diff options
author | Wim Taymans <wim.taymans@collabora.co.uk> | 2011-08-25 13:21:14 +0200 |
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committer | Wim Taymans <wim.taymans@collabora.co.uk> | 2011-08-25 13:21:14 +0200 |
commit | 81457756f0492162a6792f4331dd0eaffcfa933a (patch) | |
tree | eb5bdbc0dceff8091ea8ff1ec4bf93acd16919a7 /gst/audiotestsrc | |
parent | 6854f2bbf104b3c2ca5538ad3f12764d910f485b (diff) |
audiotestsrc: use base class fill method
Diffstat (limited to 'gst/audiotestsrc')
-rw-r--r-- | gst/audiotestsrc/gstaudiotestsrc.c | 37 |
1 files changed, 16 insertions, 21 deletions
diff --git a/gst/audiotestsrc/gstaudiotestsrc.c b/gst/audiotestsrc/gstaudiotestsrc.c index 81c5a1edc..983d9e74c 100644 --- a/gst/audiotestsrc/gstaudiotestsrc.c +++ b/gst/audiotestsrc/gstaudiotestsrc.c @@ -149,8 +149,8 @@ static void gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); static gboolean gst_audio_test_src_start (GstBaseSrc * basesrc); static gboolean gst_audio_test_src_stop (GstBaseSrc * basesrc); -static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc, - guint64 offset, guint length, GstBuffer ** buffer); +static GstFlowReturn gst_audio_test_src_fill (GstBaseSrc * basesrc, + guint64 offset, guint length, GstBuffer * buffer); static void gst_audio_test_src_class_init (GstAudioTestSrcClass * klass) @@ -220,7 +220,7 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass) GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times); gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_audio_test_src_start); gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_audio_test_src_stop); - gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_test_src_create); + gstbasesrc_class->fill = GST_DEBUG_FUNCPTR (gst_audio_test_src_fill); } static void @@ -1087,11 +1087,10 @@ gst_audio_test_src_is_seekable (GstBaseSrc * basesrc) } static GstFlowReturn -gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, - guint length, GstBuffer ** buffer) +gst_audio_test_src_fill (GstBaseSrc * basesrc, guint64 offset, + guint length, GstBuffer * buffer) { GstAudioTestSrc *src; - GstBuffer *buf; GstClockTime next_time; gint64 next_sample, next_byte; gint bytes, samples; @@ -1163,8 +1162,6 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, bytes = src->generate_samples_per_buffer * bpf; - buf = gst_buffer_new_and_alloc (bytes); - next_byte = src->next_byte + (src->reverse ? (-bytes) : bytes); next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate); @@ -1172,17 +1169,17 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, GST_LOG_OBJECT (src, "next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample, GST_TIME_ARGS (next_time)); - GST_BUFFER_OFFSET (buf) = src->next_sample; - GST_BUFFER_OFFSET_END (buf) = next_sample; + GST_BUFFER_OFFSET (buffer) = src->next_sample; + GST_BUFFER_OFFSET_END (buffer) = next_sample; if (!src->reverse) { - GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->next_time; - GST_BUFFER_DURATION (buf) = next_time - src->next_time; + GST_BUFFER_TIMESTAMP (buffer) = src->timestamp_offset + src->next_time; + GST_BUFFER_DURATION (buffer) = next_time - src->next_time; } else { - GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + next_time; - GST_BUFFER_DURATION (buf) = src->next_time - next_time; + GST_BUFFER_TIMESTAMP (buffer) = src->timestamp_offset + next_time; + GST_BUFFER_DURATION (buffer) = src->next_time - next_time; } - gst_object_sync_values (G_OBJECT (src), GST_BUFFER_TIMESTAMP (buf)); + gst_object_sync_values (G_OBJECT (src), GST_BUFFER_TIMESTAMP (buffer)); src->next_time = next_time; src->next_sample = next_sample; @@ -1190,19 +1187,17 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset, GST_LOG_OBJECT (src, "generating %u samples at ts %" GST_TIME_FORMAT, src->generate_samples_per_buffer, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); - data = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE); + data = gst_buffer_map (buffer, NULL, NULL, GST_MAP_WRITE); src->process (src, data); - gst_buffer_unmap (buf, data, bytes); + gst_buffer_unmap (buffer, data, bytes); if (G_UNLIKELY ((src->wave == GST_AUDIO_TEST_SRC_WAVE_SILENCE) || (src->volume == 0.0))) { - GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_GAP); + GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP); } - *buffer = buf; - return GST_FLOW_OK; } |