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gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time), (gst_base_audio_src_fixate), (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset), (gst_base_audio_src_create), (gst_base_audio_src_change_state): Do the delay calculation in the source/sink base classes as this is specific for the capture/playback mode. Try to fixate a bit better, like round depth up to a multiple of 8 bigger than width. Handle underruns correctly by marking DISCONT on buffers and adjusting timestamps to handle the gap. Set offset/offset_end correctly on buffers. * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause), (gst_ring_buffer_samples_done), (gst_ring_buffer_commit), (gst_ring_buffer_read): Remove resync and underrun recovery from the ringbuffer. Fix ringbuffer read code on under/overrun.
This commit is contained in:
Wim Taymans
parent
102ec386b1
commit
65b1938b38
4 changed files with 163 additions and 65 deletions
22
ChangeLog
22
ChangeLog
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@ -1,3 +1,25 @@
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2006-09-15 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
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(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
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(gst_base_audio_src_create), (gst_base_audio_src_change_state):
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Do the delay calculation in the source/sink base classes as this is
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specific for the capture/playback mode.
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Try to fixate a bit better, like round depth up to a multiple of 8
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bigger than width.
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Handle underruns correctly by marking DISCONT on buffers and adjusting
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timestamps to handle the gap.
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Set offset/offset_end correctly on buffers.
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
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(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
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(gst_ring_buffer_read):
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Remove resync and underrun recovery from the ringbuffer.
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Fix ringbuffer read code on under/overrun.
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2006-09-15 Wim Taymans <wim@fluendo.com>
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
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@ -91,7 +91,7 @@ static void gst_base_audio_sink_get_times (GstBaseSink * bsink,
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static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink,
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GstCaps * caps);
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//static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 };
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/* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
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static void
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gst_base_audio_sink_base_init (gpointer g_class)
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@ -217,18 +217,32 @@ clock_disabled:
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static GstClockTime
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gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
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{
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guint64 samples;
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guint64 raw, samples;
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guint delay;
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GstClockTime result;
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if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
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return GST_CLOCK_TIME_NONE;
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/* our processed samples are always increasing */
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samples = gst_ring_buffer_samples_done (sink->ringbuffer);
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raw = samples = gst_ring_buffer_samples_done (sink->ringbuffer);
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/* the number of samples not yet processed, this is still queued in the
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* device (not played for playback). */
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delay = gst_ring_buffer_delay (sink->ringbuffer);
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if (G_LIKELY (samples >= delay))
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samples -= delay;
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else
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samples = 0;
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result = gst_util_uint64_scale_int (samples, GST_SECOND,
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sink->ringbuffer->spec.rate);
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GST_DEBUG_OBJECT (sink,
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"processed samples: raw %llu, delay %u, real %llu, time %"
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GST_TIME_FORMAT, raw, delay, samples, GST_TIME_ARGS (result));
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return result;
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}
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@ -707,7 +721,7 @@ static void
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gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
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gpointer user_data)
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{
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//GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (data);
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/* GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (data); */
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}
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/* should be called with the LOCK */
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@ -75,7 +75,7 @@ static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
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//static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 };
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/* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
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static void
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gst_base_audio_src_base_init (gpointer g_class)
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@ -201,17 +201,28 @@ wrong_state:
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static GstClockTime
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gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src)
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{
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guint64 samples;
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guint64 raw, samples;
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guint delay;
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GstClockTime result;
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if (G_UNLIKELY (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0))
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return GST_CLOCK_TIME_NONE;
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samples = gst_ring_buffer_samples_done (src->ringbuffer);
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raw = samples = gst_ring_buffer_samples_done (src->ringbuffer);
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/* the number of samples not yet processed, this is still queued in the
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* device (not yet read for capture). */
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delay = gst_ring_buffer_delay (src->ringbuffer);
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samples += delay;
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result = gst_util_uint64_scale_int (samples, GST_SECOND,
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src->ringbuffer->spec.rate);
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GST_DEBUG_OBJECT (src,
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"processed samples: raw %llu, delay %u, real %llu, time %"
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GST_TIME_FORMAT, raw, delay, samples, GST_TIME_ARGS (result));
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return result;
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}
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@ -271,14 +282,24 @@ static void
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gst_base_audio_src_fixate (GstPad * pad, GstCaps * caps)
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{
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GstStructure *s;
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gint width, depth;
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s = gst_caps_get_structure (caps, 0);
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/* fields for all formats */
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gst_structure_fixate_field_nearest_int (s, "rate", 44100);
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gst_structure_fixate_field_nearest_int (s, "channels", 2);
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gst_structure_fixate_field_nearest_int (s, "depth", 16);
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gst_structure_fixate_field_nearest_int (s, "width", 16);
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gst_structure_set (s, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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/* fields for int */
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if (gst_structure_has_field (s, "depth")) {
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gst_structure_get_int (s, "width", &width);
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/* round width to nearest multiple of 8 for the depth */
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depth = GST_ROUND_UP_8 (width);
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gst_structure_fixate_field_nearest_int (s, "depth", depth);
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}
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if (gst_structure_has_field (s, "signed"))
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gst_structure_fixate_field_boolean (s, "signed", TRUE);
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if (gst_structure_has_field (s, "endianness"))
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gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
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}
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@ -341,9 +362,8 @@ static void
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gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end)
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{
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/* ne need to sync to a clock here, we schedule the samples based
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* on our own clock for the moment. FIXME, implement this when
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* we are not using our own clock */
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/* no need to sync to a clock here, we schedule the samples based
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* on our own clock for the moment. */
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*start = GST_CLOCK_TIME_NONE;
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*end = GST_CLOCK_TIME_NONE;
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}
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@ -369,6 +389,45 @@ gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
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return TRUE;
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}
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/* get the next offset in the ringbuffer for reading samples.
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* If the next sample is too far away, this function will position itself to the
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* next most recent sample, creating discontinuity */
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static guint64
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gst_base_audio_src_get_offset (GstBaseAudioSrc * src)
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{
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guint64 sample;
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gint readseg, segdone, segtotal, sps;
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gint diff;
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/* assume we can append to the previous sample */
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sample = src->next_sample;
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/* no previous sample, try to read from position 0 */
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if (sample == -1)
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sample = 0;
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sps = src->ringbuffer->samples_per_seg;
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segtotal = src->ringbuffer->spec.segtotal;
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/* figure out the segment and the offset inside the segment where
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* the sample should be read from. */
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readseg = sample / sps;
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/* get the currently processed segment */
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segdone = g_atomic_int_get (&src->ringbuffer->segdone)
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- src->ringbuffer->segbase;
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/* see how far away it is from the read segment, normally segdone (where new
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* data is written in the ringbuffer) is bigger than readseg (where we are
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* reading). */
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diff = segdone - readseg;
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if (diff >= segtotal) {
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/* sample would be dropped, position to next playable position */
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sample = (segdone - segtotal + 1) * sps;
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}
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return sample;
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}
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static GstFlowReturn
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gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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GstBuffer ** outbuf)
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@ -396,14 +455,17 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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/* make sure we round down to an integral number of samples */
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length -= length % bps;
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/* calculate the sequentially next sample we need to read */
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sample = (src->next_sample != -1 ? src->next_sample : 0);
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/* figure out the offset in the ringbuffer */
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if (G_UNLIKELY (offset != -1)) {
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/* if a specific offset was given it must be the next
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* sequential offset we expect or we fail. */
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if (offset / bps != sample)
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sample = offset / bps;
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/* if a specific offset was given it must be the next sequential
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* offset we expect or we fail for now. */
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if (src->next_sample != -1 && sample != src->next_sample)
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goto wrong_offset;
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} else {
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/* calculate the sequentially next sample we need to read. This can jump and
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* create a DISCONT. */
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sample = gst_base_audio_src_get_offset (src);
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}
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/* get the number of samples to read */
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@ -413,10 +475,19 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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buf = gst_buffer_new_and_alloc (length);
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data = GST_BUFFER_DATA (buf);
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/* read the sample */
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res = gst_ring_buffer_read (ringbuffer, sample, data, samples);
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if (G_UNLIKELY (res == -1))
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goto stopped;
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/* mark discontinuity if needed */
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if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {
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GST_WARNING_OBJECT (src,
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"create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
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G_GUINT64_FORMAT, sample - src->next_sample, sample);
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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}
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/* FIXME, we timestamp against our own clock, also handle the case
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* where we are slaved to another clock. We currently refuse to accept
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* any other clock than the one we provide, so this code is fine for
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@ -426,6 +497,8 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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src->next_sample = sample + samples;
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GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (src->next_sample,
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GST_SECOND, ringbuffer->spec.rate) - GST_BUFFER_TIMESTAMP (buf);
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GST_BUFFER_OFFSET (buf) = sample;
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GST_BUFFER_OFFSET_END (buf) = sample + samples;
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gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (bsrc)));
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@ -470,13 +543,6 @@ gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
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return buffer;
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}
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void
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gst_base_audio_src_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
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gpointer user_data)
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{
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//GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (data);
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}
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static GstStateChangeReturn
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gst_base_audio_src_change_state (GstElement * element,
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GstStateChange transition)
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@ -488,12 +554,10 @@ gst_base_audio_src_change_state (GstElement * element,
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case GST_STATE_CHANGE_NULL_TO_READY:
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if (src->ringbuffer == NULL) {
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src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
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gst_ring_buffer_set_callback (src->ringbuffer,
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gst_base_audio_src_callback, src);
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}
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if (!gst_ring_buffer_open_device (src->ringbuffer))
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return GST_STATE_CHANGE_FAILURE;
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src->next_sample = 0;
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src->next_sample = -1;
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
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@ -515,7 +579,7 @@ gst_base_audio_src_change_state (GstElement * element,
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
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gst_ring_buffer_release (src->ringbuffer);
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src->next_sample = 0;
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src->next_sample = -1;
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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gst_ring_buffer_close_device (src->ringbuffer);
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@ -920,6 +920,7 @@ gst_ring_buffer_pause (GstRingBuffer * buf)
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return res;
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/* ERRORS */
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flushing:
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{
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GST_OBJECT_UNLOCK (buf);
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@ -989,6 +990,12 @@ done:
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* implementation uses another internal buffer between the audio
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* device.
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*
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* For playback ringbuffers this is the amount of samples transfered from the
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* ringbuffer to the device but still not played.
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*
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* For capture ringbuffers this is the amount of samples in the device that are
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* not yet transfered to the ringbuffer.
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*
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* Returns: The number of samples queued in the audio device.
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*
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* MT safe.
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@ -1020,7 +1027,8 @@ done:
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* @buf: the #GstRingBuffer to query
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*
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* Get the number of samples that were processed by the ringbuffer
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* since it was last started.
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* since it was last started. This does not include the number of samples not
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* yet processed (see gst_ring_buffer_delay()).
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*
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* Returns: The number of samples processed by the ringbuffer.
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*
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@ -1038,18 +1046,8 @@ gst_ring_buffer_samples_done (GstRingBuffer * buf)
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/* get the amount of segments we processed */
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segdone = g_atomic_int_get (&buf->segdone);
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/* and the number of samples not yet processed */
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delay = gst_ring_buffer_delay (buf);
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raw = samples = ((guint64) segdone) * buf->samples_per_seg;
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if (G_LIKELY (samples >= delay))
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samples -= delay;
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else
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samples = 0;
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GST_DEBUG_OBJECT (buf, "processed samples: raw %llu, delay %u, real %llu",
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raw, delay, samples);
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/* convert to samples */
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samples = ((guint64) segdone) * buf->samples_per_seg;
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return samples;
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}
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@ -1237,9 +1235,10 @@ gst_ring_buffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
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GST_DEBUG
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("pointer at %d, sample %llu, write to %d-%d, to_write %d, diff %d, segtotal %d, segsize %d",
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segdone, sample, writeseg, sampleoff, to_write, diff, segtotal, sps);
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segdone, sample, writeseg, sampleoff, to_write, diff, segtotal,
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segsize);
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/* segment too far ahead, we need to drop, hopefully UNLIKELY */
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/* segment too far ahead, writer too slow, we need to drop, hopefully UNLIKELY */
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if (G_UNLIKELY (diff < 0)) {
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/* we need to drop one segment at a time, pretend we wrote a
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* segment. */
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@ -1310,6 +1309,7 @@ gst_ring_buffer_read (GstRingBuffer * buf, guint64 sample, guchar * data,
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gint segdone;
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gint segsize, segtotal, bps, sps;
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guint8 *dest;
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guint to_read;
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g_return_val_if_fail (GST_IS_RING_BUFFER (buf), -1);
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g_return_val_if_fail (buf->data != NULL, -1);
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@ -1321,13 +1321,14 @@ gst_ring_buffer_read (GstRingBuffer * buf, guint64 sample, guchar * data,
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bps = buf->spec.bytes_per_sample;
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sps = buf->samples_per_seg;
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to_read = len;
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/* read enough samples */
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while (len > 0) {
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while (to_read > 0) {
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gint sampleslen;
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gint readseg, sampleoff;
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/* figure out the segment and the offset inside the segment where
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* the sample should be written. */
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* the sample should be read from. */
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readseg = sample / sps;
|
||||
sampleoff = (sample % sps);
|
||||
|
||||
|
|
@ -1337,33 +1338,29 @@ gst_ring_buffer_read (GstRingBuffer * buf, guint64 sample, guchar * data,
|
|||
/* get the currently processed segment */
|
||||
segdone = g_atomic_int_get (&buf->segdone) - buf->segbase;
|
||||
|
||||
/* see how far away it is from the read segment */
|
||||
/* see how far away it is from the read segment, normally segdone (where
|
||||
* the hardware is writing) is bigger than readseg (where software is
|
||||
* reading) */
|
||||
diff = segdone - readseg;
|
||||
|
||||
GST_DEBUG
|
||||
("pointer at %d, sample %llu, read from %d-%d, len %d, diff %d, segtotal %d, segsize %d",
|
||||
segdone, sample, readseg, sampleoff, len, diff, segtotal, segsize);
|
||||
("pointer at %d, sample %llu, read from %d-%d, to_read %d, diff %d, segtotal %d, segsize %d",
|
||||
segdone, sample, readseg, sampleoff, to_read, diff, segtotal,
|
||||
segsize);
|
||||
|
||||
/* segment too far ahead, we need to drop */
|
||||
if (diff < 0) {
|
||||
/* we need to drop one segment at a time, pretend we read an
|
||||
* empty segment. */
|
||||
sampleslen = MIN (sps, len);
|
||||
/* segment too far ahead, reader too slow */
|
||||
if (G_UNLIKELY (diff >= segtotal)) {
|
||||
/* pretend we read an empty segment. */
|
||||
sampleslen = MIN (sps, to_read);
|
||||
memcpy (data, buf->empty_seg, sampleslen * bps);
|
||||
goto next;
|
||||
}
|
||||
|
||||
/* read segment is within readable range, we can break the loop and
|
||||
* start reading the data. */
|
||||
if (diff > 0 && diff < segtotal)
|
||||
if (diff > 0)
|
||||
break;
|
||||
|
||||
/* flush if diff has grown bigger than ringbuffer */
|
||||
if (diff >= segtotal) {
|
||||
gst_ring_buffer_clear_all (buf);
|
||||
buf->segdone = readseg;
|
||||
}
|
||||
|
||||
/* else we need to wait for the segment to become readable. */
|
||||
if (!wait_segment (buf))
|
||||
goto not_started;
|
||||
|
|
@ -1371,26 +1368,27 @@ gst_ring_buffer_read (GstRingBuffer * buf, guint64 sample, guchar * data,
|
|||
|
||||
/* we can read now */
|
||||
readseg = readseg % segtotal;
|
||||
sampleslen = MIN (sps - sampleoff, len);
|
||||
sampleslen = MIN (sps - sampleoff, to_read);
|
||||
|
||||
GST_DEBUG_OBJECT (buf, "read @%p seg %d, off %d, len %d",
|
||||
GST_DEBUG_OBJECT (buf, "read @%p seg %d, off %d, sampleslen %d",
|
||||
dest + readseg * segsize, readseg, sampleoff, sampleslen);
|
||||
|
||||
memcpy (data, dest + (readseg * segsize) + (sampleoff * bps),
|
||||
(sampleslen * bps));
|
||||
|
||||
next:
|
||||
len -= sampleslen;
|
||||
to_read -= sampleslen;
|
||||
sample += sampleslen;
|
||||
data += sampleslen * bps;
|
||||
}
|
||||
|
||||
return len;
|
||||
return len - to_read;
|
||||
|
||||
/* ERRORS */
|
||||
not_started:
|
||||
{
|
||||
GST_DEBUG_OBJECT (buf, "stopped processing");
|
||||
/* FIXME, return len - to_read after fixing caller */
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in a new issue