mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-17 11:45:25 +00:00
audiortppay: more refactoring
Unify the sample/frame buffer handling code by making the functions plugable.
This commit is contained in:
Wim Taymans
parent
fb5037f727
commit
299ab7be0e
1 changed files with 49 additions and 146 deletions
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@ -70,18 +70,19 @@
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GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
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#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
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typedef enum
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{
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AUDIO_CODEC_TYPE_NONE,
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AUDIO_CODEC_TYPE_FRAME_BASED,
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AUDIO_CODEC_TYPE_SAMPLE_BASED
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} AudioCodecType;
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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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guint bytes);
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struct _GstBaseRTPAudioPayloadPrivate
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{
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AudioCodecType type;
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GetLengthsFunc get_lengths;
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GetDurationFunc get_duration;
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GstAdapter *adapter;
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guint64 min_ptime;
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guint fragment_size;
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};
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@ -92,22 +93,30 @@ 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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static GstClockTime
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gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload * payload,
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guint bytes);
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static GstClockTime
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gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
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payload, guint bytes);
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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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static GstFlowReturn
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gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer);
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static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement
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* element, GstStateChange transition);
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static GstFlowReturn
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gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer);
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static GstStateChangeReturn
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gst_base_rtp_payload_audio_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean
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gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event);
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static gboolean gst_base_rtp_payload_audio_handle_event (GstPad * pad,
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GstEvent * event);
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GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload,
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GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);
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@ -152,10 +161,6 @@ gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
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basertpaudiopayload->priv =
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GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload);
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basertpaudiopayload->base_ts = 0;
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basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_NONE;
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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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@ -184,17 +189,19 @@ gst_base_rtp_audio_payload_finalize (GObject * object)
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*
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* Tells #GstBaseRTPAudioPayload that the child element is for a frame based
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* audio codec
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*
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*/
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void
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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->type == AUDIO_CODEC_TYPE_NONE);
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basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_BASED;
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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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}
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/**
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@ -203,17 +210,19 @@ gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
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*
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* Tells #GstBaseRTPAudioPayload that the child element is for a sample based
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* audio codec
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*
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*/
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void
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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->type == AUDIO_CODEC_TYPE_NONE);
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basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
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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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}
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/**
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@ -283,30 +292,6 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
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gst_adapter_clear (basertpaudiopayload->priv->adapter);
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}
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static GstFlowReturn
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gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstFlowReturn ret;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
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ret = GST_FLOW_ERROR;
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if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
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ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload,
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buffer);
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} else if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {
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ret = gst_base_rtp_audio_payload_handle_sample_based_buffer (basepayload,
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buffer);
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} else {
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GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set");
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}
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return ret;
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}
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/**
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* gst_base_rtp_audio_payload_flush:
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* @baseaudiopayload: a #GstBaseRTPPayload
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@ -352,6 +337,7 @@ 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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static gboolean
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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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@ -412,91 +398,6 @@ gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
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payload->frame_size);
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}
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/* this assumes all frames have a constant duration and a constant size */
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static GstFlowReturn
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gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer)
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{
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GstBaseRTPAudioPayload *payload;
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guint align;
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guint payload_len;
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GstFlowReturn ret;
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guint available;
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guint min_payload_len;
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guint max_payload_len;
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guint size;
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ret = GST_FLOW_OK;
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payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
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if (!gst_base_rtp_audio_payload_get_frame_lengths (basepayload,
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&min_payload_len, &max_payload_len, &align))
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goto config_error;
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GST_DEBUG_OBJECT (payload,
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"Calculated min_payload_len %u and max_payload_len %u",
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min_payload_len, max_payload_len);
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size = GST_BUFFER_SIZE (buffer);
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/* shortcut, we don't need to use the adapter when the packet can be pushed
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* through directly. */
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available = gst_adapter_available (payload->priv->adapter);
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if (available == 0 && (size >= min_payload_len && size <= max_payload_len)) {
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/* If buffer fits on an RTP packet, let's just push it through
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* this will check against max_ptime and max_mtu */
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GST_DEBUG_OBJECT (payload, "Fast packet push");
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ret = gst_base_rtp_audio_payload_push (payload,
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GST_BUFFER_DATA (buffer), size, GST_BUFFER_TIMESTAMP (buffer));
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gst_buffer_unref (buffer);
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} else {
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/* push the buffer in the adapter */
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gst_adapter_push (payload->priv->adapter, buffer);
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available += size;
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/* as long as we have full frames */
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while (available >= min_payload_len) {
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guint64 distance;
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GstClockTime timestamp;
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/* We send as much as we can */
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payload_len = ALIGN_DOWN (available, align);
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payload_len = MIN (max_payload_len, payload_len);
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/* calculate the timestamp */
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timestamp =
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gst_adapter_prev_timestamp (payload->priv->adapter, &distance);
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GST_LOG_OBJECT (payload,
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"last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
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GST_TIME_ARGS (timestamp), distance);
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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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gst_base_rtp_audio_payload_get_frame_duration (payload, distance);
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}
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/* and flush out the bytes from the adapter */
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ret = gst_base_rtp_audio_payload_flush (payload, payload_len, timestamp);
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available -= payload_len;
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}
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}
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return ret;
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/* ERRORS */
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config_error:
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{
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GST_DEBUG_OBJECT (payload, "Required options not set");
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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}
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static gboolean
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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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@ -551,7 +452,7 @@ gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
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}
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static GstFlowReturn
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gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer)
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{
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GstBaseRTPAudioPayload *payload;
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@ -567,8 +468,11 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
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if (!gst_base_rtp_audio_payload_get_sample_lengths (basepayload,
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&min_payload_len, &max_payload_len, &align))
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if (payload->priv->get_lengths == NULL || payload->priv->get_duration == NULL)
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goto config_error;
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if (!payload->priv->get_lengths (basepayload, &min_payload_len,
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&max_payload_len, &align))
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goto config_error;
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GST_DEBUG_OBJECT (payload,
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@ -617,8 +521,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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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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gst_base_rtp_audio_payload_get_sample_duration (payload, distance);
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timestamp += payload->priv->get_duration (payload, distance);
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}
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/* and flush out the bytes from the adapter */
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