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gst/audioconvert/gstaudioconvert.c: do conversions from/to float correctly, fix some caps nego errors, export correct...
Original commit message from CVS: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps), (gst_audio_convert_link), (gst_audio_convert_change_state), (gst_audio_convert_buffer_from_default_format): do conversions from/to float correctly, fix some caps nego errors, export correct supported caps in template and getcaps, use correct caps in try_set_caps functions
This commit is contained in:
Benjamin Otte
parent
5b32d38c0b
commit
33f79a881e
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@ -1,3 +1,12 @@
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2004-03-06 Benjamin Otte <otte@gnome.org>
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* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps),
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(gst_audio_convert_link), (gst_audio_convert_change_state),
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(gst_audio_convert_buffer_from_default_format):
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do conversions from/to float correctly, fix some caps nego errors,
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export correct supported caps in template and getcaps, use correct
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caps in try_set_caps functions
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2004-03-06 Christophe Fergeau <teuf@gnome.org>
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For some reason, I only committed a ChangeLog entry yesterday and
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@ -71,10 +71,6 @@ struct _GstAudioConvert {
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/* conversion functions */
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GstBuffer * (* convert_internal) (GstAudioConvert *this, GstBuffer *buf);
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/* for int2float */
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GstBuffer * output;
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gint output_samples_needed;
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};
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struct _GstAudioConvertClass {
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@ -96,16 +92,10 @@ static void gst_audio_convert_init (GstAudioConvert *audio_convert);
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/* gstreamer functions */
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static void gst_audio_convert_chain (GstPad *pad, GstData *_data);
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static void gst_audio_convert_chain_int2float (GstPad *pad, GstData *_data);
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static GstPadLinkReturn gst_audio_convert_link (GstPad *pad, const GstCaps *caps);
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static GstCaps * gst_audio_convert_getcaps (GstPad *pad);
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static GstElementStateReturn gst_audio_convert_change_state (GstElement *element);
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/* actual work */
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#if 0
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static gboolean gst_audio_convert_set_caps (GstPad *pad);
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#endif
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static GstBuffer * gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *buf);
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static GstBuffer * gst_audio_convert_buffer_from_default_format (GstAudioConvert *this, GstBuffer *buf);
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@ -129,26 +119,29 @@ GST_BOILERPLATE_FULL (GstAudioConvert, gst_audio_convert, GstElement, GST_TYPE_E
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/*** GSTREAMER PROTOTYPES *****************************************************/
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#define STATIC_CAPS \
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GST_STATIC_CAPS ( \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 2 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) { 8, 16, 32 }, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 2 ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 32, " \
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"buffer-frames = (int) [ 0, MAX ]" \
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)
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static GstStaticPadTemplate gst_audio_convert_src_template =
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GST_STATIC_PAD_TEMPLATE (
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"src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (
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"audio/x-raw-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
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"width = (int) { 8, 16, 32 }, "
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"depth = (int) [ 1, 32 ], "
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"signed = (boolean) { true, false }; "
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"audio/x-raw-float, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 32, "
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"buffer-frames = (int) [ 0, MAX ]"
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)
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STATIC_CAPS
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);
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static GstStaticPadTemplate gst_audio_convert_sink_template =
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@ -156,22 +149,7 @@ GST_STATIC_PAD_TEMPLATE (
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) { 8, 16, 32 }, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }; "
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"audio/x-raw-float, "
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"rate = (int) [ 1, MAX ],"
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 32, "
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"buffer-frames = (int) [ 0, MAX ]"
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)
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STATIC_CAPS
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);
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/*** TYPE FUNCTIONS ***********************************************************/
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@ -266,118 +244,6 @@ gst_audio_convert_chain (GstPad *pad, GstData *data)
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gst_pad_push (this->src, GST_DATA (buf));
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}
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/* 1 / (2^31-1) * i */
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#define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i))
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/* This custom chain handler exists because if buffer-frames is nonzero, one int
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* buffer probably doesn't correspond to one float buffer */
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static void
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gst_audio_convert_chain_int2float (GstPad *pad, GstData *data)
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{
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GstBuffer *buf = GST_BUFFER (data);
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GstAudioConvert *this;
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gint buffer_samples, samples_remaining, i;
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gint32 *in;
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gfloat *out;
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this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad));
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/* FIXME */
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if (GST_IS_EVENT (buf)) {
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gst_pad_event_default (pad, GST_EVENT (buf));
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return;
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}
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/* we know we're negotiated, because it's the link function that set the
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custom chain handler */
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/**
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* Theory of operation:
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* - convert the format (endianness, signedness, width, depth) to
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* (G_BYTE_ORDER, TRUE, 32, 32)
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* - convert rate and channels
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* - if buffer-frames is zero, convert and push.
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* - if we have an output buffer, fill it. if it becomes full, push it.
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* - while buffer-frames is less than the number of frames remaining in the
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* input, create sub-buffers, convert and push.
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* - if there are leftover frames in the input, create an output buffer and
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* fill it partially.
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*/
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buf = gst_audio_convert_buffer_to_default_format (this, buf);
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buf = gst_audio_convert_channels (this, buf);
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/* we know buf is writable */
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buffer_samples = this->srccaps.buffer_frames * this->srccaps.channels;
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in = (gint32*)GST_BUFFER_DATA (buf);
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out = (gfloat*)GST_BUFFER_DATA (buf);
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samples_remaining = buf->size / sizeof(gint32);
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if (!buffer_samples ||
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(!this->output && samples_remaining == buffer_samples)) {
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for (i=samples_remaining; i; i--)
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*(out++) = INT2FLOAT (*(in++));
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gst_pad_push (this->src, GST_DATA (buf));
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return;
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}
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if (this->output) {
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GstBuffer *output = this->output;
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gint to_process = MIN (this->output_samples_needed, samples_remaining);
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out = ((gfloat*)GST_BUFFER_DATA (output) +
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(buffer_samples - this->output_samples_needed));
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for (i=to_process; i; i--)
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*(out++) = INT2FLOAT (*(in++));
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this->output_samples_needed -= to_process;
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samples_remaining -= to_process;
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/* one of the two of these ifs will be true, and possibly both of them */
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if (!this->output_samples_needed) {
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this->output = NULL;
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gst_pad_push (this->src, GST_DATA (output));
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}
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if (!samples_remaining) {
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gst_buffer_unref (buf);
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return;
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}
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/* we have some leftover frames in buf, let's take care of them */
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out = (gfloat*)in;
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}
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while (samples_remaining > buffer_samples) {
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GstBuffer *sub_buf;
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sub_buf = gst_buffer_create_sub (buf,
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(GST_BUFFER_SIZE (buf) -
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samples_remaining * sizeof(gint32)),
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buffer_samples * sizeof(gfloat));
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/* `out' should be positioned correctly */
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for (i=buffer_samples; i; i--)
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*(out++) = INT2FLOAT (*(in++));
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samples_remaining -= buffer_samples;
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gst_pad_push (this->src, GST_DATA (sub_buf));
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}
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if (samples_remaining) {
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GstBuffer *output;
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output = this->output = gst_buffer_new_and_alloc (buffer_samples * sizeof(gfloat));
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out = (gfloat*)GST_BUFFER_DATA (output);
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for (i=samples_remaining; i; i--)
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*(out++) = INT2FLOAT (*(in++));
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this->output = output;
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this->output_samples_needed = buffer_samples - samples_remaining;
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samples_remaining = 0; /* just so we know */
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}
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gst_buffer_unref (buf);
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return;
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}
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/* this function is complicated now, but it will be unnecessary when we convert
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* rate. */
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static GstCaps *
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@ -388,7 +254,6 @@ gst_audio_convert_getcaps (GstPad *pad)
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GstStructure *structure;
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GstCaps *othercaps, *caps;
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const GstCaps *templcaps;
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gboolean has_float = FALSE, has_int = FALSE;
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int i, size;
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g_return_val_if_fail(GST_IS_PAD(pad), NULL);
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@ -403,40 +268,25 @@ gst_audio_convert_getcaps (GstPad *pad)
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size = gst_caps_get_size (othercaps);
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for (i=0; i<size; i++) {
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for (i = size - 1; i >= 0; i--) {
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structure = gst_caps_get_structure (othercaps, i);
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gst_structure_remove_field (structure, "channels");
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gst_structure_remove_field (structure, "endianness");
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gst_structure_remove_field (structure, "width");
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gst_structure_remove_field (structure, "depth");
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gst_structure_remove_field (structure, "signed");
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structure = gst_structure_copy (structure);
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if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
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if (!has_int) has_int = TRUE;
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gst_structure_remove_field (structure, "depth");
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gst_structure_remove_field (structure, "signed");
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gst_structure_set_name (structure, "audio/x-raw-float");
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gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL);
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} else {
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if (!has_float) has_float = TRUE;
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gst_structure_set_name (structure, "audio/x-raw-int");
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gst_structure_remove_field (structure, "buffer-frames");
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}
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gst_caps_append_structure (othercaps, structure);
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}
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caps = gst_caps_intersect (othercaps, templcaps);
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gst_caps_free (othercaps);
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size = gst_caps_get_size (caps);
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/* the intersection probably lost either float or int. so we take the rate
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* property and set it on a copy of the templcaps struct. */
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if (!has_int && size) {
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structure = gst_structure_copy (gst_caps_get_structure (templcaps, 0));
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gst_structure_set_value (structure, "rate",
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gst_structure_get_value (gst_caps_get_structure (caps, 0),
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"rate"));
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gst_caps_append_structure (caps, structure);
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}
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if (!has_float && size) {
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structure = gst_structure_copy (gst_caps_get_structure (templcaps, 1));
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gst_structure_set_value (structure, "rate",
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gst_structure_get_value (gst_caps_get_structure (caps, 0),
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"rate"));
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gst_caps_append_structure (caps, structure);
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}
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return caps;
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}
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@ -502,13 +352,18 @@ gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
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for (i = 0; i < gst_caps_get_size (othercaps); i++) {
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GstStructure *structure = gst_caps_get_structure (othercaps, i);
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gst_structure_set (structure, "rate", G_TYPE_INT, ac_caps.rate, NULL);
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if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float") == 0) {
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if (!ac_caps.is_int) {
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gst_structure_set (structure, "buffer-frames", G_TYPE_INT, ac_caps.buffer_frames, NULL);
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} else {
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gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL);
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}
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}
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}
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ret = gst_pad_try_set_caps_nonfixed (otherpad, othercaps);
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gst_caps_free (othercaps);
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if (ret < GST_PAD_LINK_OK)
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return ret;
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if (!gst_audio_convert_parse_caps (caps, &other_ac_caps))
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return GST_PAD_LINK_REFUSED;
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/* woohoo, got it */
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if (!gst_audio_convert_parse_caps (gst_pad_get_negotiated_caps (otherpad),
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@ -517,15 +372,6 @@ gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
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return GST_PAD_LINK_REFUSED;
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}
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if (!other_ac_caps.is_int && !ac_caps.is_int) {
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GST_DEBUG ("we don't do float-float conversions yet");
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return GST_PAD_LINK_REFUSED;
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} else if ((this->sink == pad) ? !other_ac_caps.is_int : ac_caps.is_int) {
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GST_DEBUG ("int-float conversion, setting custom chain handler");
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gst_pad_set_chain_function (this->sink, gst_audio_convert_chain_int2float);
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}
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/* float2int conversion is handled like other int formats */
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if (this->sink == pad) {
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this->srccaps = other_ac_caps;
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this->sinkcaps = ac_caps;
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@ -547,8 +393,6 @@ gst_audio_convert_change_state (GstElement *element)
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switch (GST_STATE_TRANSITION (element)) {
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case GST_STATE_PAUSED_TO_READY:
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this->convert_internal = NULL;
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GST_DEBUG_OBJECT (element, "resetting chain function to the default");
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gst_pad_set_chain_function (this->sink, gst_audio_convert_chain);
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break;
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default:
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break;
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@ -705,47 +549,65 @@ static GstBuffer *
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gst_audio_convert_buffer_from_default_format (GstAudioConvert *this, GstBuffer *buf)
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{
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GstBuffer *ret;
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guint8 *dest;
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guint count, i;
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gint32 *src;
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if (this->srccaps.width == 32 && this->srccaps.depth == 32 &&
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if (this->srccaps.is_int && this->srccaps.width == 32 && this->srccaps.depth == 32 &&
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this->srccaps.endianness == G_BYTE_ORDER && this->srccaps.sign == TRUE)
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return buf;
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count = buf->size / 4; /* size is undefined after gst_audio_convert_get_buffer! */
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ret = gst_audio_convert_get_buffer (buf, buf->size * this->srccaps.width / 32);
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if (this->srccaps.is_int) {
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guint8 *dest;
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count = buf->size / 4; /* size is undefined after gst_audio_convert_get_buffer! */
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ret = gst_audio_convert_get_buffer (buf, buf->size * this->srccaps.width / 32);
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dest = ret->data;
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src = (gint32 *) buf->data;
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dest = ret->data;
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src = (gint32 *) buf->data;
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for (i = 0; i < count; i++) {
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gint32 int_value = *src;
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src++;
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switch (this->srccaps.width) {
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case 8:
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if (this->srccaps.sign) {
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POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY);
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} else {
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POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
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for (i = 0; i < count; i++) {
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gint32 int_value = *src;
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src++;
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switch (this->srccaps.width) {
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case 8:
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if (this->srccaps.sign) {
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POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY);
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} else {
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POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
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}
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break;
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case 16:
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if (this->srccaps.sign) {
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POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE);
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} else {
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POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE);
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}
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break;
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case 32:
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if (this->srccaps.sign) {
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POPULATE (gint32, GINT32_TO_BE, GINT32_TO_LE);
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} else {
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POPULATE (guint32, GUINT32_TO_BE, GUINT32_TO_LE);
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}
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break;
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default:
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g_assert_not_reached ();
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}
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}
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break;
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case 16:
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if (this->srccaps.sign) {
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POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE);
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} else {
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POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE);
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}
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||||
break;
|
||||
case 32:
|
||||
if (this->srccaps.sign) {
|
||||
POPULATE (gint32, GINT32_TO_BE, GINT32_TO_LE);
|
||||
} else {
|
||||
POPULATE (guint32, GUINT32_TO_BE, GUINT32_TO_LE);
|
||||
}
|
||||
break;
|
||||
default:
|
||||
g_assert_not_reached ();
|
||||
} else {
|
||||
gfloat *dest;
|
||||
|
||||
/* 1 / (2^31-1) * i */
|
||||
#define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i))
|
||||
count = buf->size / 4; /* size is undefined after gst_audio_convert_get_buffer! */
|
||||
ret = gst_audio_convert_get_buffer (buf, buf->size * this->srccaps.width / 32);
|
||||
|
||||
dest = (gfloat *) ret->data;
|
||||
src = (gint32 *) buf->data;
|
||||
for (i = 0; i < count; i++) {
|
||||
*dest = (4.6566128752457969e-10 * ((gfloat) *src));
|
||||
dest++;
|
||||
src++;
|
||||
}
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue