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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 18:21:04 +00:00
audiobuffersplit: Added max-silence-time property
This commit is contained in:
Vivia Nikolaidou
Sebastian Dröge
parent
7c767f3fcd
commit
ce0be4d1ac
2 changed files with 117 additions and 82 deletions
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@ -47,6 +47,7 @@ enum
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PROP_DISCONT_WAIT,
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PROP_STRICT_BUFFER_SIZE,
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PROP_GAPLESS,
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PROP_MAX_SILENCE_TIME,
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LAST_PROP
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};
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@ -56,6 +57,7 @@ enum
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#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
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#define DEFAULT_STRICT_BUFFER_SIZE (FALSE)
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#define DEFAULT_GAPLESS (FALSE)
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#define DEFAULT_MAX_SILENCE_TIME (0)
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#define parent_class gst_audio_buffer_split_parent_class
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G_DEFINE_TYPE (GstAudioBufferSplit, gst_audio_buffer_split, GST_TYPE_ELEMENT);
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@ -123,6 +125,15 @@ gst_audio_buffer_split_class_init (GstAudioBufferSplitClass * klass)
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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g_object_class_install_property (gobject_class, PROP_MAX_SILENCE_TIME,
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g_param_spec_uint64 ("max-silence-time",
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"Maximum time of silence to insert",
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"Do not insert silence in gapless mode if the gap exceeds this "
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"period (in ns) (0 = disabled)",
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0, G_MAXUINT64, DEFAULT_MAX_SILENCE_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY));
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gst_element_class_set_static_metadata (gstelement_class,
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"Audio Buffer Split", "Audio/Filter",
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"Splits raw audio buffers into equal sized chunks",
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@ -253,6 +264,9 @@ gst_audio_buffer_split_set_property (GObject * object, guint property_id,
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case PROP_GAPLESS:
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self->gapless = g_value_get_boolean (value);
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break;
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case PROP_MAX_SILENCE_TIME:
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self->max_silence_time = g_value_get_uint64 (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
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break;
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@ -288,6 +302,9 @@ gst_audio_buffer_split_get_property (GObject * object, guint property_id,
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case PROP_GAPLESS:
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g_value_set_boolean (value, self->gapless);
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break;
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case PROP_MAX_SILENCE_TIME:
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g_value_set_uint64 (value, self->max_silence_time);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
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break;
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@ -420,6 +437,11 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
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{
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gboolean discont;
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GstFlowReturn ret = GST_FLOW_OK;
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guint avail = gst_adapter_available (self->adapter);
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guint avail_samples = avail / bpf;
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guint64 new_offset;
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GstClockTime current_timestamp;
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GstClockTime current_timestamp_end;
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GST_OBJECT_LOCK (self);
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discont =
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@ -430,71 +452,77 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
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NULL);
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GST_OBJECT_UNLOCK (self);
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if (discont) {
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guint avail = gst_adapter_available (self->adapter);
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guint avail_samples = avail / bpf;
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guint64 new_offset;
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GstClockTime current_timestamp;
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GstClockTime current_timestamp_end;
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if (!discont)
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return ret;
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/* Reset */
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self->drop_samples = 0;
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/* Reset */
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self->drop_samples = 0;
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if (self->segment.rate < 0.0) {
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current_timestamp =
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self->resync_time - gst_util_uint64_scale (self->current_offset +
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avail_samples, GST_SECOND, rate);
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current_timestamp_end =
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self->resync_time - gst_util_uint64_scale (self->current_offset,
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GST_SECOND, rate);
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if (self->segment.rate < 0.0) {
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current_timestamp =
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self->resync_time - gst_util_uint64_scale (self->current_offset +
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avail_samples, GST_SECOND, rate);
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current_timestamp_end =
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self->resync_time - gst_util_uint64_scale (self->current_offset,
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GST_SECOND, rate);
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} else {
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current_timestamp =
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self->resync_time + gst_util_uint64_scale (self->current_offset,
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GST_SECOND, rate);
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current_timestamp_end =
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self->resync_time + gst_util_uint64_scale (self->current_offset +
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avail_samples, GST_SECOND, rate);
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}
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if (self->gapless) {
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if (self->current_offset == -1) {
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/* We only set resync time on the very first buffer */
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self->current_offset = 0;
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self->resync_time = GST_BUFFER_PTS (buffer);
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discont = FALSE;
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} else {
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current_timestamp =
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self->resync_time + gst_util_uint64_scale (self->current_offset,
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GST_SECOND, rate);
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current_timestamp_end =
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self->resync_time + gst_util_uint64_scale (self->current_offset +
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avail_samples, GST_SECOND, rate);
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}
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GST_DEBUG_OBJECT (self,
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"Got discont in gapless mode: Current timestamp %" GST_TIME_FORMAT
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", current end timestamp %" GST_TIME_FORMAT
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", timestamp after discont %" GST_TIME_FORMAT,
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GST_TIME_ARGS (current_timestamp),
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GST_TIME_ARGS (current_timestamp_end),
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GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
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if (self->gapless) {
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if (self->current_offset == -1) {
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/* We only set resync time on the very first buffer */
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self->current_offset = 0;
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self->resync_time = GST_BUFFER_PTS (buffer);
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} else {
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GST_DEBUG_OBJECT (self,
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"Got discont in gapless mode: Current timestamp %" GST_TIME_FORMAT
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", current end timestamp %" GST_TIME_FORMAT
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", timestamp after discont %" GST_TIME_FORMAT,
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GST_TIME_ARGS (current_timestamp),
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GST_TIME_ARGS (current_timestamp_end),
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GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
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new_offset =
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gst_util_uint64_scale (GST_BUFFER_PTS (buffer) - self->resync_time,
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rate, GST_SECOND);
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if (GST_BUFFER_PTS (buffer) < self->resync_time) {
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guint64 drop_samples;
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new_offset =
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gst_util_uint64_scale (GST_BUFFER_PTS (buffer) - self->resync_time,
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rate, GST_SECOND);
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if (GST_BUFFER_PTS (buffer) < self->resync_time) {
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guint64 drop_samples;
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gst_util_uint64_scale (self->resync_time -
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GST_BUFFER_PTS (buffer), rate, GST_SECOND);
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drop_samples = self->current_offset + avail_samples + new_offset;
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new_offset =
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gst_util_uint64_scale (self->resync_time -
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GST_BUFFER_PTS (buffer), rate, GST_SECOND);
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drop_samples = self->current_offset + avail_samples + new_offset;
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GST_DEBUG_OBJECT (self,
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"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
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drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
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GST_SECOND, rate)));
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discont = FALSE;
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} else if (new_offset > self->current_offset + avail_samples) {
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guint64 silence_samples =
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new_offset - (self->current_offset + avail_samples);
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const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
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GstClockTime silence_time =
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gst_util_uint64_scale (silence_samples, GST_SECOND, rate);
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if (silence_time > self->max_silence_time) {
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GST_DEBUG_OBJECT (self,
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"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
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drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
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GST_SECOND, rate)));
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} else if (new_offset > self->current_offset + avail_samples) {
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guint64 silence_samples =
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new_offset - (self->current_offset + avail_samples);
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const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
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"Not inserting %" G_GUINT64_FORMAT " samples of silence (%"
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GST_TIME_FORMAT " exceeds maximum %" GST_TIME_FORMAT ")",
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silence_samples, GST_TIME_ARGS (silence_time),
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GST_TIME_ARGS (self->max_silence_time));
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} else {
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GST_DEBUG_OBJECT (self,
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"Inserting %" G_GUINT64_FORMAT " samples of silence (%"
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GST_TIME_FORMAT ")", silence_samples,
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GST_TIME_ARGS (gst_util_uint64_scale (silence_samples, GST_SECOND,
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rate)));
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GST_TIME_ARGS (silence_time));
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/* Insert silence buffers to fill the gap in 1s chunks */
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while (silence_samples > 0) {
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@ -517,41 +545,47 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
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silence_samples -= n_samples;
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}
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} else if (new_offset < self->current_offset + avail_samples) {
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guint64 drop_samples =
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self->current_offset + avail_samples - new_offset;
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GST_DEBUG_OBJECT (self,
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"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
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drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
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GST_SECOND, rate)));
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self->drop_samples = drop_samples;
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discont = FALSE;
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}
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}
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} else {
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GST_DEBUG_OBJECT (self,
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"Got discont: Current timestamp %" GST_TIME_FORMAT
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", current end timestamp %" GST_TIME_FORMAT
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", timestamp after discont %" GST_TIME_FORMAT,
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GST_TIME_ARGS (current_timestamp),
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GST_TIME_ARGS (current_timestamp_end),
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GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
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} else if (new_offset < self->current_offset + avail_samples) {
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guint64 drop_samples =
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self->current_offset + avail_samples - new_offset;
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if (self->strict_buffer_size) {
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gst_adapter_clear (self->adapter);
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ret = GST_FLOW_OK;
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} else {
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ret =
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gst_audio_buffer_split_output (self, TRUE, rate, bpf,
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samples_per_buffer);
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GST_DEBUG_OBJECT (self,
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"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
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drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
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GST_SECOND, rate)));
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self->drop_samples = drop_samples;
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discont = FALSE;
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}
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self->current_offset = 0;
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self->accumulated_error = 0;
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self->resync_time = GST_BUFFER_PTS (buffer);
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}
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}
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if (discont) {
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/* We might end up in here also in gapless mode, if the above code decided
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* that no silence is to be inserted, because e.g. the gap is too big */
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GST_DEBUG_OBJECT (self,
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"Got discont: Current timestamp %" GST_TIME_FORMAT
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", current end timestamp %" GST_TIME_FORMAT
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", timestamp after discont %" GST_TIME_FORMAT,
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GST_TIME_ARGS (current_timestamp),
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GST_TIME_ARGS (current_timestamp_end),
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GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
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if (self->strict_buffer_size) {
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gst_adapter_clear (self->adapter);
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ret = GST_FLOW_OK;
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} else {
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ret =
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gst_audio_buffer_split_output (self, TRUE, rate, bpf,
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samples_per_buffer);
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}
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self->current_offset = 0;
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self->accumulated_error = 0;
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self->resync_time = GST_BUFFER_PTS (buffer);
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}
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return ret;
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}
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@ -63,6 +63,7 @@ struct _GstAudioBufferSplit {
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gboolean strict_buffer_size;
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gboolean gapless;
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GstClockTime max_silence_time;
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};
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struct _GstAudioBufferSplitClass {
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