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author | Vivia Nikolaidou <vivia@ahiru.eu> | 2019-02-21 15:16:37 +0000 |
---|---|---|
committer | Sebastian Dröge <slomo@coaxion.net> | 2019-02-21 15:16:37 +0000 |
commit | ce0be4d1acbfaee49cb00b85c078c7a90032e5c2 (patch) | |
tree | 9c8fa1b21dd1b5eecedcd1d74a48b63aba5ac84d /gst/audiobuffersplit/gstaudiobuffersplit.c | |
parent | 7c767f3fcd5a7b40d205bb4d588dad6c6275c729 (diff) | |
download | gstreamer-plugins-bad-ce0be4d1acbfaee49cb00b85c078c7a90032e5c2.tar.gz |
audiobuffersplit: Added max-silence-time property
Diffstat (limited to 'gst/audiobuffersplit/gstaudiobuffersplit.c')
-rw-r--r-- | gst/audiobuffersplit/gstaudiobuffersplit.c | 198 |
1 files changed, 116 insertions, 82 deletions
diff --git a/gst/audiobuffersplit/gstaudiobuffersplit.c b/gst/audiobuffersplit/gstaudiobuffersplit.c index c6a730ca7..8d7767a65 100644 --- a/gst/audiobuffersplit/gstaudiobuffersplit.c +++ b/gst/audiobuffersplit/gstaudiobuffersplit.c @@ -47,6 +47,7 @@ enum PROP_DISCONT_WAIT, PROP_STRICT_BUFFER_SIZE, PROP_GAPLESS, + PROP_MAX_SILENCE_TIME, LAST_PROP }; @@ -56,6 +57,7 @@ enum #define DEFAULT_DISCONT_WAIT (1 * GST_SECOND) #define DEFAULT_STRICT_BUFFER_SIZE (FALSE) #define DEFAULT_GAPLESS (FALSE) +#define DEFAULT_MAX_SILENCE_TIME (0) #define parent_class gst_audio_buffer_split_parent_class G_DEFINE_TYPE (GstAudioBufferSplit, gst_audio_buffer_split, GST_TYPE_ELEMENT); @@ -123,6 +125,15 @@ 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_MAX_SILENCE_TIME, + g_param_spec_uint64 ("max-silence-time", + "Maximum time of silence to insert", + "Do not insert silence in gapless mode if the gap exceeds this " + "period (in ns) (0 = disabled)", + 0, G_MAXUINT64, DEFAULT_MAX_SILENCE_TIME, + 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", @@ -253,6 +264,9 @@ gst_audio_buffer_split_set_property (GObject * object, guint property_id, case PROP_GAPLESS: self->gapless = g_value_get_boolean (value); break; + case PROP_MAX_SILENCE_TIME: + self->max_silence_time = g_value_get_uint64 (value); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; @@ -288,6 +302,9 @@ gst_audio_buffer_split_get_property (GObject * object, guint property_id, case PROP_GAPLESS: g_value_set_boolean (value, self->gapless); break; + case PROP_MAX_SILENCE_TIME: + g_value_set_uint64 (value, self->max_silence_time); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; @@ -420,6 +437,11 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self, { gboolean discont; GstFlowReturn 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; GST_OBJECT_LOCK (self); discont = @@ -430,71 +452,77 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self, NULL); GST_OBJECT_UNLOCK (self); - if (discont) { - guint avail = gst_adapter_available (self->adapter); - guint avail_samples = avail / bpf; - guint64 new_offset; - GstClockTime current_timestamp; - GstClockTime current_timestamp_end; + if (!discont) + return ret; - /* Reset */ - self->drop_samples = 0; + /* 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); + 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 { + 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); + } + + 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); + discont = FALSE; } else { - 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); - } + 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))); - 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 (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_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))); + discont = FALSE; + } 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); + GstClockTime silence_time = + gst_util_uint64_scale (silence_samples, GST_SECOND, rate); + + if (silence_time > self->max_silence_time) { 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); - + "Not inserting %" G_GUINT64_FORMAT " samples of silence (%" + GST_TIME_FORMAT " exceeds maximum %" GST_TIME_FORMAT ")", + silence_samples, GST_TIME_ARGS (silence_time), + GST_TIME_ARGS (self->max_silence_time)); + } else { 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))); + GST_TIME_ARGS (silence_time)); /* Insert silence buffers to fill the gap in 1s chunks */ while (silence_samples > 0) { @@ -517,39 +545,45 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self, 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; + discont = FALSE; } - } - } 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))); + } else if (new_offset < self->current_offset + avail_samples) { + guint64 drop_samples = + self->current_offset + avail_samples - new_offset; - 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); + 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; + discont = FALSE; } + } + } - self->current_offset = 0; - self->accumulated_error = 0; - self->resync_time = GST_BUFFER_PTS (buffer); + if (discont) { + /* We might end up in here also in gapless mode, if the above code decided + * that no silence is to be inserted, because e.g. the gap is too big */ + 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; |