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audiotestsrc: use base class fill method

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This commit is contained in:
Wim Taymans 2011-08-25 13:21:14 +02:00
parent 6854f2bbf1
commit 81457756f0
1 changed files with 16 additions and 21 deletions
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37
gst/audiotestsrc/gstaudiotestsrc.c
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@ -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;
}