diff options
Diffstat (limited to 'gst/audiotestsrc/gstaudiotestsrc.c')
| -rw-r--r-- | gst/audiotestsrc/gstaudiotestsrc.c | 42 |
1 files changed, 26 insertions, 16 deletions
diff --git a/gst/audiotestsrc/gstaudiotestsrc.c b/gst/audiotestsrc/gstaudiotestsrc.c index 77463fb2..fc3f6448 100644 --- a/gst/audiotestsrc/gstaudiotestsrc.c +++ b/gst/audiotestsrc/gstaudiotestsrc.c @@ -64,12 +64,11 @@ #define M_PI_M2 ( M_PI + M_PI ) -GstElementDetails gst_audio_test_src_details = { - "Audio test source", - "Source/Audio", - "Creates audio test signals of given frequency and volume", - "Stefan Kost <ensonic@users.sf.net>" -}; +static GstElementDetails gst_audio_test_src_details = +GST_ELEMENT_DETAILS ("Audio test source", + "Source/Audio", + "Creates audio test signals of given frequency and volume", + "Stefan Kost <ensonic@users.sf.net>"); enum @@ -116,7 +115,7 @@ gst_audiostestsrc_wave_get_type (void) {0, NULL, NULL}, }; - if (!audiostestsrc_wave_type) { + if (G_UNLIKELY (audiostestsrc_wave_type == 0)) { audiostestsrc_wave_type = g_enum_register_static ("GstAudioTestSrcWave", audiostestsrc_waves); } @@ -140,8 +139,10 @@ static gboolean gst_audio_test_src_query (GstBaseSrc * basesrc, static void gst_audio_test_src_change_wave (GstAudioTestSrc * src); +/* static void gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); +*/ static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, guint length, GstBuffer ** buffer); @@ -195,8 +196,10 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass) GST_DEBUG_FUNCPTR (gst_audio_test_src_is_seekable); gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_test_src_do_seek); gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_test_src_query); - gstbasesrc_class->get_times = - GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times); + /* + gstbasesrc_class->get_times = + GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times); + */ gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_test_src_create); } @@ -576,6 +579,7 @@ gst_audio_test_src_change_volume (GstAudioTestSrc * src) } } +#ifdef __DISABLE_NO_LIVE__ static void gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) @@ -598,6 +602,7 @@ gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer, *end = -1; } } +#endif static gboolean gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment) @@ -636,6 +641,7 @@ static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, guint length, GstBuffer ** buffer) { + GstFlowReturn res; GstAudioTestSrc *src; GstBuffer *buf; GstClockTime next_time; @@ -646,6 +652,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, if (src->eos_reached) return GST_FLOW_UNEXPECTED; + /* example for tagging generated data */ if (!src->tags_pushed) { GstTagList *taglist; GstEvent *event; @@ -660,6 +667,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, src->tags_pushed = TRUE; } + /* check for eos */ if (src->check_seek_stop && (src->n_samples_stop > src->n_samples) && (src->n_samples_stop < src->n_samples + src->samples_per_buffer) @@ -673,15 +681,17 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, src->generate_samples_per_buffer = src->samples_per_buffer; n_samples = src->n_samples + src->samples_per_buffer; } - next_time = n_samples * GST_SECOND / src->samplerate; - - buf = - gst_buffer_new_and_alloc (src->generate_samples_per_buffer * - sizeof (gint16)); - gst_buffer_set_caps (buf, GST_PAD_CAPS (basesrc->srcpad)); + next_time = gst_util_uint64_scale (n_samples, GST_SECOND, + (guint64) src->samplerate); + + /* allocate a new buffer suitable for this pad */ + if ((res = gst_pad_alloc_buffer (basesrc->srcpad, src->n_samples, + src->generate_samples_per_buffer * sizeof (gint16), + GST_PAD_CAPS (basesrc->srcpad), &buf)) != GST_FLOW_OK) { + return res; + } GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->running_time; - GST_BUFFER_OFFSET (buf) = src->n_samples; GST_BUFFER_OFFSET_END (buf) = n_samples; GST_BUFFER_DURATION (buf) = next_time - src->running_time; |