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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Revert "mxf: add a return to fix the build when turning assert off"
This reverts commit 68ca62b369.
This commit is contained in:
Stefan Kost
parent
68ca62b369
commit
76daceadca
3 changed files with 50 additions and 16 deletions
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@ -69,7 +69,8 @@ GST_STATIC_PAD_TEMPLATE ("src",
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 6 ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 32, " "depth = (int) 16, " "signed = (boolean) true")
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"width = (int) { 16, 32 }, "
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"depth = (int) 16, " "signed = (boolean) true")
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);
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static GstStaticPadTemplate vorbis_dec_sink_factory =
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@ -237,7 +238,7 @@ vorbis_dec_convert (GstPad * pad,
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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scale = sizeof (gint32) * dec->vi.channels;
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scale = dec->width * dec->vi.channels;
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case GST_FORMAT_DEFAULT:
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*dest_value =
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scale * gst_util_uint64_scale_int (src_value, dec->vi.rate,
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@ -250,7 +251,7 @@ vorbis_dec_convert (GstPad * pad,
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case GST_FORMAT_DEFAULT:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = src_value * sizeof (gint32) * dec->vi.channels;
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*dest_value = src_value * dec->width * dec->vi.channels;
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break;
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case GST_FORMAT_TIME:
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*dest_value =
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@ -263,11 +264,11 @@ vorbis_dec_convert (GstPad * pad,
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_DEFAULT:
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*dest_value = src_value / (sizeof (gint32) * dec->vi.channels);
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*dest_value = src_value / (dec->width * dec->vi.channels);
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break;
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case GST_FORMAT_TIME:
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*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
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dec->vi.rate * sizeof (gint32) * dec->vi.channels);
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dec->vi.rate * dec->width * dec->vi.channels);
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break;
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default:
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res = FALSE;
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@ -574,6 +575,7 @@ vorbis_handle_identification_packet (GstIVorbisDec * vd)
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{
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GstCaps *caps;
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const GstAudioChannelPosition *pos = NULL;
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gint width = 16;
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switch (vd->vi.channels) {
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case 1:
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@ -626,10 +628,24 @@ vorbis_handle_identification_packet (GstIVorbisDec * vd)
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goto channel_count_error;
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}
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/* negotiate with downstream */
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caps = gst_pad_get_allowed_caps (vd->srcpad);
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if (caps) {
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if (!gst_caps_is_empty (caps)) {
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GstStructure *s;
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s = gst_caps_get_structure (caps, 0);
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/* template ensures 16 or 32 */
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gst_structure_get_int (s, "width", &width);
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}
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gst_caps_unref (caps);
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}
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vd->width = width >> 3;
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, vd->vi.rate,
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"channels", G_TYPE_INT, vd->vi.channels,
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"endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32,
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"endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, width,
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"depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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if (pos) {
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@ -830,6 +846,18 @@ copy_samples (gint32 * out, ogg_int32_t ** in, guint samples, gint channels)
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}
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}
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static void
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copy_samples_16 (gint16 * out, ogg_int32_t ** in, guint samples, gint channels)
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{
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gint i, j;
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for (j = 0; j < samples; j++) {
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for (i = 0; i < channels; i++) {
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*out++ = CLIP_TO_15 (in[i][j] >> 9);
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}
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}
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}
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/* clip output samples to the segment boundaries
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*/
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static gboolean
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@ -853,7 +881,7 @@ vorbis_do_clip (GstIVorbisDec * dec, GstBuffer * buf)
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/* bring clipped time to samples */
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diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND);
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/* samples to bytes */
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diff *= (sizeof (gint32) * dec->vi.channels);
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diff *= (dec->width * dec->vi.channels);
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GST_DEBUG_OBJECT (dec, "clipping start to %" GST_TIME_FORMAT " %"
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G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstart), diff);
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GST_BUFFER_DATA (buf) += diff;
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@ -865,7 +893,7 @@ vorbis_do_clip (GstIVorbisDec * dec, GstBuffer * buf)
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/* bring clipped time to samples and then to bytes */
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diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND);
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diff *= (sizeof (gint32) * dec->vi.channels);
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diff *= (dec->width * dec->vi.channels);
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GST_DEBUG_OBJECT (dec, "clipping stop to %" GST_TIME_FORMAT " %"
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G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstop), diff);
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GST_BUFFER_SIZE (buf) -= diff;
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@ -904,8 +932,7 @@ vorbis_dec_push (GstIVorbisDec * dec, GstBuffer * buf)
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walk = g_list_previous (walk)) {
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GstBuffer *buffer = GST_BUFFER (walk->data);
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outoffset -=
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GST_BUFFER_SIZE (buffer) / (sizeof (gint32) * dec->vi.channels);
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outoffset -= GST_BUFFER_SIZE (buffer) / (dec->width * dec->vi.channels);
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GST_BUFFER_OFFSET (buffer) = outoffset;
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GST_BUFFER_TIMESTAMP (buffer) =
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@ -986,7 +1013,7 @@ vorbis_handle_data_packet (GstIVorbisDec * vd, ogg_packet * packet)
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if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0)
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goto done;
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size = sample_count * vd->vi.channels * sizeof (gint32);
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size = sample_count * vd->vi.channels * vd->width;
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/* alloc buffer for it */
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result =
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@ -1000,8 +1027,15 @@ vorbis_handle_data_packet (GstIVorbisDec * vd, ogg_packet * packet)
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goto wrong_samples;
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/* copy samples in buffer */
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if (vd->width == 4) {
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copy_samples ((gint32 *) GST_BUFFER_DATA (out), pcm, sample_count,
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vd->vi.channels);
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} else if (vd->width == 2) {
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copy_samples_16 ((gint16 *) GST_BUFFER_DATA (out), pcm, sample_count,
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vd->vi.channels);
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} else {
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g_assert_not_reached ();
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}
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GST_BUFFER_SIZE (out) = size;
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GST_BUFFER_OFFSET (out) = vd->granulepos;
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@ -1183,6 +1217,7 @@ vorbis_dec_change_state (GstElement * element, GstStateChange transition)
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vorbis_info_init (&vd->vi);
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vorbis_comment_init (&vd->vc);
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vd->initialized = FALSE;
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vd->width = 2;
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gst_ivorbis_dec_reset (vd);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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@ -524,7 +524,6 @@ mxf_up_get_descriptor (GstPadTemplate * tmpl, GstCaps * caps,
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}
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g_assert_not_reached ();
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return NULL;
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}
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static void
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@ -6,7 +6,7 @@
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#undef ENABLE_NLS
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/* gettext package name */
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#define GETTEXT_PACKAGE "NULL"
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#define GETTEXT_PACKAGE "gst-plugins-bad-0.10"
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/* PREFIX - specifically added for Windows for easier moving */
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#define PREFIX "C:\\gstreamer"
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@ -27,7 +27,7 @@
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#define GST_PACKAGE_NAME "GStreamer Bad Plug-ins git"
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/* package origin */
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#define GST_PACKAGE_ORIGIN "http://gstreamer.freedesktop.org"
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#define GST_PACKAGE_ORIGIN "Unknown package origin"
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/* Define if the host CPU is an Alpha */
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#undef HAVE_CPU_ALPHA
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