return_if_fail (pad != NULL); g_return_if_fail (GST_IS_PAD (pad)); g_return_if_fail (buf != NULL); audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); - if (audioscale->passthru) { + if (audioscale->passthru && audioscale->num_iterations == 0) { gst_pad_push (audioscale->srcpad, GST_DATA (buf)); return; } @@ -478,11 +628,41 @@ static void gst_audioscale_chain (GstPad * pad, GstData * _data) GST_DEBUG ("gst_audioscale_chain: got buffer of %ld bytes in '%s'\n", size, gst_element_get_name (GST_ELEMENT (audioscale))); - gst_resample_scale (audioscale->gst_resample, data, size); + tempbuf = buf; + outrate = audioscale->gst_resample_template.i_rate; + if (audioscale->increase && !audioscale->passthru) { + GST_DEBUG ("doing gstresample"); + gst_resample_scale (audioscale->gst_resample, data, size); + tempbuf = audioscale->outbuf; + gst_buffer_unref (buf); + outrate = audioscale->gst_resample->o_rate; + } + for (i = 0; i < audioscale->num_iterations; i++) { + tempbuf2 = tempbuf; + GST_DEBUG ("doing %s", + audioscale-> + increase ? "gst_audioscale_increase_rate" : + "gst_audioscale_decrease_rate"); - gst_pad_push (audioscale->srcpad, GST_DATA (audioscale->outbuf)); + if (audioscale->increase) { + outrate *= 2; + tempbuf = gst_audioscale_increase_rate (audioscale, tempbuf, outrate, i); + } else { + outrate /= 2; + tempbuf = gst_audioscale_decrease_rate (audioscale, tempbuf, outrate, i); + } + + gst_buffer_unref (tempbuf2); + data = GST_BUFFER_DATA (tempbuf); + size = GST_BUFFER_SIZE (tempbuf); + } + if (!audioscale->increase && !audioscale->passthru) { + gst_resample_scale (audioscale->gst_resample, data, size); + gst_buffer_unref (tempbuf); + tempbuf = audioscale->outbuf; + } + gst_pad_push (audioscale->srcpad, GST_DATA (tempbuf)); - gst_buffer_unref (buf); } static void @@ -495,7 +675,7 @@ static void /* it's not null if we got it, but it might not be ours */ g_return_if_fail (GST_IS_AUDIOSCALE (object)); src = GST_AUDIOSCALE (object); - r = src->gst_resample; + r = &(src->gst_resample_template); switch (prop_id) { case ARG_FILTERLEN: @@ -520,7 +700,7 @@ static void gst_resample_t *r; src = GST_AUDIOSCALE (object); - r = src->gst_resample; + r = &(src->gst_resample_template); switch (prop_id) { case ARG_FILTERLEN: diff --git a/gst/audioscale/gstaudioscale.h b/gst/audioscale/gstaudioscale.h index 710980cb27..29d6edd493 100644 --- a/gst/audioscale/gstaudioscale.h +++ b/gst/audioscale/gstaudioscale.h @@ -57,11 +57,15 @@ struct _Audioscale { /* audio state */ gboolean passthru; - gint64 offset; + gint64 gst_resample_offset; - gst_resample_t *gst_resample; - - GstBuffer *outbuf; + gint64* offsets; + gboolean increase; /* is the rate change an increase */ + gint num_iterations; /* number of iterations through gst_audioscale/(increase|decrease)_rate */ + + gst_resample_t gst_resample_template; + gst_resample_t* gst_resample; + GstBuffer* outbuf; }; struct _AudioscaleClass {