mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
audiortppayload: refactor and cleanup
Always use the adapter when we need to fragment the incomming buffer. Use more modern adapter functions to avoid malloc and memcpy. The overall result is that the code looks cleaner while it should be equally fast and in some case avoid a memcpy and malloc. Use the adapter timestamping functions for more precise timestamps in case of weird disconts. Cache some values instead of recalculating them. Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from the internal adapter. API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
This commit is contained in:
Wim Taymans
parent
432612b035
commit
1c6b71af03
3 changed files with 177 additions and 159 deletions
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@ -82,6 +82,7 @@ struct _GstBaseRTPAudioPayloadPrivate
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AudioCodecType type;
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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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@ -242,7 +243,6 @@ gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
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* @sample_size: Size per sample in bytes.
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*
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* Sets the options for sample based audio codecs.
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*
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*/
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void
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gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
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@ -251,9 +251,8 @@ gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
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g_return_if_fail (basertpaudiopayload != NULL);
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/* sample_size is in bits internally */
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basertpaudiopayload->sample_size = sample_size * 8;
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gst_adapter_clear (basertpaudiopayload->priv->adapter);
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gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
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sample_size * 8);
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}
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/**
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@ -269,10 +268,18 @@ void
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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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g_return_if_fail (basertpaudiopayload != NULL);
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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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gst_adapter_clear (basertpaudiopayload->priv->adapter);
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}
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@ -300,6 +307,49 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload,
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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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* @payload_len: length of payload
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* @timestamp: a #GstClockTime
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*
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* Create an RTP buffer and store @payload_len bytes of the adapter as the
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* payload. Set the timestamp on the new buffer to @timestamp before pushing
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* the buffer downstream.
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*
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* Returns: a #GstFlowReturn
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*
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* Since: 0.10.25
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*/
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GstFlowReturn
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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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GstBuffer *outbuf;
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guint8 *payload;
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GstFlowReturn ret;
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basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
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GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
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payload_len, GST_TIME_ARGS (timestamp));
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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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/* copy payload */
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gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt);
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payload = gst_rtp_buffer_get_payload (outbuf);
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gst_adapter_copy (baseaudiopayload->priv->adapter, payload, 0, payload_len);
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gst_adapter_flush (baseaudiopayload->priv->adapter, payload_len);
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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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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@ -307,7 +357,6 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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guint payload_len;
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const guint8 *data = NULL;
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GstFlowReturn ret;
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guint available;
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gint frame_size, frame_duration;
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@ -316,8 +365,8 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
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guint minptime_octets = 0;
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guint min_payload_len;
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guint max_payload_len;
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gboolean use_adapter = FALSE;
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guint minptime_ms;
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guint size;
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ret = GST_FLOW_OK;
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@ -366,69 +415,57 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
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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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if (gst_adapter_available (basertpaudiopayload->priv->adapter)) {
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/* If there is always data in the adapter, we have to use it */
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gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
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available = gst_adapter_available (basertpaudiopayload->priv->adapter);
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use_adapter = TRUE;
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} else {
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/* let's set the base timestamp */
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basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
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size = GST_BUFFER_SIZE (buffer);
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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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if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
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GST_BUFFER_SIZE (buffer) <= max_payload_len) {
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ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
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GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
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GST_BUFFER_TIMESTAMP (buffer));
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gst_buffer_unref (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 (basertpaudiopayload->priv->adapter);
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return ret;
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}
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available = GST_BUFFER_SIZE (buffer);
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data = (guint8 *) GST_BUFFER_DATA (buffer);
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}
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/* as long as we have full frames */
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while (available >= min_payload_len) {
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gfloat ts_inc;
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/* We send as much as we can */
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payload_len = MIN (max_payload_len, (available / frame_size) * frame_size);
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if (use_adapter) {
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data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
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}
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ret =
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gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
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basertpaudiopayload->base_ts);
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ts_inc = (payload_len * frame_duration) / frame_size;
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ts_inc = ts_inc * GST_MSECOND;
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basertpaudiopayload->base_ts += gst_gdouble_to_guint64 (ts_inc);
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if (use_adapter) {
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gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
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available = gst_adapter_available (basertpaudiopayload->priv->adapter);
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} else {
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available -= payload_len;
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data += payload_len;
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}
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}
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if (!use_adapter) {
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if (available != 0) {
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GstBuffer *buf;
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buf = gst_buffer_create_sub (buffer,
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GST_BUFFER_SIZE (buffer) - available, available);
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gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
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}
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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 (basertpaudiopayload, "Fast packet push");
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ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
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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 (basertpaudiopayload->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 =
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MIN (max_payload_len, (available / frame_size) * frame_size);
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/* calculate the timestamp */
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timestamp =
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gst_adapter_prev_timestamp (basertpaudiopayload->priv->adapter,
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&distance);
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GST_LOG_OBJECT (basertpaudiopayload,
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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_util_uint64_scale (distance, frame_duration * GST_MSECOND,
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frame_size);
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}
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/* and flush out the bytes from the adapter */
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ret =
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gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
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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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@ -447,16 +484,15 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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guint payload_len;
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const guint8 *data = NULL;
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GstFlowReturn ret;
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guint available;
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guint maxptime_octets = G_MAXUINT;
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guint minptime_octets = 0;
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guint min_payload_len;
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guint max_payload_len;
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gboolean use_adapter = FALSE;
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guint fragment_size;
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guint64 datarate;
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guint size;
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ret = GST_FLOW_OK;
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@ -466,10 +502,8 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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goto config_error;
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/* sample_size is in bits and is converted into multiple bytes */
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fragment_size = basertpaudiopayload->sample_size;
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while ((fragment_size % 8) != 0)
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fragment_size += fragment_size;
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fragment_size /= 8;
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fragment_size = basertpaudiopayload->priv->fragment_size;
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datarate = basertpaudiopayload->sample_size * basepayload->clock_rate;
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/* max number of bytes based on given ptime */
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if (basepayload->max_ptime != -1) {
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@ -498,74 +532,62 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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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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if (gst_adapter_available (basertpaudiopayload->priv->adapter)) {
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/* If there is always data in the adapter, we have to use it */
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gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
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available = gst_adapter_available (basertpaudiopayload->priv->adapter);
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use_adapter = TRUE;
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} else {
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/* let's set the base timestamp */
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basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
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size = GST_BUFFER_SIZE (buffer);
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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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if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
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GST_BUFFER_SIZE (buffer) <= max_payload_len) {
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ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
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GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
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GST_BUFFER_TIMESTAMP (buffer));
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gst_buffer_unref (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 (basertpaudiopayload->priv->adapter);
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return ret;
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}
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GST_DEBUG_OBJECT (basertpaudiopayload, "got buffer size %u, available %u",
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size, available);
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available = GST_BUFFER_SIZE (buffer);
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data = (guint8 *) GST_BUFFER_DATA (buffer);
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}
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while (available >= min_payload_len) {
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gfloat num, datarate;
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payload_len =
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MIN (max_payload_len, (available / fragment_size) * fragment_size);
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if (use_adapter) {
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data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
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}
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ret =
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gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
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basertpaudiopayload->base_ts);
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num = payload_len * 8;
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datarate = (basertpaudiopayload->sample_size * basepayload->clock_rate);
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basertpaudiopayload->base_ts +=
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/* payload_len (bits) * nsecs/sec / datarate (bits*sec) */
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gst_gdouble_to_guint64 (num / datarate * GST_SECOND);
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GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT,
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GST_TIME_ARGS (basertpaudiopayload->base_ts));
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if (use_adapter) {
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gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
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available = gst_adapter_available (basertpaudiopayload->priv->adapter);
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} else {
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available -= payload_len;
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data += payload_len;
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}
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}
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if (!use_adapter) {
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if (available != 0) {
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GstBuffer *buf;
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buf = gst_buffer_create_sub (buffer,
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GST_BUFFER_SIZE (buffer) - available, available);
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gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
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}
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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 (basertpaudiopayload, "Fast packet push");
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ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
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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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}
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} else {
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/* push the buffer in the adapter */
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gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
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available += size;
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GST_DEBUG_OBJECT (basertpaudiopayload, "available now %u", available);
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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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payload_len =
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MIN (max_payload_len, (available / fragment_size) * fragment_size);
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/* calculate the timestamp */
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timestamp =
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gst_adapter_prev_timestamp (basertpaudiopayload->priv->adapter,
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&distance);
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GST_LOG_OBJECT (basertpaudiopayload,
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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 += gst_util_uint64_scale (distance * 8, GST_SECOND, datarate);
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}
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/* and flush out the bytes from the adapter */
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ret =
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gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
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timestamp);
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available -= payload_len;
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GST_DEBUG_OBJECT (basertpaudiopayload, "available after push %u",
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available);
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}
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}
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return ret;
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/* ERRORS */
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@ -68,33 +68,28 @@ struct _GstBaseRTPAudioPayloadClass
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GType gst_base_rtp_audio_payload_get_type (void);
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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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/* configure frame based */
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void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
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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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void gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload *basertpaudiopayload,
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gint frame_duration, gint frame_size);
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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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/* configure sample based */
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void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
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void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload *basertpaudiopayload,
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gint sample_size);
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void gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload *basertpaudiopayload,
|
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gint sample_size);
|
||||
|
||||
void
|
||||
gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
|
||||
*basertpaudiopayload, gint sample_size);
|
||||
/* get the internal adapter */
|
||||
GstAdapter* gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload *basertpaudiopayload);
|
||||
|
||||
void
|
||||
gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
|
||||
*basertpaudiopayload, gint sample_size);
|
||||
|
||||
GstFlowReturn
|
||||
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
|
||||
const guint8 * data, guint payload_len, GstClockTime timestamp);
|
||||
|
||||
GstAdapter*
|
||||
gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload
|
||||
*basertpaudiopayload);
|
||||
/* push and flushing data */
|
||||
GstFlowReturn gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
|
||||
const guint8 * data, guint payload_len,
|
||||
GstClockTime timestamp);
|
||||
GstFlowReturn gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
|
||||
guint payload_len, GstClockTime timestamp);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
|
|
|
|||
|
|
@ -1,4 +1,5 @@
|
|||
EXPORTS
|
||||
gst_base_rtp_audio_payload_flush
|
||||
gst_base_rtp_audio_payload_get_adapter
|
||||
gst_base_rtp_audio_payload_get_type
|
||||
gst_base_rtp_audio_payload_push
|
||||
|
|
|
|||
Loading…
Reference in a new issue