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audiobuffersplit: Add a gapless mode which inserts silence/drops samples on disconts

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The output is always a continguous stream without any gaps.
This commit is contained in:
Sebastian Dröge 2018-08-17 16:37:45 +03:00
parent 2f761b89df
commit f19edc8c83
2 changed files with 178 additions and 11 deletions
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187
gst/audiobuffersplit/gstaudiobuffersplit.c
View file

@ -46,6 +46,7 @@ enum
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
PROP_STRICT_BUFFER_SIZE,
PROP_GAPLESS,
LAST_PROP
};
@ -54,6 +55,7 @@ enum
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
#define DEFAULT_STRICT_BUFFER_SIZE (FALSE)
#define DEFAULT_GAPLESS (FALSE)
#define parent_class gst_audio_buffer_split_parent_class
G_DEFINE_TYPE (GstAudioBufferSplit, gst_audio_buffer_split, GST_TYPE_ELEMENT);
@ -114,6 +116,13 @@ gst_audio_buffer_split_class_init (GstAudioBufferSplitClass * klass)
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_GAPLESS,
g_param_spec_boolean ("gapless", "Gapless",
"Insert silence/drop samples instead of creating a discontinuity",
DEFAULT_GAPLESS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
gst_element_class_set_static_metadata (gstelement_class,
"Audio Buffer Split", "Audio/Filter",
"Splits raw audio buffers into equal sized chunks",
@ -148,6 +157,7 @@ gst_audio_buffer_split_init (GstAudioBufferSplit * self)
self->output_buffer_duration_n = DEFAULT_OUTPUT_BUFFER_DURATION_N;
self->output_buffer_duration_d = DEFAULT_OUTPUT_BUFFER_DURATION_D;
self->strict_buffer_size = DEFAULT_STRICT_BUFFER_SIZE;
self->gapless = DEFAULT_GAPLESS;
self->adapter = gst_adapter_new ();
@ -240,6 +250,9 @@ gst_audio_buffer_split_set_property (GObject * object, guint property_id,
case PROP_STRICT_BUFFER_SIZE:
self->strict_buffer_size = g_value_get_boolean (value);
break;
case PROP_GAPLESS:
self->gapless = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
@ -272,6 +285,9 @@ gst_audio_buffer_split_get_property (GObject * object, guint property_id,
case PROP_STRICT_BUFFER_SIZE:
g_value_set_boolean (value, self->strict_buffer_size);
break;
case PROP_GAPLESS:
g_value_set_boolean (value, self->gapless);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
@ -399,7 +415,8 @@ gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
static GstFlowReturn
gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
GstBuffer * buffer, gint rate, gint bpf, guint samples_per_buffer)
GstBuffer * buffer, GstAudioFormat format, gint rate, gint bpf,
guint samples_per_buffer)
{
gboolean discont;
GstFlowReturn ret = GST_FLOW_OK;
@ -414,18 +431,125 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
GST_OBJECT_UNLOCK (self);
if (discont) {
if (self->strict_buffer_size) {
gst_adapter_clear (self->adapter);
ret = GST_FLOW_OK;
guint avail = gst_adapter_available (self->adapter);
guint avail_samples = avail / bpf;
guint64 new_offset;
GstClockTime current_timestamp;
GstClockTime current_timestamp_end;
/* Reset */
self->drop_samples = 0;
if (self->segment.rate < 0.0) {
current_timestamp =
self->resync_time - gst_util_uint64_scale (self->current_offset +
avail_samples, GST_SECOND, rate);
current_timestamp_end =
self->resync_time - gst_util_uint64_scale (self->current_offset,
GST_SECOND, rate);
} else {
ret =
gst_audio_buffer_split_output (self, TRUE, rate, bpf,
samples_per_buffer);
current_timestamp =
self->resync_time + gst_util_uint64_scale (self->current_offset,
GST_SECOND, rate);
current_timestamp_end =
self->resync_time + gst_util_uint64_scale (self->current_offset +
avail_samples, GST_SECOND, rate);
}
self->current_offset = 0;
self->accumulated_error = 0;
self->resync_time = GST_BUFFER_PTS (buffer);
if (self->gapless) {
if (self->current_offset == -1) {
/* We only set resync time on the very first buffer */
self->current_offset = 0;
self->resync_time = GST_BUFFER_PTS (buffer);
} else {
GST_DEBUG_OBJECT (self,
"Got discont in gapless mode: Current timestamp %" GST_TIME_FORMAT
", current end timestamp %" GST_TIME_FORMAT
", timestamp after discont %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_timestamp),
GST_TIME_ARGS (current_timestamp_end),
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
new_offset =
gst_util_uint64_scale (GST_BUFFER_PTS (buffer) - self->resync_time,
rate, GST_SECOND);
if (GST_BUFFER_PTS (buffer) < self->resync_time) {
guint64 drop_samples;
new_offset =
gst_util_uint64_scale (self->resync_time -
GST_BUFFER_PTS (buffer), rate, GST_SECOND);
drop_samples = self->current_offset + avail_samples + new_offset;
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
} else if (new_offset > self->current_offset + avail_samples) {
guint64 silence_samples =
new_offset - (self->current_offset + avail_samples);
const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
GST_DEBUG_OBJECT (self,
"Inserting %" G_GUINT64_FORMAT " samples of silence (%"
GST_TIME_FORMAT ")", silence_samples,
GST_TIME_ARGS (gst_util_uint64_scale (silence_samples, GST_SECOND,
rate)));
/* Insert silence buffers to fill the gap in 1s chunks */
while (silence_samples > 0) {
guint n_samples = MIN (silence_samples, rate);
GstBuffer *silence;
GstMapInfo map;
silence = gst_buffer_new_and_alloc (n_samples * bpf);
GST_BUFFER_FLAG_SET (silence, GST_BUFFER_FLAG_GAP);
gst_buffer_map (silence, &map, GST_MAP_WRITE);
gst_audio_format_fill_silence (info, map.data, map.size);
gst_buffer_unmap (silence, &map);
gst_adapter_push (self->adapter, silence);
ret =
gst_audio_buffer_split_output (self, FALSE, rate, bpf,
samples_per_buffer);
if (ret != GST_FLOW_OK)
return ret;
silence_samples -= n_samples;
}
} else if (new_offset < self->current_offset + avail_samples) {
guint64 drop_samples =
self->current_offset + avail_samples - new_offset;
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
self->drop_samples = drop_samples;
}
}
} else {
GST_DEBUG_OBJECT (self,
"Got discont: Current timestamp %" GST_TIME_FORMAT
", current end timestamp %" GST_TIME_FORMAT
", timestamp after discont %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_timestamp),
GST_TIME_ARGS (current_timestamp_end),
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
if (self->strict_buffer_size) {
gst_adapter_clear (self->adapter);
ret = GST_FLOW_OK;
} else {
ret =
gst_audio_buffer_split_output (self, TRUE, rate, bpf,
samples_per_buffer);
}
self->current_offset = 0;
self->accumulated_error = 0;
self->resync_time = GST_BUFFER_PTS (buffer);
}
}
return ret;
@ -438,6 +562,41 @@ gst_audio_buffer_split_clip_buffer (GstAudioBufferSplit * self,
return gst_audio_buffer_clip (buffer, segment, rate, bpf);
}
static GstBuffer *
gst_audio_buffer_split_clip_buffer_start_for_gapless (GstAudioBufferSplit *
self, GstBuffer * buffer, gint rate, gint bpf)
{
guint nsamples;
if (!self->gapless || self->drop_samples == 0)
return buffer;
nsamples = gst_buffer_get_size (buffer) / bpf;
GST_DEBUG_OBJECT (self, "Have to drop %lu samples, got %u samples",
self->drop_samples, nsamples);
if (nsamples <= self->drop_samples) {
gst_buffer_unref (buffer);
self->drop_samples -= nsamples;
return NULL;
}
if (self->segment.rate < 0.0) {
buffer =
gst_audio_buffer_truncate (buffer, bpf, 0,
nsamples - self->drop_samples);
self->drop_samples = 0;
return buffer;
} else {
buffer = gst_audio_buffer_truncate (buffer, bpf, self->drop_samples, -1);
self->drop_samples = 0;
return buffer;
}
return buffer;
}
static GstFlowReturn
gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
@ -468,13 +627,19 @@ gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
return GST_FLOW_OK;
ret =
gst_audio_buffer_split_handle_discont (self, buffer, rate, bpf,
gst_audio_buffer_split_handle_discont (self, buffer, format, rate, bpf,
samples_per_buffer);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
return ret;
}
buffer =
gst_audio_buffer_split_clip_buffer_start_for_gapless (self, buffer, rate,
bpf);
if (!buffer)
return GST_FLOW_OK;
gst_adapter_push (self->adapter, buffer);
return gst_audio_buffer_split_output (self, FALSE, rate, bpf,

2
gst/audiobuffersplit/gstaudiobuffersplit.h
View file

@ -55,12 +55,14 @@ struct _GstAudioBufferSplit {
GstAudioStreamAlign *stream_align;
GstClockTime resync_time;
guint64 current_offset; /* offset from start time in samples */
guint64 drop_samples; /* number of samples to drop in gapless mode */
guint samples_per_buffer;
guint error_per_buffer;
guint accumulated_error;
gboolean strict_buffer_size;
gboolean gapless;
};
struct _GstAudioBufferSplitClass {