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rtph264pay: Make it actually work after cleanups

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This commit is contained in:
Olivier Crête 2012-08-08 17:25:36 -07:00
parent 6f74b2afb7
commit 264bcf7d6f
2 changed files with 38 additions and 26 deletions
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35
gst/rtp/gstrtph264pay.c
View file

@ -432,7 +432,7 @@ gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
if (stream_format) {
if (g_str_equal (stream_format, "avc"))
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_AVC;
if (g_str_equal (stream_format, "bytestream"))
if (g_str_equal (stream_format, "byte-stream"))
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_BYTESTREAM;
}
@ -988,8 +988,8 @@ gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload,
/* insert payload memory block */
gst_buffer_append_memory (outbuf,
gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, (guint8 *) data + pos,
limitedSize, 0, limitedSize, NULL, NULL));
gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY,
(guint8 *) data + pos, limitedSize, 0, limitedSize, NULL, NULL));
/* add the buffer to the buffer list */
gst_buffer_list_add (list, outbuf);
@ -1026,9 +1026,12 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
/* the input buffer contains one or more NAL units */
avc = rtph264pay->stream_format = GST_H264_STREAM_FORMAT_AVC;
avc = rtph264pay->stream_format == GST_H264_STREAM_FORMAT_AVC;
if (avc) {
/* In AVC mode, there is no adapter, so nothign to flush */
if (buffer == NULL)
return GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
@ -1038,6 +1041,7 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
} else {
dts = gst_adapter_prev_dts (rtph264pay->adapter, NULL);
pts = gst_adapter_prev_pts (rtph264pay->adapter, NULL);
if (buffer)
gst_adapter_push (rtph264pay->adapter, buffer);
size = gst_adapter_available (rtph264pay->adapter);
data = gst_adapter_map (rtph264pay->adapter, size);
@ -1045,11 +1049,13 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
"got %" G_GSIZE_FORMAT " bytes (%" G_GSIZE_FORMAT ")", size,
gst_buffer_get_size (buffer));
if (buffer) {
if (!GST_CLOCK_TIME_IS_VALID (dts))
dts = GST_BUFFER_DTS (buffer);
if (!GST_CLOCK_TIME_IS_VALID (pts))
pts = GST_BUFFER_PTS (buffer);
}
}
ret = GST_FLOW_OK;
@ -1135,8 +1141,9 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
*/
next = next_start_code (data, size);
if (next == size) {
/* Didn't find the start of next NAL, handle it next time */
if (next == size && buffer != NULL) {
/* Didn't find the start of next NAL and it's not EOS,
* handle it next time */
break;
}
@ -1194,12 +1201,12 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
/* skip start code */
data += 3;
/* Trim the end unless we're the last NAL in the buffer.
/* Trim the end unless we're the last NAL in the stream.
* In case we're not at the end of the buffer we know the next block
* starts with 0x000001 so all the 0x00 bytes at the end of this one are
* trailing 0x0 that can be discarded */
size = nal_len;
if (i + 1 != nal_queue->len)
if (i + 1 != nal_queue->len || buffer != NULL)
for (; size > 1 && data[size - 1] == 0x0; size--)
/* skip */ ;
@ -1210,10 +1217,11 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
* actually payload the NAL so we can know if the current NAL is
* the last one of an access unit or not if we are in bytestream mode
*/
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU &&
if ((rtph264pay->alignment == GST_H264_ALIGNMENT_AU || buffer == NULL) &&
i == nal_queue->len - 1)
end_of_au = TRUE;
/* put the data in one or more RTP packets */
ret =
gst_rtp_h264_pay_payload_nal (basepayload, data, size, dts, pts,
@ -1224,7 +1232,6 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
/* move to next NAL packet */
data += nal_len;
size -= nal_len;
pushed += nal_len + 3;
}
g_array_set_size (nal_queue, 0);
@ -1271,6 +1278,14 @@ gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
rtph264pay->send_spspps = TRUE;
}
break;
case GST_EVENT_EOS:
{
/* call handle_buffer with NULL to flush last NAL from adapter
* in byte-stream mode
*/
gst_rtp_h264_pay_handle_buffer (payload, NULL);
break;
}
default:
break;
}

15
tests/check/elements/rtp-payloading.c
View file

@ -279,8 +279,7 @@ rtp_pipeline_run (rtp_pipeline * p)
}
/*
* Enables buffer lists. Sets the buffer-list property of the payloader
* and adds a chain_list_function to the depayloader.
* Enables buffer lists and adds a chain_list_function to the depayloader.
* @param p Pointer to the RTP pipeline.
*/
static void
@ -288,9 +287,6 @@ rtp_pipeline_enable_lists (rtp_pipeline * p, guint mtu_size)
{
GstPad *pad;
/* use buffer lists */
g_object_set (p->rtppay, "buffer-list", TRUE, NULL);
/* set mtu size if needed */
if (mtu_size) {
g_object_set (p->rtppay, "mtu", mtu_size, NULL);
@ -525,7 +521,7 @@ GST_END_TEST;
static const guint8 rtp_h264_list_lt_mtu_frame_data[] =
/* not packetized, next NAL starts with 0001 */
{ 0x00, 0x00, 0x00, 0x01, 0x00, 0x10, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00,
0xad, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0d, 0x00
};
@ -533,8 +529,8 @@ static int rtp_h264_list_lt_mtu_frame_data_size = 16;
static int rtp_h264_list_lt_mtu_frame_count = 2;
/* NAL = 4 bytes */
static int rtp_h264_list_lt_mtu_bytes_sent = 2 * (16 - 4);
/* NAL = 4 bytes + 12 bytes RTP header */
static int rtp_h264_list_lt_mtu_bytes_sent = 2 * (12 + 16 - 4);
static int rtp_h264_list_lt_mtu_mtu_size = 1024;
@ -564,7 +560,8 @@ static int rtp_h264_list_gt_mtu_frame_data_size = 64;
static int rtp_h264_list_gt_mtu_frame_count = 1;
/* NAL = 4 bytes. When data does not fit into 1 mtu, 1 byte will be skipped */
static int rtp_h264_list_gt_mtu_bytes_sent = 1 * (64 - 4) - 1;
/* Also 12 byte RTP header + 2 byte fragment header */
static int rtp_h264_list_gt_mtu_bytes_sent = 1 * (64 - 4) - 1 + (5 * 14);
static int rtp_h264_list_gt_mtu_mty_size = 28;