mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
audiortppay: handle gaps
Add various conversion functions between time<->bytes<->rtptime that will be used later on. Refactor the min/max packet length code so that it can be used for both sample/frame based payloaders. Cache the returned values. code cleanups. When we discover a DISCONT buffer, make the outgoing RTP timestamps have the same gap as the GStreamer timestamps gap.
This commit is contained in:
Wim Taymans
parent
3a3c6f309c
commit
c1db9ebb20
1 changed files with 240 additions and 178 deletions
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@ -70,26 +70,36 @@
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GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
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#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
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/* function to calculate the min/max length and alignment of a packet */
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typedef gboolean (*GetLengthsFunc) (GstBaseRTPPayload * basepayload,
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guint * min_payload_len, guint * max_payload_len, guint * align);
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/* function to convert bytes to a duration */
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typedef GstClockTime (*GetDurationFunc) (GstBaseRTPAudioPayload * payload,
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/* function to convert bytes to a time */
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typedef GstClockTime (*GetBytesToTimeFunc) (GstBaseRTPAudioPayload * payload,
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guint64 bytes);
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/* function to convert bytes to RTP timestamp */
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typedef guint32 (*GetRTPTimeFunc) (GstBaseRTPAudioPayload * payload,
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/* function to convert bytes to a RTP time */
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typedef guint32 (*GetBytesToRTPTimeFunc) (GstBaseRTPAudioPayload * payload,
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guint64 bytes);
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/* function to convert time to bytes */
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typedef guint64 (*GetTimeToBytesFunc) (GstBaseRTPAudioPayload * payload,
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GstClockTime time);
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struct _GstBaseRTPAudioPayloadPrivate
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{
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GetLengthsFunc get_lengths;
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GetDurationFunc get_duration;
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GetRTPTimeFunc get_rtptime;
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GetBytesToTimeFunc bytes_to_time;
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GetBytesToRTPTimeFunc bytes_to_rtptime;
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GetTimeToBytesFunc time_to_bytes;
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GstAdapter *adapter;
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guint fragment_size;
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GstClockTime frame_duration_ns;
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gboolean discont;
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guint64 offset;
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GstClockTime last_timestamp;
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guint32 last_rtptime;
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guint align;
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guint cached_mtu;
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guint cached_min_ptime;
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guint cached_max_ptime;
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guint cached_min_length;
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guint cached_max_length;
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};
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@ -99,29 +109,29 @@ struct _GstBaseRTPAudioPayloadPrivate
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static void gst_base_rtp_audio_payload_finalize (GObject * object);
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/* length functions */
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static gboolean gst_base_rtp_audio_payload_get_frame_lengths (GstBaseRTPPayload
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* basepayload, guint * min_payload_len, guint * max_payload_len,
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guint * align);
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static gboolean gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload
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* basepayload, guint * min_payload_len, guint * max_payload_len,
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guint * align);
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/* duration functions */
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/* bytes to time functions */
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static GstClockTime
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gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload * payload,
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guint64 bytes);
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gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload *
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payload, guint64 bytes);
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static GstClockTime
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gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
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gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload *
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payload, guint64 bytes);
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/* rtptime functions */
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/* bytes to RTP time functions */
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static guint32
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gst_base_rtp_audio_payload_get_frame_rtptime (GstBaseRTPAudioPayload * payload,
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guint64 bytes);
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static guint32
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gst_base_rtp_audio_payload_get_sample_rtptime (GstBaseRTPAudioPayload *
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gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload *
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payload, guint64 bytes);
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static guint32
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gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload *
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payload, guint64 bytes);
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/* time to bytes functions */
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static guint64
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gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload *
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payload, GstClockTime time);
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static guint64
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gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload *
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payload, GstClockTime time);
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static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
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* payload, GstBuffer * buffer);
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@ -169,30 +179,29 @@ gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass)
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}
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static void
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gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
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gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * payload,
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GstBaseRTPAudioPayloadClass * klass)
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{
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basertpaudiopayload->priv =
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GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload);
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payload->priv = GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (payload);
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/* these need to be set by child object if frame based */
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basertpaudiopayload->frame_size = 0;
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basertpaudiopayload->frame_duration = 0;
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payload->frame_size = 0;
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payload->frame_duration = 0;
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/* these need to be set by child object if sample based */
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basertpaudiopayload->sample_size = 0;
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payload->sample_size = 0;
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basertpaudiopayload->priv->adapter = gst_adapter_new ();
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payload->priv->adapter = gst_adapter_new ();
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}
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static void
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gst_base_rtp_audio_payload_finalize (GObject * object)
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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GstBaseRTPAudioPayload *payload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
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payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
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g_object_unref (basertpaudiopayload->priv->adapter);
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g_object_unref (payload->priv->adapter);
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GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
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}
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@ -209,15 +218,16 @@ gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
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basertpaudiopayload)
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{
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g_return_if_fail (basertpaudiopayload != NULL);
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g_return_if_fail (basertpaudiopayload->priv->get_lengths == NULL);
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g_return_if_fail (basertpaudiopayload->priv->get_duration == NULL);
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g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL);
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g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL);
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g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL);
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basertpaudiopayload->priv->get_lengths =
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gst_base_rtp_audio_payload_get_frame_lengths;
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basertpaudiopayload->priv->get_duration =
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gst_base_rtp_audio_payload_get_frame_duration;
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basertpaudiopayload->priv->get_rtptime =
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gst_base_rtp_audio_payload_get_frame_rtptime;
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basertpaudiopayload->priv->bytes_to_time =
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gst_base_rtp_audio_payload_frame_bytes_to_time;
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basertpaudiopayload->priv->bytes_to_rtptime =
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gst_base_rtp_audio_payload_frame_bytes_to_rtptime;
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basertpaudiopayload->priv->time_to_bytes =
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gst_base_rtp_audio_payload_frame_time_to_bytes;
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}
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/**
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@ -232,15 +242,16 @@ gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *
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basertpaudiopayload)
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{
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g_return_if_fail (basertpaudiopayload != NULL);
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g_return_if_fail (basertpaudiopayload->priv->get_lengths == NULL);
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g_return_if_fail (basertpaudiopayload->priv->get_duration == NULL);
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g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL);
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g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL);
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g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL);
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basertpaudiopayload->priv->get_lengths =
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gst_base_rtp_audio_payload_get_sample_lengths;
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basertpaudiopayload->priv->get_duration =
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gst_base_rtp_audio_payload_get_sample_duration;
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basertpaudiopayload->priv->get_rtptime =
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gst_base_rtp_audio_payload_get_sample_rtptime;
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basertpaudiopayload->priv->bytes_to_time =
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gst_base_rtp_audio_payload_sample_bytes_to_time;
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basertpaudiopayload->priv->bytes_to_rtptime =
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gst_base_rtp_audio_payload_sample_bytes_to_rtptime;
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basertpaudiopayload->priv->time_to_bytes =
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gst_base_rtp_audio_payload_sample_time_to_bytes;
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}
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/**
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@ -256,12 +267,18 @@ void
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gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
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* basertpaudiopayload, gint frame_duration, gint frame_size)
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{
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GstBaseRTPAudioPayloadPrivate *priv;
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g_return_if_fail (basertpaudiopayload != NULL);
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basertpaudiopayload->frame_duration = frame_duration;
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basertpaudiopayload->frame_size = frame_size;
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priv = basertpaudiopayload->priv;
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gst_adapter_clear (basertpaudiopayload->priv->adapter);
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basertpaudiopayload->frame_duration = frame_duration;
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priv->frame_duration_ns = frame_duration * GST_MSECOND;
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basertpaudiopayload->frame_size = frame_size;
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priv->align = frame_size;
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gst_adapter_clear (priv->adapter);
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GST_DEBUG_OBJECT (basertpaudiopayload, "frame set to %d ms and size %d",
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frame_duration, frame_size);
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@ -299,18 +316,22 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
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* basertpaudiopayload, gint sample_size)
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{
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guint fragment_size;
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GstBaseRTPAudioPayloadPrivate *priv;
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g_return_if_fail (basertpaudiopayload != NULL);
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priv = basertpaudiopayload->priv;
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basertpaudiopayload->sample_size = sample_size;
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/* sample_size is in bits and is converted into multiple bytes */
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fragment_size = sample_size;
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while ((fragment_size % 8) != 0)
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fragment_size += fragment_size;
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basertpaudiopayload->priv->fragment_size = fragment_size / 8;
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priv->fragment_size = fragment_size / 8;
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priv->align = priv->fragment_size;
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gst_adapter_clear (basertpaudiopayload->priv->adapter);
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gst_adapter_clear (priv->adapter);
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GST_DEBUG_OBJECT (basertpaudiopayload,
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"Samplebits set to sample size %d bits", sample_size);
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@ -321,25 +342,31 @@ gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload,
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GstBuffer * buffer, guint payload_len, GstClockTime timestamp)
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{
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GstBaseRTPPayload *basepayload;
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GstBaseRTPAudioPayloadPrivate *priv;
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basepayload = GST_BASE_RTP_PAYLOAD_CAST (payload);
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priv = payload->priv;
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/* set payload type */
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gst_rtp_buffer_set_payload_type (buffer, basepayload->pt);
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/* set marker bit for disconts */
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if (payload->priv->discont) {
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if (priv->discont) {
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GST_DEBUG_OBJECT (payload, "Setting marker and DISCONT");
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gst_rtp_buffer_set_marker (buffer, TRUE);
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GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
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payload->priv->discont = FALSE;
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priv->discont = FALSE;
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}
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GST_BUFFER_TIMESTAMP (buffer) = timestamp;
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/* get the offset in bytes */
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GST_BUFFER_OFFSET (buffer) =
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payload->priv->get_rtptime (payload, payload->priv->offset);
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/* get the offset in RTP time */
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GST_BUFFER_OFFSET (buffer) = priv->bytes_to_rtptime (payload, priv->offset);
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payload->priv->offset += payload_len;
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priv->offset += payload_len;
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/* remember the last rtptime/timestamp pair. We will use this to realign our
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* RTP timestamp after a buffer discont */
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priv->last_rtptime = GST_BUFFER_OFFSET (buffer);
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priv->last_timestamp = timestamp;
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}
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/**
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@ -374,14 +401,14 @@ gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* set metadata */
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gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
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timestamp);
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/* copy payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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memcpy (payload, data, payload_len);
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/* set metadata */
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gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
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timestamp);
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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@ -409,13 +436,15 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
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guint payload_len, GstClockTime timestamp)
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{
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GstBaseRTPPayload *basepayload;
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GstBaseRTPAudioPayloadPrivate *priv;
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GstBuffer *outbuf;
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guint8 *payload;
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GstFlowReturn ret;
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GstAdapter *adapter;
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guint64 distance;
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adapter = baseaudiopayload->priv->adapter;
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priv = baseaudiopayload->priv;
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adapter = priv->adapter;
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basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
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@ -437,8 +466,7 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
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if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
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/* convert the number of bytes since the last timestamp to time and add to
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* the last seen timestamp */
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timestamp +=
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baseaudiopayload->priv->get_duration (baseaudiopayload, distance);
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timestamp += priv->bytes_to_time (baseaudiopayload, distance);
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}
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}
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@ -448,14 +476,14 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* set metadata */
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gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
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timestamp);
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payload = gst_rtp_buffer_get_payload (outbuf);
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gst_adapter_copy (adapter, payload, 0, payload_len);
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gst_adapter_flush (adapter, payload_len);
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/* set metadata */
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gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
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timestamp);
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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@ -463,138 +491,142 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
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#define ALIGN_DOWN(val,len) ((val) - ((val) % (len)))
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/* this assumes all frames have a constant duration and a constant size */
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/* calculate the min and max length of a packet. This depends on the configured
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* mtu and min/max_ptime values. We cache those so that we don't have to redo
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* all the calculations */
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static gboolean
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gst_base_rtp_audio_payload_get_frame_lengths (GstBaseRTPPayload *
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gst_base_rtp_audio_payload_get_lengths (GstBaseRTPPayload *
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basepayload, guint * min_payload_len, guint * max_payload_len,
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guint * align)
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{
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GstBaseRTPAudioPayload *payload;
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guint frame_size;
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guint frame_duration;
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guint max_frames;
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GstBaseRTPAudioPayloadPrivate *priv;
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guint max_mtu, mtu;
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guint maxptime_octets;
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guint minptime_octets;
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payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
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priv = payload->priv;
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if (payload->frame_size == 0 || payload->frame_duration == 0)
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if (priv->align == 0)
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return FALSE;
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*align = frame_size = payload->frame_size;
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frame_duration = payload->frame_duration * GST_MSECOND;
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*align = priv->align;
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mtu = GST_BASE_RTP_PAYLOAD_MTU (payload);
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/* check cached values */
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if (G_LIKELY (priv->cached_mtu == mtu
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&& priv->cached_max_ptime == basepayload->max_ptime
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&& priv->cached_min_ptime == basepayload->min_ptime)) {
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/* if nothing changed, return cached values */
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*min_payload_len = priv->cached_min_length;
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*max_payload_len = priv->cached_max_length;
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return TRUE;
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}
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/* ptime max */
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if (basepayload->max_ptime != -1) {
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maxptime_octets =
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gst_util_uint64_scale (frame_size, basepayload->max_ptime,
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frame_duration);
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/* must be a multiple of the frame_size */
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maxptime_octets = MAX (frame_size, maxptime_octets);
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maxptime_octets = priv->time_to_bytes (payload, basepayload->max_ptime);
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} else {
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maxptime_octets = G_MAXUINT;
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}
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||||
/* MTU max */
|
||||
max_frames =
|
||||
gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (payload), 0,
|
||||
0);
|
||||
/* round down to frame_size */
|
||||
max_frames = ALIGN_DOWN (max_frames, frame_size);
|
||||
/* max payload length */
|
||||
*max_payload_len = MIN (max_frames, maxptime_octets);
|
||||
max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
|
||||
/* round down to alignment */
|
||||
max_mtu = ALIGN_DOWN (max_mtu, *align);
|
||||
|
||||
/* min number of bytes based on a given ptime, has to be a multiple
|
||||
of frame duration */
|
||||
minptime_octets =
|
||||
gst_util_uint64_scale (frame_size, basepayload->min_ptime,
|
||||
frame_duration);
|
||||
*min_payload_len = MAX (minptime_octets, frame_size);
|
||||
|
||||
if (*min_payload_len > *max_payload_len)
|
||||
*min_payload_len = *max_payload_len;
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static GstClockTime
|
||||
gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
|
||||
payload, guint64 bytes)
|
||||
{
|
||||
return (bytes / payload->frame_size) * (payload->frame_duration *
|
||||
GST_MSECOND);
|
||||
}
|
||||
|
||||
static guint32
|
||||
gst_base_rtp_audio_payload_get_frame_rtptime (GstBaseRTPAudioPayload * payload,
|
||||
guint64 bytes)
|
||||
{
|
||||
GstClockTime duration;
|
||||
|
||||
duration =
|
||||
(bytes / payload->frame_size) * (payload->frame_duration * GST_MSECOND);
|
||||
|
||||
return gst_util_uint64_scale_int (duration,
|
||||
GST_BASE_RTP_PAYLOAD_CAST (payload)->clock_rate, GST_SECOND);
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload *
|
||||
basepayload, guint * min_payload_len, guint * max_payload_len,
|
||||
guint * align)
|
||||
{
|
||||
GstBaseRTPAudioPayload *payload;
|
||||
guint maxptime_octets;
|
||||
guint minptime_octets;
|
||||
|
||||
payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
|
||||
|
||||
if (payload->sample_size == 0)
|
||||
return FALSE;
|
||||
|
||||
/* sample_size is in bits and is converted into multiple bytes */
|
||||
*align = payload->priv->fragment_size;
|
||||
|
||||
/* max number of bytes based on given ptime */
|
||||
if (basepayload->max_ptime != -1) {
|
||||
maxptime_octets = gst_util_uint64_scale (basepayload->max_ptime * 8,
|
||||
basepayload->clock_rate, payload->sample_size * GST_SECOND);
|
||||
} else {
|
||||
maxptime_octets = G_MAXUINT;
|
||||
}
|
||||
|
||||
*max_payload_len = MIN (
|
||||
/* MTU max */
|
||||
gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
|
||||
(payload), 0, 0),
|
||||
/* ptime max */
|
||||
maxptime_octets);
|
||||
|
||||
/* min number of bytes based on a given ptime, has to be a multiple
|
||||
* of sample rate */
|
||||
minptime_octets = gst_util_uint64_scale (basepayload->min_ptime * 8,
|
||||
basepayload->clock_rate, payload->sample_size * GST_SECOND);
|
||||
/* combine max ptime and max payload length */
|
||||
*max_payload_len = MIN (max_mtu, maxptime_octets);
|
||||
|
||||
/* min number of bytes based on a given ptime */
|
||||
minptime_octets = priv->time_to_bytes (payload, basepayload->min_ptime);
|
||||
/* must be at least one frame size */
|
||||
*min_payload_len = MAX (minptime_octets, *align);
|
||||
|
||||
if (*min_payload_len > *max_payload_len)
|
||||
*min_payload_len = *max_payload_len;
|
||||
|
||||
/* cache values */
|
||||
priv->cached_mtu = mtu;
|
||||
priv->cached_min_ptime = basepayload->min_ptime;
|
||||
priv->cached_max_ptime = basepayload->max_ptime;
|
||||
priv->cached_min_length = *min_payload_len;
|
||||
priv->cached_max_length = *max_payload_len;
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
/* frame conversions functions */
|
||||
static GstClockTime
|
||||
gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
|
||||
gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload *
|
||||
payload, guint64 bytes)
|
||||
{
|
||||
return gst_util_uint64_scale (bytes * 8, GST_SECOND,
|
||||
GST_BASE_RTP_PAYLOAD (payload)->clock_rate * payload->sample_size);
|
||||
return (bytes / payload->frame_size) * (payload->priv->frame_duration_ns);
|
||||
}
|
||||
|
||||
static guint32
|
||||
gst_base_rtp_audio_payload_get_sample_rtptime (GstBaseRTPAudioPayload *
|
||||
gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload *
|
||||
payload, guint64 bytes)
|
||||
{
|
||||
return (bytes * 8) / payload->sample_size;
|
||||
guint64 time;
|
||||
|
||||
time = (bytes / payload->frame_size) * (payload->priv->frame_duration_ns);
|
||||
|
||||
return gst_util_uint64_scale_int (time,
|
||||
GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND);
|
||||
}
|
||||
|
||||
static guint64
|
||||
gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload *
|
||||
payload, GstClockTime time)
|
||||
{
|
||||
return gst_util_uint64_scale (time, payload->frame_size,
|
||||
payload->priv->frame_duration_ns);
|
||||
}
|
||||
|
||||
/* sample conversion functions */
|
||||
static GstClockTime
|
||||
gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload *
|
||||
payload, guint64 bytes)
|
||||
{
|
||||
guint64 rtptime;
|
||||
|
||||
/* avoid division when we can */
|
||||
if (G_LIKELY (payload->sample_size != 8))
|
||||
rtptime = gst_util_uint64_scale_int (bytes, 8, payload->sample_size);
|
||||
else
|
||||
rtptime = bytes;
|
||||
|
||||
return gst_util_uint64_scale_int (rtptime, GST_SECOND,
|
||||
GST_BASE_RTP_PAYLOAD (payload)->clock_rate);
|
||||
}
|
||||
|
||||
static guint32
|
||||
gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload *
|
||||
payload, guint64 bytes)
|
||||
{
|
||||
/* avoid division when we can */
|
||||
if (G_LIKELY (payload->sample_size != 8))
|
||||
return gst_util_uint64_scale_int (bytes, 8, payload->sample_size);
|
||||
else
|
||||
return bytes;
|
||||
}
|
||||
|
||||
static guint64
|
||||
gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload *
|
||||
payload, guint64 time)
|
||||
{
|
||||
guint64 samples;
|
||||
|
||||
samples = gst_util_uint64_scale_int (time,
|
||||
GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND);
|
||||
|
||||
/* avoid multiplication when we can */
|
||||
if (G_LIKELY (payload->sample_size != 8))
|
||||
return gst_util_uint64_scale_int (samples, payload->sample_size, 8);
|
||||
else
|
||||
return samples;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
|
|
@ -602,6 +634,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
|
|||
basepayload, GstBuffer * buffer)
|
||||
{
|
||||
GstBaseRTPAudioPayload *payload;
|
||||
GstBaseRTPAudioPayloadPrivate *priv;
|
||||
guint payload_len;
|
||||
GstFlowReturn ret;
|
||||
guint available;
|
||||
|
|
@ -614,19 +647,37 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
|
|||
ret = GST_FLOW_OK;
|
||||
|
||||
payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
|
||||
|
||||
if (payload->priv->get_lengths == NULL || payload->priv->get_duration == NULL)
|
||||
goto config_error;
|
||||
priv = payload->priv;
|
||||
|
||||
discont = GST_BUFFER_IS_DISCONT (buffer);
|
||||
if (discont) {
|
||||
GstClockTime timestamp;
|
||||
|
||||
GST_DEBUG_OBJECT (payload, "Got DISCONT");
|
||||
/* flush everything out of the adapter, mark DISCONT */
|
||||
ret = gst_base_rtp_audio_payload_flush (payload, -1, -1);
|
||||
payload->priv->discont = TRUE;
|
||||
priv->discont = TRUE;
|
||||
|
||||
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
||||
|
||||
/* get the distance between the timestamp gap and produce the same gap in
|
||||
* the RTP timestamps */
|
||||
if (priv->last_timestamp != -1 && timestamp != -1) {
|
||||
/* we had a last timestamp, compare it to the new timestamp and update the
|
||||
* offset counter for RTP timestamps. The effect is that we will produce
|
||||
* output buffers containing the same RTP timestamp gap as the gap
|
||||
* between the GST timestamps. */
|
||||
if (timestamp > priv->last_timestamp) {
|
||||
/* we're only going to apply a positive gap, otherwise we let the marker
|
||||
* bit do its thing. simply convert to bytes and add the the current
|
||||
* offset */
|
||||
priv->offset +=
|
||||
priv->time_to_bytes (payload, timestamp - priv->last_timestamp);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if (!payload->priv->get_lengths (basepayload, &min_payload_len,
|
||||
if (!gst_base_rtp_audio_payload_get_lengths (basepayload, &min_payload_len,
|
||||
&max_payload_len, &align))
|
||||
goto config_error;
|
||||
|
||||
|
|
@ -638,7 +689,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
|
|||
|
||||
/* shortcut, we don't need to use the adapter when the packet can be pushed
|
||||
* through directly. */
|
||||
available = gst_adapter_available (payload->priv->adapter);
|
||||
available = gst_adapter_available (priv->adapter);
|
||||
|
||||
GST_DEBUG_OBJECT (payload, "got buffer size %u, available %u",
|
||||
size, available);
|
||||
|
|
@ -652,13 +703,14 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
|
|||
gst_buffer_unref (buffer);
|
||||
} else {
|
||||
/* push the buffer in the adapter */
|
||||
gst_adapter_push (payload->priv->adapter, buffer);
|
||||
gst_adapter_push (priv->adapter, buffer);
|
||||
available += size;
|
||||
|
||||
GST_DEBUG_OBJECT (payload, "available now %u", available);
|
||||
|
||||
/* as long as we have full frames */
|
||||
while (available >= min_payload_len) {
|
||||
/* get multiple of alignment */
|
||||
payload_len = ALIGN_DOWN (available, align);
|
||||
payload_len = MIN (max_payload_len, payload_len);
|
||||
|
||||
|
|
@ -691,6 +743,16 @@ gst_base_rtp_payload_audio_change_state (GstElement * element,
|
|||
|
||||
basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
basertppayload->priv->cached_mtu = -1;
|
||||
basertppayload->priv->last_rtptime = -1;
|
||||
basertppayload->priv->last_timestamp = -1;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
|
|
|
|||
Loading…
Reference in a new issue