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 *
- basepayload, GstBuffer * buffer)
+#define ALIGN_DOWN(val,len) ((val) - ((val) % (len)))
+
+static gboolean
+gst_base_rtp_audio_payload_get_frame_lengths (GstBaseRTPPayload *
+ basepayload, guint * min_payload_len, guint * max_payload_len,
+ guint * align)
{
- GstBaseRTPAudioPayload *basertpaudiopayload;
- guint payload_len;
- GstFlowReturn ret;
- guint available;
- gint frame_size, frame_duration;
+ GstBaseRTPAudioPayload *payload;
+ guint frame_size;
+ guint frame_duration;
guint max_frames;
- guint maxptime_octets = G_MAXUINT;
- guint minptime_octets = 0;
- guint min_payload_len;
- guint max_payload_len;
- guint minptime_ms;
- guint size;
-
- ret = GST_FLOW_OK;
+ guint maxptime_octets;
+ guint minptime_octets;
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
- if (basertpaudiopayload->frame_size == 0 ||
- basertpaudiopayload->frame_duration == 0)
- goto config_error;
+ if (payload->frame_size == 0 || payload->frame_duration == 0)
+ return FALSE;
- frame_size = basertpaudiopayload->frame_size;
- frame_duration = basertpaudiopayload->frame_duration;
+ *align = frame_size = payload->frame_size;
+ frame_duration = payload->frame_duration * GST_MSECOND;
- /* max number of bytes based on given ptime, has to be multiple of
- * frame_duration */
if (basepayload->max_ptime != -1) {
- guint ptime_ms = basepayload->max_ptime / 1000000;
-
- maxptime_octets = frame_size * (int) (ptime_ms / frame_duration);
- if (maxptime_octets == 0) {
- GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than"
- " minimum %d ms, overwriting to minimum", ptime_ms, frame_duration);
- maxptime_octets = frame_size;
- }
+ maxptime_octets =
+ gst_util_uint64_scale (frame_size, basepayload->max_ptime,
+ frame_duration);
+ /* must be a multiple of the frame_size */
+ maxptime_octets = MAX (frame_size, maxptime_octets);
+ } else {
+ maxptime_octets = G_MAXUINT;
}
/* MTU max */
- max_frames = gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
- (basertpaudiopayload), 0, 0);
+ max_frames =
+ gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (payload), 0,
+ 0);
/* round down to frame_size */
- max_frames = (max_frames / frame_size) * frame_size;
+ max_frames = ALIGN_DOWN (max_frames, frame_size);
/* max payload length */
- max_payload_len = MIN (max_frames, maxptime_octets);
+ *max_payload_len = MIN (max_frames, maxptime_octets);
/* min number of bytes based on a given ptime, has to be a multiple
of frame duration */
- minptime_ms = basepayload->min_ptime / 1000000;
+ minptime_octets =
+ gst_util_uint64_scale (frame_size, basepayload->min_ptime,
+ frame_duration);
+ *min_payload_len = MAX (minptime_octets, frame_size);
- minptime_octets = frame_size * (int) (minptime_ms / frame_duration);
+ if (*min_payload_len > *max_payload_len)
+ *min_payload_len = *max_payload_len;
- min_payload_len = MAX (minptime_octets, frame_size);
+ return TRUE;
+}
+
+static GstClockTime
+gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
+ payload, guint bytes)
+{
+ return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND,
+ payload->frame_size);
+}
+
+/* this assumes all frames have a constant duration and a constant size */
+static GstFlowReturn
+gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
+ basepayload, GstBuffer * buffer)
+{
+ GstBaseRTPAudioPayload *payload;
+ guint align;
+ guint payload_len;
+ GstFlowReturn ret;
+ guint available;
+ guint min_payload_len;
+ guint max_payload_len;
+ guint size;
+
+ ret = GST_FLOW_OK;
- if (min_payload_len > max_payload_len)
- min_payload_len = max_payload_len;
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
+
+ if (!gst_base_rtp_audio_payload_get_frame_lengths (basepayload,
+ &min_payload_len, &max_payload_len, &align))
+ goto config_error;
- GST_DEBUG_OBJECT (basertpaudiopayload,
+ GST_DEBUG_OBJECT (payload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
/* shortcut, we don't need to use the adapter when the packet can be pushed
* through directly. */
- available = gst_adapter_available (basertpaudiopayload->priv->adapter);
+ available = gst_adapter_available (payload->priv->adapter);
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_DEBUG_OBJECT (payload, "Fast packet push");
+ ret = gst_base_rtp_audio_payload_push (payload,
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);
+ gst_adapter_push (payload->priv->adapter, buffer);
available += size;
/* as long as we have full frames */
GstClockTime timestamp;
/* We send as much as we can */
- payload_len =
- MIN (max_payload_len, (available / frame_size) * frame_size);
+ payload_len = ALIGN_DOWN (available, align);
+ payload_len = MIN (max_payload_len, payload_len);
/* calculate the timestamp */
timestamp =
- gst_adapter_prev_timestamp (basertpaudiopayload->priv->adapter,
- &distance);
+ gst_adapter_prev_timestamp (payload->priv->adapter, &distance);
- GST_LOG_OBJECT (basertpaudiopayload,
+ GST_LOG_OBJECT (payload,
"last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
GST_TIME_ARGS (timestamp), distance);
/* 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);
+ gst_base_rtp_audio_payload_get_frame_duration (payload, distance);
}
/* and flush out the bytes from the adapter */
- ret =
- gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
- timestamp);
+ ret = gst_base_rtp_audio_payload_flush (payload, payload_len, timestamp);
available -= payload_len;
}
/* ERRORS */
config_error:
{
- GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
+ GST_DEBUG_OBJECT (payload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
-static GstFlowReturn
-gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
- basepayload, GstBuffer * buffer)
+static gboolean
+gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload *
+ basepayload, guint * min_payload_len, guint * max_payload_len,
+ guint * align)
{
- GstBaseRTPAudioPayload *basertpaudiopayload;
- guint payload_len;
- GstFlowReturn ret;
- guint available;
- guint maxptime_octets = G_MAXUINT;
- guint minptime_octets = 0;
- guint min_payload_len;
- guint max_payload_len;
- guint fragment_size;
- guint64 datarate;
- guint size;
+ GstBaseRTPAudioPayload *payload;
+ guint maxptime_octets;
+ guint minptime_octets;
- ret = GST_FLOW_OK;
-
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
- if (basertpaudiopayload->sample_size == 0)
- goto config_error;
+ if (payload->sample_size == 0)
+ return FALSE;
/* sample_size is in bits and is converted into multiple bytes */
- fragment_size = basertpaudiopayload->priv->fragment_size;
- datarate = basertpaudiopayload->sample_size * basepayload->clock_rate;
+ *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, basertpaudiopayload->sample_size * GST_SECOND);
+ basepayload->clock_rate, payload->sample_size * GST_SECOND);
+ } else {
+ maxptime_octets = G_MAXUINT;
}
- max_payload_len = MIN (
+ *max_payload_len = MIN (
/* MTU max */
gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
- (basertpaudiopayload), 0, 0),
+ (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, basertpaudiopayload->sample_size * GST_SECOND);
+ basepayload->clock_rate, payload->sample_size * GST_SECOND);
+
+ *min_payload_len = MAX (minptime_octets, *align);
+
+ if (*min_payload_len > *max_payload_len)
+ *min_payload_len = *max_payload_len;
+
+ return TRUE;
+}
+
+static GstClockTime
+gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
+ payload, guint bytes)
+{
+ return gst_util_uint64_scale (bytes * 8 * GST_SECOND,
+ GST_BASE_RTP_PAYLOAD (payload)->clock_rate, payload->sample_size);
+}
+
+static GstFlowReturn
+gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
+ basepayload, GstBuffer * buffer)
+{
+ GstBaseRTPAudioPayload *payload;
+ guint payload_len;
+ GstFlowReturn ret;
+ guint available;
+ guint min_payload_len;
+ guint max_payload_len;
+ guint align;
+ guint size;
- min_payload_len = MAX (minptime_octets, fragment_size);
+ ret = GST_FLOW_OK;
+
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
- if (min_payload_len > max_payload_len)
- min_payload_len = max_payload_len;
+ if (!gst_base_rtp_audio_payload_get_sample_lengths (basepayload,
+ &min_payload_len, &max_payload_len, &align))
+ goto config_error;
- GST_DEBUG_OBJECT (basertpaudiopayload,
+ GST_DEBUG_OBJECT (payload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
/* shortcut, we don't need to use the adapter when the packet can be pushed
* through directly. */
- available = gst_adapter_available (basertpaudiopayload->priv->adapter);
+ available = gst_adapter_available (payload->priv->adapter);
- GST_DEBUG_OBJECT (basertpaudiopayload, "got buffer size %u, available %u",
+ GST_DEBUG_OBJECT (payload, "got buffer size %u, available %u",
size, available);
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_DEBUG_OBJECT (payload, "Fast packet push");
+ ret = gst_base_rtp_audio_payload_push (payload,
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);
+ gst_adapter_push (payload->priv->adapter, buffer);
available += size;
- GST_DEBUG_OBJECT (basertpaudiopayload, "available now %u", available);
+ GST_DEBUG_OBJECT (payload, "available now %u", available);
/* as long as we have full frames */
while (available >= min_payload_len) {
guint64 distance;
GstClockTime timestamp;
- payload_len =
- MIN (max_payload_len, (available / fragment_size) * fragment_size);
+ payload_len = ALIGN_DOWN (available, align);
+ payload_len = MIN (max_payload_len, payload_len);
/* calculate the timestamp */
timestamp =
- gst_adapter_prev_timestamp (basertpaudiopayload->priv->adapter,
- &distance);
+ gst_adapter_prev_timestamp (payload->priv->adapter, &distance);
- GST_LOG_OBJECT (basertpaudiopayload,
+ GST_LOG_OBJECT (payload,
"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 * 8, GST_SECOND, datarate);
+ timestamp +=
+ gst_base_rtp_audio_payload_get_sample_duration (payload, distance);
}
/* and flush out the bytes from the adapter */
- ret =
- gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
- timestamp);
+ ret = gst_base_rtp_audio_payload_flush (payload, payload_len, timestamp);
available -= payload_len;
- GST_DEBUG_OBJECT (basertpaudiopayload, "available after push %u",
- available);
+ GST_DEBUG_OBJECT (payload, "available after push %u", available);
}
}
return ret;
/* ERRORS */
config_error:
{
- GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
+ GST_DEBUG_OBJECT (payload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
static gboolean
gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
{
- GstBaseRTPAudioPayload *basertpaudiopayload;
+ GstBaseRTPAudioPayload *payload;
gboolean res = FALSE;
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad));
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* FIXME. push remaining bytes */
- gst_adapter_clear (basertpaudiopayload->priv->adapter);
+ gst_adapter_clear (payload->priv->adapter);
break;
case GST_EVENT_FLUSH_STOP:
- gst_adapter_clear (basertpaudiopayload->priv->adapter);
+ gst_adapter_clear (payload->priv->adapter);
break;
default:
break;
}
- gst_object_unref (basertpaudiopayload);
+ gst_object_unref (payload);
/* return FALSE to let parent handle the remainder of the event */
return res;
projects / platform / upstream / gstreamer.git / commitdiff