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audiobasesrc: Break some too long lines
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6b17d86692
commit
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1 changed files with 18 additions and 13 deletions
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@ -176,9 +176,9 @@ gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
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g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
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g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
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g_param_spec_int64 ("latency-time", "Latency Time",
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g_param_spec_int64 ("latency-time", "Latency Time",
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"The minimum amount of data to read in each iteration in microseconds, "
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"The minimum amount of data to read in each iteration in "
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"this is the minimum latency that the source reports", 1,
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"microseconds, this is the minimum latency that the source reports",
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G_MAXINT64, DEFAULT_LATENCY_TIME,
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1, G_MAXINT64, DEFAULT_LATENCY_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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/**
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@ -895,7 +895,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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segments_written = g_atomic_int_get (&ringbuffer->segdone);
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segments_written = g_atomic_int_get (&ringbuffer->segdone);
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/* subtract the base to segments_written to get the number of the
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/* subtract the base to segments_written to get the number of the
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last written segment in the ringbuffer (one segment written = segment 0) */
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* last written segment in the ringbuffer
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* (one segment written = segment 0) */
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last_written_segment = segments_written - ringbuffer->segbase - 1;
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last_written_segment = segments_written - ringbuffer->segbase - 1;
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/* samples per segment */
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/* samples per segment */
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@ -910,7 +911,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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/* get the running_time */
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/* get the running_time */
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running_time = current_time - base_time;
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running_time = current_time - base_time;
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/* the running_time converted to a sample (relative to the ringbuffer) */
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/* the running_time converted to a sample
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* (relative to the ringbuffer) */
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running_time_sample =
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running_time_sample =
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gst_util_uint64_scale_int (running_time, rate, GST_SECOND);
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gst_util_uint64_scale_int (running_time, rate, GST_SECOND);
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@ -920,7 +922,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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/* the segment currently read from the ringbuffer */
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/* the segment currently read from the ringbuffer */
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last_read_segment = sample / sps;
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last_read_segment = sample / sps;
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/* the skew we have between running_time and the ringbuffertime (last written to) */
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/* the skew we have between running_time and the ringbuffertime
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* (last written to) */
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segment_skew = running_time_segment - last_written_segment;
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segment_skew = running_time_segment - last_written_segment;
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GST_DEBUG_OBJECT (bsrc,
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GST_DEBUG_OBJECT (bsrc,
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@ -983,9 +986,10 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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{
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{
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GstClockTime base_time, latency;
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GstClockTime base_time, latency;
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/* We are slaved to another clock, take running time of the pipeline clock and
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/* We are slaved to another clock, take running time of the pipeline
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* timestamp against it. Somebody else in the pipeline should figure out the
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* clock and timestamp against it. Somebody else in the pipeline should
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* clock drift. We keep the duration we calculated above. */
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* figure out the clock drift. We keep the duration we calculated
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* above. */
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timestamp = gst_clock_get_time (clock);
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timestamp = gst_clock_get_time (clock);
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base_time = GST_ELEMENT_CAST (src)->base_time;
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base_time = GST_ELEMENT_CAST (src)->base_time;
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@ -1011,7 +1015,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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/* the read method returned a timestamp so we use this instead */
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/* the read method returned a timestamp so we use this instead */
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timestamp = rb_timestamp;
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timestamp = rb_timestamp;
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} else {
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} else {
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/* to get the timestamp against the clock we also need to add our offset */
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/* to get the timestamp against the clock we also need to add our
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* offset */
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timestamp = gst_audio_clock_adjust (clock, timestamp);
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timestamp = gst_audio_clock_adjust (clock, timestamp);
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}
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}
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@ -1085,9 +1090,9 @@ got_error:
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* gst_audio_base_src_create_ringbuffer:
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* gst_audio_base_src_create_ringbuffer:
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* @src: a #GstAudioBaseSrc.
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* @src: a #GstAudioBaseSrc.
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*
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*
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* Create and return the #GstAudioRingBuffer for @src. This function will call the
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* Create and return the #GstAudioRingBuffer for @src. This function will call
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* ::create_ringbuffer vmethod and will set @src as the parent of the returned
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* the ::create_ringbuffer vmethod and will set @src as the parent of the
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* buffer (see gst_object_set_parent()).
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* returned buffer (see gst_object_set_parent()).
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*
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*
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* Returns: (transfer none): The new ringbuffer of @src.
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* Returns: (transfer none): The new ringbuffer of @src.
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*/
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*/
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