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-rw-r--r--ChangeLog14
m---------common0
-rw-r--r--gst/rtp/gstrtph264depay.c1
-rw-r--r--gst/rtp/gstrtph264pay.c253
4 files changed, 167 insertions, 101 deletions
diff --git a/ChangeLog b/ChangeLog
index 13e22797..f8c5c239 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,17 @@
+2008-05-14 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
+ Small comment added.
+
+ * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
+ (gst_rtp_h264_pay_decode_nal), (gst_rtp_h264_pay_parse_sps_pps),
+ (gst_rtp_h264_pay_payload_nal), (gst_rtp_h264_pay_handle_buffer):
+ Debug string cleanups (remove trailing \n)
+ Refactor and clean up the payloader a bit and make sure that we only
+ put one NAL unit in an RTP packet even if the input buffer contains
+ multiple NAL units.
+ Add suport for AVC format input.
+
2008-05-14 Peter Kjellerstedt <pkj@axis.com>
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
diff --git a/common b/common
-Subproject d78efae300d4dff0291717ccf7a2b82c713a81e
+Subproject 0823ac6b46b7332109bbc5f1ef40e24e648fae5
diff --git a/gst/rtp/gstrtph264depay.c b/gst/rtp/gstrtph264depay.c
index 2aeb88b4..9b798a34 100644
--- a/gst/rtp/gstrtph264depay.c
+++ b/gst/rtp/gstrtph264depay.c
@@ -367,6 +367,7 @@ gst_rtp_h264_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
case 26:
/* MTAP16 Multi-time aggregation packet 5.7.2 */
header_len = 5;
+ /* fallthrough, not implemented */
case 27:
/* MTAP24 Multi-time aggregation packet 5.7.2 */
header_len = 6;
diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
index e5b3532a..f638b174 100644
--- a/gst/rtp/gstrtph264pay.c
+++ b/gst/rtp/gstrtph264pay.c
@@ -121,14 +121,14 @@ gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass)
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_PROFILE_LEVEL_ID,
g_param_spec_string ("profile-level-id", "profile-level-id",
- "The profile-level-id to set in out caps (set to NULL to extract from stream)",
+ "The base64 profile-level-id to set in out caps (set to NULL to extract from stream)",
DEFAULT_PROFILE_LEVEL_ID,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
ARG_SPROP_PARAMETER_SETS, g_param_spec_string ("sprop-parameter-sets",
"sprop-parameter-sets",
- "The sprop-parameter-sets to set in out caps (set to NULL to extract from stream)",
+ "The base64 sprop-parameter-sets to set in out caps (set to NULL to extract from stream)",
DEFAULT_SPROP_PARAMETER_SETS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@@ -445,20 +445,20 @@ gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
* independently */
if (SPS_TYPE_ID == type) {
/* encode the entire SPS NAL in base64 */
- GST_DEBUG ("Found SPS %x %x %x Len=%u\n", (header >> 7),
+ GST_DEBUG ("Found SPS %x %x %x Len=%u", (header >> 7),
(header >> 5) & 3, type, len);
sps = data;
sps_len = len;
} else if (PPS_TYPE_ID == type) {
/* encoder the entire PPS NAL in base64 */
- GST_DEBUG ("Found PPS %x %x %x Len = %u\n",
+ GST_DEBUG ("Found PPS %x %x %x Len = %u",
(header >> 7), (header >> 5) & 3, type, len);
pps = data;
pps_len = len;
} else {
- GST_DEBUG ("NAL: %x %x %x Len = %u\n", (header >> 7),
+ GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7),
(header >> 5) & 3, type, len);
}
@@ -543,7 +543,7 @@ gst_rtp_h264_pay_parse_sps_pps (GstBaseRTPPayload * basepayload,
G_TYPE_STRING, profile,
"sprop-parameter-sets", G_TYPE_STRING, sprops, NULL);
- GST_DEBUG ("outcaps udpate: profile=%s, sps=%s, pps=%s\n",
+ GST_DEBUG ("outcaps udpate: profile=%s, sps=%s, pps=%s",
profile, GST_STR_NULL (sps), GST_STR_NULL (pps));
g_free (sprops);
@@ -554,95 +554,38 @@ gst_rtp_h264_pay_parse_sps_pps (GstBaseRTPPayload * basepayload,
}
static GstFlowReturn
-gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
- GstBuffer * buffer)
+gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload, guint8 * data,
+ guint size, GstClockTime timestamp)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
- guint size, idxdata;
- GstBuffer *outbuf;
- guint8 *payload, *data, *pdata;
guint8 nalType;
- GstClockTime timestamp;
guint packet_len, payload_len, mtu;
+ GstBuffer *outbuf;
+ guint8 *payload;
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
- size = GST_BUFFER_SIZE (buffer);
- data = GST_BUFFER_DATA (buffer);
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
-
- GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
+ nalType = data[0] & 0x1f;
+ GST_DEBUG_OBJECT (rtph264pay, "Processing Buffer with NAL TYPE=%d", nalType);
- /* we don't support AVC input yet */
- if (rtph264pay->packetized)
- goto not_supported;
-
- /* H264 stream analysis */
- pdata = data;
-
- /* use next_start_code() to scan buffer.
- * next_start_code() returns the offset in data,
- * starting from zero to the first byte of 0.0.0.1
- * If no start code is found, it returns the value of the
- * 'size' parameter.
- * pdata is unchanged by the call to next_start_code()
- */
- {
- guint offset = next_start_code (pdata, size);
-
- pdata += offset;
- idxdata = size - offset;
- }
-
- if (idxdata < 5) {
- GST_DEBUG_OBJECT (basepayload,
- "Returning GST_FLOW_OK without creating RTP packet");
- gst_buffer_unref (buffer);
- return GST_FLOW_OK;
- }
-
- pdata += 4;
- idxdata -= 4;
-
- if (rtph264pay->profile_level_id != NULL &&
- rtph264pay->sprop_parameter_sets != NULL) {
- if (rtph264pay->update_caps) {
- gst_basertppayload_set_outcaps (basepayload, "profile-level-id",
- G_TYPE_STRING, rtph264pay->profile_level_id, "sprop-parameter-sets",
- G_TYPE_STRING, rtph264pay->sprop_parameter_sets, NULL);
- rtph264pay->update_caps = FALSE;
-
- GST_DEBUG
- ("outcaps udpate: profile-level-id=%s, sprop-parameter-sets=%s\n",
- rtph264pay->profile_level_id, rtph264pay->sprop_parameter_sets);
- }
- } else {
- /* We know our stream is a valid H264 NAL packet,
- * go parse it for SPS/PPS to enrich the caps */
- gst_rtp_h264_pay_parse_sps_pps (basepayload, pdata, idxdata);
- }
-
- nalType = pdata[0] & 0x1f;
- GST_DEBUG_OBJECT (basepayload, "Processing Buffer with NAL TYPE=%d", nalType);
-
- packet_len = gst_rtp_buffer_calc_packet_len (idxdata, 0, 0);
+ packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
if (packet_len < mtu) {
GST_DEBUG_OBJECT (basepayload,
- "NAL Unit fit in one packet datasize=%d mtu=%d", idxdata, mtu);
+ "NAL Unit fit in one packet datasize=%d mtu=%d", size, mtu);
/* will fit in one packet */
- outbuf = gst_rtp_buffer_new_allocate (idxdata, 0, 0);
+ outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
+
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
gst_rtp_buffer_set_marker (outbuf, 1);
payload = gst_rtp_buffer_get_payload (outbuf);
- GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", idxdata);
- memcpy (payload, pdata, idxdata);
- gst_buffer_unref (buffer);
+ GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", size);
+ memcpy (payload, data, size);
+
ret = gst_basertppayload_push (basepayload, outbuf);
- return ret;
} else {
/* Fragmentation Units FU-A */
guint8 nalHeader;
@@ -650,22 +593,22 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
int ii = 0, start = 1, end = 0, first = 0;
GST_DEBUG_OBJECT (basepayload,
- "NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", idxdata, mtu);
+ "NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", size, mtu);
- nalHeader = *pdata;
- pdata++;
- idxdata--;
+ nalHeader = *data;
+ data++;
+ size--;
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
- idxdata);
+ size);
/* We keep 2 bytes for FU indicator and FU Header */
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
while (end == 0) {
- limitedSize = idxdata < payload_len ? idxdata : payload_len;
+ limitedSize = size < payload_len ? size : payload_len;
GST_DEBUG_OBJECT (basepayload,
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
ii);
@@ -674,9 +617,8 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
payload = gst_rtp_buffer_get_payload (outbuf);
- if (limitedSize == idxdata) {
- GST_DEBUG_OBJECT (basepayload, "end idxdata=%d iteration=%d", idxdata,
- ii);
+ if (limitedSize == size) {
+ GST_DEBUG_OBJECT (basepayload, "end size=%d iteration=%d", size, ii);
end = 1;
}
gst_rtp_buffer_set_marker (outbuf, end);
@@ -687,7 +629,7 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
/* FU Header */
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
- memcpy (&payload[2], pdata + first, limitedSize);
+ memcpy (&payload[2], data + first, limitedSize);
GST_DEBUG_OBJECT (basepayload,
"recorded %d payload bytes into packet iteration=%d", limitedSize + 2,
ii);
@@ -696,30 +638,139 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
if (ret != GST_FLOW_OK)
break;
- idxdata -= limitedSize;
+ size -= limitedSize;
first += limitedSize;
ii++;
start = 0;
}
-
- gst_buffer_unref (buffer);
- return ret;
}
+ return ret;
+}
- /* ERRORS */
- GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
- (NULL), ("Should not be there !!"));
- gst_buffer_unref (buffer);
+static GstFlowReturn
+gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpH264Pay *rtph264pay;
+ GstFlowReturn ret;
+ guint size, nal_len;
+ guint8 *data;
+ GstClockTime timestamp;
+
+ rtph264pay = GST_RTP_H264_PAY (basepayload);
- return GST_FLOW_ERROR;
+ /* the input buffer contains one or more NAL units */
+ size = GST_BUFFER_SIZE (buffer);
+ data = GST_BUFFER_DATA (buffer);
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ ret = GST_FLOW_OK;
-not_supported:
- {
- GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
- (NULL), ("AVC H264 is not supported yet"));
- gst_buffer_unref (buffer);
- return GST_FLOW_NOT_SUPPORTED;
+ GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
+
+ /* now loop over all NAL units and put them in a packet
+ * FIXME, we should really try to pack multiple NAL units into one RTP packet
+ * if we can, especially for the config packets that wont't cause decoder
+ * latency. */
+ if (rtph264pay->packetized) {
+ guint nal_length_size;
+
+ nal_length_size = rtph264pay->nal_length_size;
+
+ while (size > nal_length_size) {
+ gint i;
+
+ nal_len = 0;
+ for (i = 0; i < nal_length_size; i++) {
+ nal_len = ((nal_len << 8) + data[i]);
+ }
+
+ /* skip the length bytes, make sure we don't run past the buffer size */
+ data += nal_length_size;
+ size -= nal_length_size;
+
+ if (size >= nal_len) {
+ GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
+ } else {
+ nal_len = size;
+ GST_DEBUG_OBJECT (basepayload, "got incomplete NAL of size %u",
+ nal_len);
+ }
+
+ ret =
+ gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp);
+ if (ret != GST_FLOW_OK)
+ break;
+
+ data += nal_len;
+ size -= nal_len;
+ }
+ } else {
+ guint next;
+
+ /* get offset of first start code */
+ next = next_start_code (data, size);
+
+ /* skip to start code, if no start code is found, next will be size and we
+ * will not collect data. */
+ data += next;
+ size -= next;
+
+ GST_DEBUG_OBJECT (basepayload, "found first start at %u, bytes left %u",
+ next, size);
+
+ while (size > 4) {
+ /* skip start code */
+ data += 4;
+ size -= 4;
+
+ /* use next_start_code() to scan buffer.
+ * next_start_code() returns the offset in data,
+ * starting from zero to the first byte of 0.0.0.1
+ * If no start code is found, it returns the value of the
+ * 'size' parameter.
+ * data is unchanged by the call to next_start_code()
+ */
+ next = next_start_code (data, size);
+
+ /* nal length is distance to next start code */
+ nal_len = next;
+
+ GST_DEBUG_OBJECT (basepayload, "found next start at %u of size %u", next,
+ nal_len);
+
+ if (rtph264pay->profile_level_id != NULL &&
+ rtph264pay->sprop_parameter_sets != NULL) {
+ if (rtph264pay->update_caps) {
+ gst_basertppayload_set_outcaps (basepayload, "profile-level-id",
+ G_TYPE_STRING, rtph264pay->profile_level_id,
+ "sprop-parameter-sets", G_TYPE_STRING,
+ rtph264pay->sprop_parameter_sets, NULL);
+ rtph264pay->update_caps = FALSE;
+
+ GST_DEBUG
+ ("outcaps udpate: profile-level-id=%s, sprop-parameter-sets=%s",
+ rtph264pay->profile_level_id, rtph264pay->sprop_parameter_sets);
+ }
+ } else {
+ /* We know our stream is a valid H264 NAL packet,
+ * go parse it for SPS/PPS to enrich the caps */
+ gst_rtp_h264_pay_parse_sps_pps (basepayload, data, nal_len);
+ }
+
+ /* put the data in one or more RTP packets */
+ ret =
+ gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp);
+ if (ret != GST_FLOW_OK)
+ break;
+
+ /* move to next NAL packet */
+ data += nal_len;
+ size -= nal_len;
+ }
}
+ gst_buffer_unref (buffer);
+
+ return ret;
}
static GstStateChangeReturn
generated by cgit (git 2.34.1) at 2024-06-24 05:48:00 +0000