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audiobuffersplit: Allow changing the output-buffer-duration at any time

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Previously this was only allowed before the CAPS event arrived.
This commit is contained in:
Sebastian Dröge 2017-05-11 18:39:46 +02:00
parent a7737b6dce
commit 6026d12d90
1 changed files with 87 additions and 40 deletions
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127
gst/audiobuffersplit/gstaudiobuffersplit.c
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@ -167,6 +167,44 @@ gst_audio_buffer_split_finalize (GObject * object)
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_audio_buffer_split_update_samples_per_buffer (GstAudioBufferSplit * self)
{
gboolean ret = TRUE;
GST_OBJECT_LOCK (self);
/* For a later time */
if (!self->info.finfo
|| GST_AUDIO_INFO_FORMAT (&self->info) == GST_AUDIO_FORMAT_UNKNOWN) {
self->samples_per_buffer = 0;
goto out;
}
self->samples_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
self->output_buffer_duration_n) / self->output_buffer_duration_d;
if (self->samples_per_buffer == 0) {
ret = FALSE;
goto out;
}
self->error_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
self->output_buffer_duration_n) % self->output_buffer_duration_d;
self->accumulated_error = 0;
GST_DEBUG_OBJECT (self, "Buffer duration: %u/%u",
self->output_buffer_duration_n, self->output_buffer_duration_d);
GST_DEBUG_OBJECT (self, "Samples per buffer: %u (error: %u/%u)",
self->samples_per_buffer, self->error_per_buffer,
self->output_buffer_duration_d);
out:
GST_OBJECT_UNLOCK (self);
return ret;
}
static void
gst_audio_buffer_split_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
@ -178,6 +216,7 @@ gst_audio_buffer_split_set_property (GObject * object, guint property_id,
self->output_buffer_duration_n = gst_value_get_fraction_numerator (value);
self->output_buffer_duration_d =
gst_value_get_fraction_denominator (value);
gst_audio_buffer_split_update_samples_per_buffer (self);
break;
case PROP_ALIGNMENT_THRESHOLD:
self->alignment_threshold = g_value_get_uint64 (value);
@ -260,19 +299,13 @@ gst_audio_buffer_split_change_state (GstElement * element,
}
static GstFlowReturn
gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force)
gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
gint rate, gint bpf, guint samples_per_buffer)
{
gint rate, bpf;
gint size, avail;
GstFlowReturn ret = GST_FLOW_OK;
rate = GST_AUDIO_INFO_RATE (&self->info);
bpf = GST_AUDIO_INFO_BPF (&self->info);
if (self->samples_per_buffer == 0)
return GST_FLOW_NOT_NEGOTIATED;
size = self->samples_per_buffer * bpf;
size = samples_per_buffer * bpf;
/* If we accumulated enough error for one sample, include one
* more sample in this buffer. Accumulated error is updated below */
@ -332,19 +365,16 @@ gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force)
static GstFlowReturn
gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
GstBuffer * buffer)
GstBuffer * buffer, gint rate, gint bpf, guint samples_per_buffer)
{
GstClockTime timestamp;
gsize size;
guint64 start_offset, end_offset;
gint rate, bpf;
gboolean discont = FALSE;
GstFlowReturn ret = GST_FLOW_OK;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
rate = GST_AUDIO_INFO_RATE (&self->info);
bpf = GST_AUDIO_INFO_BPF (&self->info);
start_offset = gst_util_uint64_scale (timestamp, rate, GST_SECOND);
size = gst_buffer_get_size (buffer);
end_offset = start_offset + size / bpf;
@ -399,7 +429,9 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
gst_adapter_clear (self->adapter);
ret = GST_FLOW_OK;
} else {
ret = gst_audio_buffer_split_output (self, TRUE);
ret =
gst_audio_buffer_split_output (self, TRUE, rate, bpf,
samples_per_buffer);
}
}
self->next_offset = end_offset;
@ -423,10 +455,9 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
static GstBuffer *
gst_audio_buffer_split_clip_buffer (GstAudioBufferSplit * self,
GstBuffer * buffer)
GstBuffer * buffer, const GstSegment * segment, gint rate, gint bpf)
{
return gst_audio_buffer_clip (buffer, &self->segment,
GST_AUDIO_INFO_RATE (&self->info), GST_AUDIO_INFO_BPF (&self->info));
return gst_audio_buffer_clip (buffer, segment, rate, bpf);
}
static GstFlowReturn
@ -435,17 +466,32 @@ gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
GstFlowReturn ret;
GstAudioFormat format;
gint rate, bpf, samples_per_buffer;
if (GST_AUDIO_INFO_FORMAT (&self->info) == GST_AUDIO_FORMAT_UNKNOWN) {
GST_OBJECT_LOCK (self);
format =
self->info.
finfo ? GST_AUDIO_INFO_FORMAT (&self->info) : GST_AUDIO_FORMAT_UNKNOWN;
rate = GST_AUDIO_INFO_RATE (&self->info);
bpf = GST_AUDIO_INFO_BPF (&self->info);
samples_per_buffer = self->samples_per_buffer;
GST_OBJECT_UNLOCK (self);
if (format == GST_AUDIO_FORMAT_UNKNOWN || samples_per_buffer == 0) {
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
buffer = gst_audio_buffer_split_clip_buffer (self, buffer);
buffer =
gst_audio_buffer_split_clip_buffer (self, buffer, &self->segment, rate,
bpf);
if (!buffer)
return GST_FLOW_OK;
ret = gst_audio_buffer_split_handle_discont (self, buffer);
ret =
gst_audio_buffer_split_handle_discont (self, buffer, rate, bpf,
samples_per_buffer);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
return ret;
@ -453,7 +499,8 @@ gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
gst_adapter_push (self->adapter, buffer);
return gst_audio_buffer_split_output (self, FALSE);
return gst_audio_buffer_split_output (self, FALSE, rate, bpf,
samples_per_buffer);
}
static gboolean
@ -472,23 +519,8 @@ gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
ret = gst_audio_info_from_caps (&self->info, caps);
if (ret) {
self->samples_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
self->output_buffer_duration_n) / self->output_buffer_duration_d;
if (self->samples_per_buffer == 0)
ret = FALSE;
self->error_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
self->output_buffer_duration_n) % self->output_buffer_duration_d;
self->accumulated_error = 0;
GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps);
GST_DEBUG_OBJECT (self, "Buffer duration: %u/%u",
self->output_buffer_duration_n, self->output_buffer_duration_d);
GST_DEBUG_OBJECT (self, "Samples per buffer: %u (error: %u/%u)",
self->samples_per_buffer, self->error_per_buffer,
self->output_buffer_duration_d);
ret = gst_audio_buffer_split_update_samples_per_buffer (self);
} else {
ret = FALSE;
}
@ -520,10 +552,25 @@ gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
}
break;
case GST_EVENT_EOS:
if (self->strict_buffer_size)
if (self->strict_buffer_size) {
gst_adapter_clear (self->adapter);
else
gst_audio_buffer_split_output (self, TRUE);
} else {
GstAudioFormat format;
gint rate, bpf, samples_per_buffer;
GST_OBJECT_LOCK (self);
format =
self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
GST_AUDIO_FORMAT_UNKNOWN;
rate = GST_AUDIO_INFO_RATE (&self->info);
bpf = GST_AUDIO_INFO_BPF (&self->info);
samples_per_buffer = self->samples_per_buffer;
GST_OBJECT_UNLOCK (self);
if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
gst_audio_buffer_split_output (self, TRUE, rate, bpf,
samples_per_buffer);
}
ret = gst_pad_event_default (pad, parent, event);
break;
default: