AudioCodecType type;
GstAdapter *adapter;
guint64 min_ptime;
+ guint fragment_size;
};
* @sample_size: Size per sample in bytes.
*
* Sets the options for sample based audio codecs.
- *
*/
void
gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
g_return_if_fail (basertpaudiopayload != NULL);
/* sample_size is in bits internally */
- basertpaudiopayload->sample_size = sample_size * 8;
-
- gst_adapter_clear (basertpaudiopayload->priv->adapter);
+ gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+ sample_size * 8);
}
/**
gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
* basertpaudiopayload, gint sample_size)
{
+ guint fragment_size;
+
g_return_if_fail (basertpaudiopayload != NULL);
basertpaudiopayload->sample_size = sample_size;
+ /* sample_size is in bits and is converted into multiple bytes */
+ fragment_size = sample_size;
+ while ((fragment_size % 8) != 0)
+ fragment_size += fragment_size;
+ basertpaudiopayload->priv->fragment_size = fragment_size / 8;
+
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
return ret;
}
+/**
+ * gst_base_rtp_audio_payload_flush:
+ * @baseaudiopayload: a #GstBaseRTPPayload
+ * @payload_len: length of payload
+ * @timestamp: a #GstClockTime
+ *
+ * Create an RTP buffer and store @payload_len bytes of the adapter as the
+ * payload. Set the timestamp on the new buffer to @timestamp before pushing
+ * the buffer downstream.
+ *
+ * Returns: a #GstFlowReturn
+ *
+ * Since: 0.10.25
+ */
+GstFlowReturn
+gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
+ guint payload_len, GstClockTime timestamp)
+{
+ GstBaseRTPPayload *basepayload;
+ GstBuffer *outbuf;
+ guint8 *payload;
+ GstFlowReturn ret;
+
+ basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
+
+ GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
+ payload_len, GST_TIME_ARGS (timestamp));
+
+ /* create buffer to hold the payload */
+ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+
+ /* copy payload */
+ gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt);
+ payload = gst_rtp_buffer_get_payload (outbuf);
+ gst_adapter_copy (baseaudiopayload->priv->adapter, payload, 0, payload_len);
+ gst_adapter_flush (baseaudiopayload->priv->adapter, payload_len);
+
+ GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ ret = gst_basertppayload_push (basepayload, outbuf);
+
+ return ret;
+}
+
/* this assumes all frames have a constant duration and a constant size */
static GstFlowReturn
gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
{
GstBaseRTPAudioPayload *basertpaudiopayload;
guint payload_len;
- const guint8 *data = NULL;
GstFlowReturn ret;
guint available;
gint frame_size, frame_duration;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
- gboolean use_adapter = FALSE;
guint minptime_ms;
+ guint size;
ret = GST_FLOW_OK;
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
- if (gst_adapter_available (basertpaudiopayload->priv->adapter)) {
- /* If there is always data in the adapter, we have to use it */
- gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
- available = gst_adapter_available (basertpaudiopayload->priv->adapter);
- use_adapter = TRUE;
- } else {
- /* let's set the base timestamp */
- basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
-
- /* If buffer fits on an RTP packet, let's just push it through */
- /* this will check against max_ptime and max_mtu */
- if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
- GST_BUFFER_SIZE (buffer) <= max_payload_len) {
- ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
- GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
- GST_BUFFER_TIMESTAMP (buffer));
- gst_buffer_unref (buffer);
-
- return ret;
- }
-
- available = GST_BUFFER_SIZE (buffer);
- data = (guint8 *) GST_BUFFER_DATA (buffer);
- }
-
- /* as long as we have full frames */
- while (available >= min_payload_len) {
- gfloat ts_inc;
+ size = GST_BUFFER_SIZE (buffer);
- /* We send as much as we can */
- payload_len = MIN (max_payload_len, (available / frame_size) * frame_size);
+ /* shortcut, we don't need to use the adapter when the packet can be pushed
+ * through directly. */
+ available = gst_adapter_available (basertpaudiopayload->priv->adapter);
- if (use_adapter) {
- data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
- }
-
- ret =
- gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
- basertpaudiopayload->base_ts);
-
- ts_inc = (payload_len * frame_duration) / frame_size;
-
- ts_inc = ts_inc * GST_MSECOND;
- basertpaudiopayload->base_ts += gst_gdouble_to_guint64 (ts_inc);
+ if (available == 0 && (size >= min_payload_len && size <= max_payload_len)) {
+ /* If buffer fits on an RTP packet, let's just push it through
+ * this will check against max_ptime and max_mtu */
+ GST_DEBUG_OBJECT (basertpaudiopayload, "Fast packet push");
+ ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+ GST_BUFFER_DATA (buffer), size, GST_BUFFER_TIMESTAMP (buffer));
+ gst_buffer_unref (buffer);
+ } else {
+ /* push the buffer in the adapter */
+ gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
+ available += size;
+
+ /* as long as we have full frames */
+ while (available >= min_payload_len) {
+ guint64 distance;
+ GstClockTime timestamp;
+
+ /* We send as much as we can */
+ payload_len =
+ MIN (max_payload_len, (available / frame_size) * frame_size);
+
+ /* calculate the timestamp */
+ timestamp =
+ gst_adapter_prev_timestamp (basertpaudiopayload->priv->adapter,
+ &distance);
+
+ GST_LOG_OBJECT (basertpaudiopayload,
+ "last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
+ GST_TIME_ARGS (timestamp), distance);
+
+ if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
+ /* convert the number of bytes since the last timestamp to time and add to
+ * the last seen timestamp */
+ timestamp +=
+ gst_util_uint64_scale (distance, frame_duration * GST_MSECOND,
+ frame_size);
+ }
+
+ /* and flush out the bytes from the adapter */
+ ret =
+ gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
+ timestamp);
- if (use_adapter) {
- gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
- available = gst_adapter_available (basertpaudiopayload->priv->adapter);
- } else {
available -= payload_len;
- data += payload_len;
}
}
-
- if (!use_adapter) {
- if (available != 0) {
- GstBuffer *buf;
-
- buf = gst_buffer_create_sub (buffer,
- GST_BUFFER_SIZE (buffer) - available, available);
- gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
- }
- gst_buffer_unref (buffer);
- }
return ret;
/* ERRORS */
{
GstBaseRTPAudioPayload *basertpaudiopayload;
guint payload_len;
- const guint8 *data = NULL;
GstFlowReturn ret;
guint available;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
- gboolean use_adapter = FALSE;
-
guint fragment_size;
+ guint64 datarate;
+ guint size;
ret = GST_FLOW_OK;
goto config_error;
/* sample_size is in bits and is converted into multiple bytes */
- fragment_size = basertpaudiopayload->sample_size;
- while ((fragment_size % 8) != 0)
- fragment_size += fragment_size;
- fragment_size /= 8;
+ fragment_size = basertpaudiopayload->priv->fragment_size;
+ datarate = basertpaudiopayload->sample_size * basepayload->clock_rate;
/* max number of bytes based on given ptime */
if (basepayload->max_ptime != -1) {
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
- if (gst_adapter_available (basertpaudiopayload->priv->adapter)) {
- /* If there is always data in the adapter, we have to use it */
- gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
- available = gst_adapter_available (basertpaudiopayload->priv->adapter);
- use_adapter = TRUE;
- } else {
- /* let's set the base timestamp */
- basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
-
- /* If buffer fits on an RTP packet, let's just push it through */
- /* this will check against max_ptime and max_mtu */
- if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
- GST_BUFFER_SIZE (buffer) <= max_payload_len) {
- ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
- GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
- GST_BUFFER_TIMESTAMP (buffer));
- gst_buffer_unref (buffer);
-
- return ret;
- }
+ size = GST_BUFFER_SIZE (buffer);
- available = GST_BUFFER_SIZE (buffer);
- data = (guint8 *) GST_BUFFER_DATA (buffer);
- }
+ /* shortcut, we don't need to use the adapter when the packet can be pushed
+ * through directly. */
+ available = gst_adapter_available (basertpaudiopayload->priv->adapter);
+
+ GST_DEBUG_OBJECT (basertpaudiopayload, "got buffer size %u, available %u",
+ size, available);
- while (available >= min_payload_len) {
- gfloat num, datarate;
+ if (available == 0 && (size >= min_payload_len && size <= max_payload_len)) {
+ /* If buffer fits on an RTP packet, let's just push it through
+ * this will check against max_ptime and max_mtu */
+ GST_DEBUG_OBJECT (basertpaudiopayload, "Fast packet push");
+ ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+ GST_BUFFER_DATA (buffer), size, GST_BUFFER_TIMESTAMP (buffer));
+ gst_buffer_unref (buffer);
+ } else {
+ /* push the buffer in the adapter */
+ gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
+ available += size;
- payload_len =
- MIN (max_payload_len, (available / fragment_size) * fragment_size);
+ GST_DEBUG_OBJECT (basertpaudiopayload, "available now %u", available);
- if (use_adapter) {
- data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
- }
+ /* as long as we have full frames */
+ while (available >= min_payload_len) {
+ guint64 distance;
+ GstClockTime timestamp;
- ret =
- gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
- basertpaudiopayload->base_ts);
+ payload_len =
+ MIN (max_payload_len, (available / fragment_size) * fragment_size);
- num = payload_len * 8;
- datarate = (basertpaudiopayload->sample_size * basepayload->clock_rate);
+ /* calculate the timestamp */
+ timestamp =
+ gst_adapter_prev_timestamp (basertpaudiopayload->priv->adapter,
+ &distance);
- basertpaudiopayload->base_ts +=
- /* payload_len (bits) * nsecs/sec / datarate (bits*sec) */
- gst_gdouble_to_guint64 (num / datarate * GST_SECOND);
- GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT,
- GST_TIME_ARGS (basertpaudiopayload->base_ts));
+ GST_LOG_OBJECT (basertpaudiopayload,
+ "last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
+ GST_TIME_ARGS (timestamp), distance);
- if (use_adapter) {
- gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
- available = gst_adapter_available (basertpaudiopayload->priv->adapter);
- } else {
- available -= payload_len;
- data += payload_len;
- }
- }
+ if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
+ /* convert the number of bytes since the last timestamp to time and add to
+ * the last seen timestamp */
+ timestamp += gst_util_uint64_scale (distance * 8, GST_SECOND, datarate);
+ }
- if (!use_adapter) {
- if (available != 0) {
- GstBuffer *buf;
+ /* and flush out the bytes from the adapter */
+ ret =
+ gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
+ timestamp);
- buf = gst_buffer_create_sub (buffer,
- GST_BUFFER_SIZE (buffer) - available, available);
- gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
+ available -= payload_len;
+ GST_DEBUG_OBJECT (basertpaudiopayload, "available after push %u",
+ available);
}
- gst_buffer_unref (buffer);
}
-
return ret;
/* ERRORS */
projects / platform / upstream / gstreamer.git / commitdiff