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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
gstaudioaggregator: expose output-buffer-duration-fraction
The code for this is mostly lifted from audiobuffersplit, it allows use cases such as keeping the buffers output by compositor on one branch and audiomixer on another perfectly aligned, by requiring the compositor to output a n/d frame rate, and setting output-buffer-duration to d/n on the audiomixer. The old output-buffer-duration property now simply maps to its fractional counterpart, the last set property wins.
This commit is contained in:
Mathieu Duponchelle
parent
d90ac65af1
commit
31ac4f4665
1 changed files with 178 additions and 34 deletions
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@ -210,8 +210,8 @@ gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad
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aaggcpad->priv->converter =
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gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE,
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in_info, out_info,
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aaggcpad->priv->converter_config ? gst_structure_copy (aaggcpad->
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priv->converter_config) : NULL);
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aaggcpad->priv->converter_config ? gst_structure_copy (aaggcpad->priv->
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converter_config) : NULL);
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}
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aaggcpad->priv->converter_config_changed = FALSE;
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@ -368,10 +368,17 @@ struct _GstAudioAggregatorPrivate
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/* All three properties are unprotected, can't be modified while streaming */
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/* Size in frames that is output per buffer */
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GstClockTime output_buffer_duration;
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GstClockTime alignment_threshold;
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GstClockTime discont_wait;
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gint output_buffer_duration_n;
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gint output_buffer_duration_d;
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guint samples_per_buffer;
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guint error_per_buffer;
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guint accumulated_error;
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guint current_blocksize;
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/* Protected by srcpad stream clock */
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/* Output buffer starting at offset containing blocksize frames (calculated
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* from output_buffer_duration) */
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@ -424,6 +431,8 @@ static GstCaps *gst_audio_aggregator_fixate_src_caps (GstAggregator * agg,
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#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
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#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
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#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
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#define DEFAULT_OUTPUT_BUFFER_DURATION_N (1)
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#define DEFAULT_OUTPUT_BUFFER_DURATION_D (100)
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enum
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{
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@ -431,6 +440,7 @@ enum
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PROP_OUTPUT_BUFFER_DURATION,
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PROP_ALIGNMENT_THRESHOLD,
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PROP_DISCONT_WAIT,
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PROP_OUTPUT_BUFFER_DURATION_FRACTION,
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};
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G_DEFINE_ABSTRACT_TYPE_WITH_PRIVATE (GstAudioAggregator, gst_audio_aggregator,
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@ -448,6 +458,79 @@ gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
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return klass->convert_buffer (aaggpad, in_info, out_info, buffer);
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}
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static void
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gst_audio_aggregator_translate_output_buffer_duration (GstAudioAggregator *
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aagg, GstClockTime duration)
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{
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gint gcd;
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aagg->priv->output_buffer_duration_n = duration;
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aagg->priv->output_buffer_duration_d = GST_SECOND;
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gcd = gst_util_greatest_common_divisor (aagg->priv->output_buffer_duration_n,
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aagg->priv->output_buffer_duration_d);
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if (gcd) {
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aagg->priv->output_buffer_duration_n /= gcd;
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aagg->priv->output_buffer_duration_d /= gcd;
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}
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}
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static gboolean
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gst_audio_aggregator_update_samples_per_buffer (GstAudioAggregator * aagg)
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{
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gboolean ret = TRUE;
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GstAudioAggregatorPad *srcpad =
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GST_AUDIO_AGGREGATOR_PAD (GST_AGGREGATOR_SRC_PAD (aagg));
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if (!srcpad->info.finfo
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|| GST_AUDIO_INFO_FORMAT (&srcpad->info) == GST_AUDIO_FORMAT_UNKNOWN) {
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ret = FALSE;
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goto out;
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}
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aagg->priv->samples_per_buffer =
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(((guint64) GST_AUDIO_INFO_RATE (&srcpad->info)) *
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aagg->priv->output_buffer_duration_n) /
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aagg->priv->output_buffer_duration_d;
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if (aagg->priv->samples_per_buffer == 0) {
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ret = FALSE;
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goto out;
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}
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aagg->priv->error_per_buffer =
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(((guint64) GST_AUDIO_INFO_RATE (&srcpad->info)) *
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aagg->priv->output_buffer_duration_n) %
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aagg->priv->output_buffer_duration_d;
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aagg->priv->accumulated_error = 0;
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GST_DEBUG_OBJECT (aagg, "Buffer duration: %u/%u",
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aagg->priv->output_buffer_duration_n,
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aagg->priv->output_buffer_duration_d);
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GST_DEBUG_OBJECT (aagg, "Samples per buffer: %u (error: %u/%u)",
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aagg->priv->samples_per_buffer, aagg->priv->error_per_buffer,
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aagg->priv->output_buffer_duration_d);
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out:
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return ret;
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}
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static void
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gst_audio_aggregator_recalculate_latency (GstAudioAggregator * aagg)
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{
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guint64 latency = gst_util_uint64_scale_int (GST_SECOND,
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aagg->priv->output_buffer_duration_n,
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aagg->priv->output_buffer_duration_d);
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gst_aggregator_set_latency (GST_AGGREGATOR (aagg), latency, latency);
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GST_OBJECT_LOCK (aagg);
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/* Force recalculating in aggregate */
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aagg->priv->samples_per_buffer = 0;
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GST_OBJECT_UNLOCK (aagg);
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}
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static void
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gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
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{
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@ -489,6 +572,16 @@ gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
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G_MAXUINT64, DEFAULT_OUTPUT_BUFFER_DURATION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_OUTPUT_BUFFER_DURATION_FRACTION,
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gst_param_spec_fraction ("output-buffer-duration-fraction",
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"Output buffer duration fraction",
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"Output block size in nanoseconds, expressed as a fraction", 1,
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G_MAXINT, G_MAXINT, 1, DEFAULT_OUTPUT_BUFFER_DURATION_N,
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DEFAULT_OUTPUT_BUFFER_DURATION_D,
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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_ALIGNMENT_THRESHOLD,
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g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
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"Timestamp alignment threshold in nanoseconds", 0,
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@ -510,14 +603,14 @@ gst_audio_aggregator_init (GstAudioAggregator * aagg)
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g_mutex_init (&aagg->priv->mutex);
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aagg->priv->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION;
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aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
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aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;
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aagg->current_caps = NULL;
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gst_audio_aggregator_translate_output_buffer_duration (aagg,
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DEFAULT_OUTPUT_BUFFER_DURATION);
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gst_audio_aggregator_recalculate_latency (aagg);
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gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
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aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration);
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aagg->current_caps = NULL;
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}
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static void
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@ -540,10 +633,10 @@ gst_audio_aggregator_set_property (GObject * object, guint prop_id,
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switch (prop_id) {
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case PROP_OUTPUT_BUFFER_DURATION:
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aagg->priv->output_buffer_duration = g_value_get_uint64 (value);
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gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
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aagg->priv->output_buffer_duration,
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aagg->priv->output_buffer_duration);
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gst_audio_aggregator_translate_output_buffer_duration (aagg,
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g_value_get_uint64 (value));
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g_object_notify (object, "output-buffer-duration-fraction");
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gst_audio_aggregator_recalculate_latency (aagg);
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break;
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case PROP_ALIGNMENT_THRESHOLD:
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aagg->priv->alignment_threshold = g_value_get_uint64 (value);
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@ -551,6 +644,14 @@ gst_audio_aggregator_set_property (GObject * object, guint prop_id,
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case PROP_DISCONT_WAIT:
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aagg->priv->discont_wait = g_value_get_uint64 (value);
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break;
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case PROP_OUTPUT_BUFFER_DURATION_FRACTION:
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aagg->priv->output_buffer_duration_n =
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gst_value_get_fraction_numerator (value);
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aagg->priv->output_buffer_duration_d =
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gst_value_get_fraction_denominator (value);
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g_object_notify (object, "output-buffer-duration");
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gst_audio_aggregator_recalculate_latency (aagg);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -565,7 +666,9 @@ gst_audio_aggregator_get_property (GObject * object, guint prop_id,
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switch (prop_id) {
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case PROP_OUTPUT_BUFFER_DURATION:
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g_value_set_uint64 (value, aagg->priv->output_buffer_duration);
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g_value_set_uint64 (value, gst_util_uint64_scale_int (GST_SECOND,
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aagg->priv->output_buffer_duration_n,
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aagg->priv->output_buffer_duration_d));
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break;
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case PROP_ALIGNMENT_THRESHOLD:
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g_value_set_uint64 (value, aagg->priv->alignment_threshold);
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@ -573,6 +676,10 @@ gst_audio_aggregator_get_property (GObject * object, guint prop_id,
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case PROP_DISCONT_WAIT:
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g_value_set_uint64 (value, aagg->priv->discont_wait);
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break;
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case PROP_OUTPUT_BUFFER_DURATION_FRACTION:
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gst_value_set_fraction (value, aagg->priv->output_buffer_duration_n,
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aagg->priv->output_buffer_duration_d);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -866,8 +973,8 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
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gst_audio_aggregator_update_converters (aagg, &info, &old_info);
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if (srcpad_klass->update_conversion_info)
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srcpad_klass->
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update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->srcpad));
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srcpad_klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->
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srcpad));
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if (aagg->priv->current_buffer) {
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GstBuffer *converted;
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@ -878,6 +985,9 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
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gst_buffer_unref (aagg->priv->current_buffer);
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aagg->priv->current_buffer = converted;
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}
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/* Force recalculating in aggregate */
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aagg->priv->samples_per_buffer = 0;
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}
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GST_OBJECT_UNLOCK (aagg);
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@ -1180,8 +1290,8 @@ gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
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switch (format) {
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case GST_FORMAT_TIME:
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gst_query_set_position (query, format,
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gst_segment_to_stream_time (&GST_AGGREGATOR_PAD (agg->
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srcpad)->segment, GST_FORMAT_TIME,
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gst_segment_to_stream_time (&GST_AGGREGATOR_PAD (agg->srcpad)->
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segment, GST_FORMAT_TIME,
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GST_AGGREGATOR_PAD (agg->srcpad)->segment.position));
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res = TRUE;
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break;
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@ -1247,6 +1357,7 @@ gst_audio_aggregator_reset (GstAudioAggregator * aagg)
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gst_audio_info_init (&GST_AUDIO_AGGREGATOR_PAD (agg->srcpad)->info);
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gst_caps_replace (&aagg->current_caps, NULL);
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gst_buffer_replace (&aagg->priv->current_buffer, NULL);
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aagg->priv->accumulated_error = 0;
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GST_OBJECT_UNLOCK (aagg);
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GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
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}
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@ -1280,6 +1391,7 @@ gst_audio_aggregator_flush (GstAggregator * agg)
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GST_OBJECT_LOCK (aagg);
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GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
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aagg->priv->offset = -1;
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aagg->priv->accumulated_error = 0;
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gst_buffer_replace (&aagg->priv->current_buffer, NULL);
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GST_OBJECT_UNLOCK (aagg);
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GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
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@ -1698,6 +1810,15 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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GST_AUDIO_AGGREGATOR_LOCK (aagg);
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GST_OBJECT_LOCK (agg);
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if (aagg->priv->samples_per_buffer == 0) {
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if (!gst_audio_aggregator_update_samples_per_buffer (aagg)) {
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GST_ERROR_OBJECT (aagg,
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"Failed to calculate the number of samples per buffer");
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GST_OBJECT_UNLOCK (agg);
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goto not_negotiated;
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}
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}
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/* Update position from the segment start/stop if needed */
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if (agg_segment->position == -1) {
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if (agg_segment->rate > 0.0)
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@ -1706,16 +1827,31 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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agg_segment->position = agg_segment->stop;
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}
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rate = GST_AUDIO_INFO_RATE (&srcpad->info);
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bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
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if (G_UNLIKELY (srcpad->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
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if (timeout) {
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GstClockTime output_buffer_duration;
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GST_DEBUG_OBJECT (aagg,
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"Got timeout before receiving any caps, don't output anything");
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blocksize = aagg->priv->samples_per_buffer;
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if (aagg->priv->error_per_buffer + aagg->priv->accumulated_error >=
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aagg->priv->output_buffer_duration_d)
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blocksize += 1;
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aagg->priv->accumulated_error =
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(aagg->priv->accumulated_error +
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aagg->priv->error_per_buffer) % aagg->priv->output_buffer_duration_d;
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output_buffer_duration =
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gst_util_uint64_scale (blocksize, GST_SECOND, rate);
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/* Advance position */
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if (agg_segment->rate > 0.0)
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agg_segment->position += aagg->priv->output_buffer_duration;
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else if (agg_segment->position > aagg->priv->output_buffer_duration)
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agg_segment->position -= aagg->priv->output_buffer_duration;
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agg_segment->position += output_buffer_duration;
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else if (agg_segment->position > output_buffer_duration)
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agg_segment->position -= output_buffer_duration;
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else
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agg_segment->position = 0;
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@ -1728,9 +1864,6 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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}
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}
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rate = GST_AUDIO_INFO_RATE (&srcpad->info);
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bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
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if (aagg->priv->offset == -1) {
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aagg->priv->offset =
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gst_util_uint64_scale (agg_segment->position - agg_segment->start, rate,
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@ -1739,9 +1872,29 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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aagg->priv->offset);
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}
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blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
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rate, GST_SECOND);
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blocksize = MAX (1, blocksize);
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if (aagg->priv->current_buffer == NULL) {
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blocksize = aagg->priv->samples_per_buffer;
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if (aagg->priv->error_per_buffer + aagg->priv->accumulated_error >=
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aagg->priv->output_buffer_duration_d)
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blocksize += 1;
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aagg->priv->current_blocksize = blocksize;
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aagg->priv->accumulated_error =
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(aagg->priv->accumulated_error +
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aagg->priv->error_per_buffer) % aagg->priv->output_buffer_duration_d;
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GST_OBJECT_UNLOCK (agg);
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aagg->priv->current_buffer =
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GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
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blocksize);
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/* Be careful, some things could have changed ? */
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GST_OBJECT_LOCK (agg);
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GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
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} else {
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blocksize = aagg->priv->current_blocksize;
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}
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/* FIXME: Reverse mixing does not work at all yet */
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if (agg_segment->rate > 0.0) {
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@ -1755,15 +1908,6 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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agg_segment->start + gst_util_uint64_scale (next_offset, GST_SECOND,
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rate);
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if (aagg->priv->current_buffer == NULL) {
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GST_OBJECT_UNLOCK (agg);
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aagg->priv->current_buffer =
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GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
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blocksize);
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/* Be careful, some things could have changed ? */
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GST_OBJECT_LOCK (agg);
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GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
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}
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outbuf = aagg->priv->current_buffer;
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GST_LOG_OBJECT (agg,
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