mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
fix wavparse so that it handles mp1/2/3 unref buffers that aren't pushed out
Original commit message from CVS: fix wavparse so that it handles mp1/2/3 unref buffers that aren't pushed out
This commit is contained in:
Thomas Vander Stichele
parent
52590aae8f
commit
bfedb2363f
2 changed files with 113 additions and 31 deletions
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@ -57,8 +57,8 @@ GST_PAD_TEMPLATE_FACTORY (sink_template_factory,
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"wavparse_wav",
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"audio/x-wav",
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"wavparse_wav",
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"audio/x-wav",
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NULL
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)
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)
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@ -68,14 +68,14 @@ GST_PAD_TEMPLATE_FACTORY (src_template_factory,
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"wavparse_raw",
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"audio/raw",
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"wavparse_raw",
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"audio/raw",
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"format", GST_PROPS_STRING ("int"),
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"law", GST_PROPS_INT (0),
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_LIST (
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GST_PROPS_BOOLEAN (FALSE),
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GST_PROPS_BOOLEAN (TRUE)
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GST_PROPS_BOOLEAN (FALSE),
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GST_PROPS_BOOLEAN (TRUE)
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),
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"width", GST_PROPS_LIST (
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GST_PROPS_INT (8),
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@ -85,14 +85,19 @@ GST_PAD_TEMPLATE_FACTORY (src_template_factory,
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GST_PROPS_INT (8),
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GST_PROPS_INT (16)
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),
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"rate", GST_PROPS_INT_RANGE (8000, 48000),
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"rate", GST_PROPS_INT_RANGE (8000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2)
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),
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GST_CAPS_NEW (
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"wavparse_mp3",
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"audio/x-mp3",
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NULL
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)
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)
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/* typefactory for 'wav' */
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static GstTypeDefinition
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wavdefinition =
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static GstTypeDefinition
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wavdefinition =
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{
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"wavparse_audio/x-wav",
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"audio/x-wav",
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@ -164,7 +169,7 @@ gst_wavparse_init (GstWavParse *wavparse)
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gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
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gst_pad_set_formats_function (wavparse->srcpad, gst_wavparse_get_formats);
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gst_pad_set_convert_function (wavparse->srcpad, gst_wavparse_pad_convert);
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gst_pad_set_query_type_function (wavparse->srcpad,
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gst_pad_set_query_type_function (wavparse->srcpad,
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gst_wavparse_get_query_types);
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gst_pad_set_query_function (wavparse->srcpad, gst_wavparse_pad_query);
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@ -192,6 +197,30 @@ wav_type_find (GstBuffer *buf, gpointer private)
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}
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/* set timestamp on outgoing buffer
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* returns TRUE if a timestamp was set
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*/
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static gboolean
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gst_wavparse_set_timestamp (GstWavParse *wavparse, GstBuffer *buf)
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{
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gboolean retval = FALSE;
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/* only do timestamps on linear audio */
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switch (wavparse->format)
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{
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case GST_RIFF_WAVE_FORMAT_PCM:
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GST_BUFFER_TIMESTAMP (buf) = wavparse->offset * GST_SECOND
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/ wavparse->rate;
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wavparse->offset += GST_BUFFER_SIZE (buf) * 8
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/ (wavparse->width * wavparse->channels);
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retval = TRUE;
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break;
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default:
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break;
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}
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return retval;
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}
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static void
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gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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{
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@ -215,20 +244,22 @@ gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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/* we're in the data region */
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if (wavparse->state == GST_WAVPARSE_DATA) {
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/* if we're expected to see a new chunk in this buffer */
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if ((wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf)) < GST_BUFFER_SIZE (buf)) {
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GST_BUFFER_SIZE (buf) = wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf);
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if ((wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf))
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< GST_BUFFER_SIZE (buf)) {
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GST_BUFFER_SIZE (buf) = wavparse->riff_nextlikely
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- GST_BUFFER_OFFSET (buf);
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wavparse->state = GST_WAVPARSE_OTHER;
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/* I suppose we could signal an EOF at this point, but that may be
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premature. We've stopped data flow, that's the main thing. */
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}
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}
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GST_BUFFER_TIMESTAMP (buf) = wavparse->offset * GST_SECOND / wavparse->rate;
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wavparse->offset += GST_BUFFER_SIZE (buf) * 8 / wavparse->width / wavparse->channels;
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gst_wavparse_set_timestamp (wavparse, buf);
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if (GST_PAD_IS_USABLE (wavparse->srcpad))
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gst_pad_push (wavparse->srcpad, buf);
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else
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gst_buffer_unref (buf);
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return;
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}
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@ -250,7 +281,7 @@ gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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/* create a new RIFF parser */
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wavparse->riff = gst_riff_new ();
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/* give it the current buffer to start parsing */
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retval = gst_riff_next_buffer (wavparse->riff, buf, 0);
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buffer_riffed = TRUE;
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@ -287,8 +318,7 @@ gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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/* if we've got something, deal with it */
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if (fmt != NULL) {
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GstCaps *caps;
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GstCaps *caps = NULL;
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/* we can gather format information now */
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format = (GstWavParseFormat *)((guchar *) GST_BUFFER_DATA (buf) + fmt->offset);
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@ -297,20 +327,37 @@ gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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wavparse->rate = GUINT32_FROM_LE(format->dwSamplesPerSec);
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wavparse->channels = GUINT16_FROM_LE(format->wChannels);
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wavparse->width = GUINT16_FROM_LE(format->wBitsPerSample);
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wavparse->format = GINT16_FROM_LE(format->wFormatTag);
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/* set the caps on the src pad */
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caps = GST_CAPS_NEW (
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/* FIXME: handle all of the other formats as well */
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switch (wavparse->format)
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{
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case GST_RIFF_WAVE_FORMAT_PCM:
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caps = GST_CAPS_NEW (
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"parsewav_src",
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"audio/raw",
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"format", GST_PROPS_STRING ("int"),
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"law", GST_PROPS_INT (0), /*FIXME */
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_BOOLEAN ((wavparse->width > 8) ? TRUE : FALSE),
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"signed", GST_PROPS_BOOLEAN ((wavparse->width > 8) ? TRUE : FALSE),
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"width", GST_PROPS_INT (wavparse->width),
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"depth", GST_PROPS_INT (wavparse->width),
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"rate", GST_PROPS_INT (wavparse->rate),
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"channels", GST_PROPS_INT (wavparse->channels)
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);
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);
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break;
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case GST_RIFF_WAVE_FORMAT_MPEGL12:
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case GST_RIFF_WAVE_FORMAT_MPEGL3:
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caps = GST_CAPS_NEW (
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"parsewav_src",
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"audio/x-mp3",
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NULL
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);
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break;
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default:
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g_warning ("wavparse: format %d not handled", wavparse->format);
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}
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if (gst_pad_try_set_caps (wavparse->srcpad, caps) <= 0) {
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gst_element_error (GST_ELEMENT (wavparse), "Could not set caps");
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@ -318,7 +365,7 @@ gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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}
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GST_DEBUG (0, "frequency %d, channels %d",
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wavparse->rate, wavparse->channels);
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wavparse->rate, wavparse->channels);
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/* we're now looking for the data chunk */
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wavparse->state = GST_WAVPARSE_CHUNK_DATA;
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@ -355,19 +402,22 @@ gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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/* now we construct a new buffer for the remainder */
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subsize = size - datachunk->offset;
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GST_DEBUG (0, "sending last %ld bytes along as audio", subsize);
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newbuf = gst_buffer_new ();
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GST_BUFFER_DATA (newbuf) = g_malloc (subsize);
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GST_BUFFER_SIZE (newbuf) = subsize;
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GST_BUFFER_TIMESTAMP (newbuf) = wavparse->offset * GST_SECOND / wavparse->rate;
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wavparse->offset += subsize * 8 / wavparse->width / wavparse->channels;
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memcpy (GST_BUFFER_DATA (newbuf), GST_BUFFER_DATA (buf) + datachunk->offset, subsize);
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gst_wavparse_set_timestamp (wavparse, newbuf);
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memcpy (GST_BUFFER_DATA (newbuf),
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GST_BUFFER_DATA (buf) + datachunk->offset, subsize);
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gst_buffer_unref (buf);
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if (GST_PAD_IS_USABLE (wavparse->srcpad))
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gst_pad_push (wavparse->srcpad, newbuf);
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else
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gst_buffer_unref (newbuf);
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/* now we're ready to go, the next buffer should start data */
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wavparse->state = GST_WAVPARSE_DATA;
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@ -69,6 +69,9 @@ struct _GstWavParse {
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/* expected length of audio */
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gulong size;
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/* format of audio, see defines below */
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gint format;
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/* useful audio data */
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gint bps;
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gint rate;
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@ -95,9 +98,38 @@ struct _GstWavParseFormat {
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guint16 wBitsPerSample;
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};
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/**** from public Microsoft RIFF docs ******/
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#define GST_RIFF_WAVE_FORMAT_UNKNOWN (0x0000)
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#define GST_RIFF_WAVE_FORMAT_PCM (0x0001)
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#define GST_RIFF_WAVE_FORMAT_ADPCM (0x0002)
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#define GST_RIFF_WAVE_FORMAT_IBM_CVSD (0x0005)
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#define GST_RIFF_WAVE_FORMAT_ALAW (0x0006)
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#define GST_RIFF_WAVE_FORMAT_MULAW (0x0007)
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#define GST_RIFF_WAVE_FORMAT_OKI_ADPCM (0x0010)
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#define GST_RIFF_WAVE_FORMAT_DVI_ADPCM (0x0011)
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#define GST_RIFF_WAVE_FORMAT_DIGISTD (0x0015)
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#define GST_RIFF_WAVE_FORMAT_DIGIFIX (0x0016)
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#define GST_RIFF_WAVE_FORMAT_YAMAHA_ADPCM (0x0020)
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#define GST_RIFF_WAVE_FORMAT_DSP_TRUESPEECH (0x0022)
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#define GST_RIFF_WAVE_FORMAT_GSM610 (0x0031)
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#define GST_RIFF_WAVE_FORMAT_MSN (0x0032)
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#define GST_RIFF_WAVE_FORMAT_MPEGL12 (0x0050)
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#define GST_RIFF_WAVE_FORMAT_MPEGL3 (0x0055)
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#define GST_RIFF_IBM_FORMAT_MULAW (0x0101)
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#define GST_RIFF_IBM_FORMAT_ALAW (0x0102)
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#define GST_RIFF_IBM_FORMAT_ADPCM (0x0103)
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#define GST_RIFF_WAVE_FORMAT_DIVX (0x0160)
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#define GST_RIFF_WAVE_FORMAT_divx (0x0161)
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#define GST_RIFF_WAVE_FORMAT_VORBIS1 (0x674f)
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#define GST_RIFF_WAVE_FORMAT_VORBIS2 (0x6750)
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#define GST_RIFF_WAVE_FORMAT_VORBIS3 (0x6751)
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#define GST_RIFF_WAVE_FORMAT_VORBIS1PLUS (0x676f)
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#define GST_RIFF_WAVE_FORMAT_VORBIS2PLUS (0x6770)
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#define GST_RIFF_WAVE_FORMAT_VORBIS3PLUS (0x6771)
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#ifdef __cplusplus
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}
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#endif /* __cplusplus */
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#endif /* __GST_PARSEAU_H__ */
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#endif /* __GST_WAVPARSE_H__ */
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