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| author | Thomas Vander Stichele <thomas@apestaart.org> | 2006-02-12 13:10:20 +0000 |
|---|---|---|
| committer | Thomas Vander Stichele <thomas@apestaart.org> | 2006-02-12 13:10:20 +0000 |
| commit | e2a22e497c78b6d196e91a429d2a10fb005daea5 (patch) | |
| tree | 24d350af29de717f83bbec674979590075068980 /gst/rtp/gstrtpamrpay.c | |
| parent | a28e6777bd4a856f8b197a99bf990e4e9ec5a924 (diff) | |
gst/rtp/gstrtpamrpay.c: set timestamps if no incoming timestamps set
Original commit message from CVS:
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
set timestamps if no incoming timestamps set
Diffstat (limited to 'gst/rtp/gstrtpamrpay.c')
| -rw-r--r-- | gst/rtp/gstrtpamrpay.c | 14 |
1 files changed, 11 insertions, 3 deletions
diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c index da9944e5..27588d8c 100644 --- a/gst/rtp/gstrtpamrpay.c +++ b/gst/rtp/gstrtpamrpay.c @@ -36,7 +36,7 @@ GST_DEBUG_CATEGORY (rtpamrpay_debug); static GstElementDetails gst_rtp_amrpay_details = { "RTP packet parser", "Codec/Payloader/Network", - "Payode AMR audio into RTP packets (RFC 3267)", + "Payload-encode AMR audio into RTP packets (RFC 3267)", "Wim Taymans <wim@fluendo.com>" }; @@ -230,8 +230,16 @@ gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload, /* now alloc output buffer */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); - /* copy timestamp */ - GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + /* copy timestamp, or fabricate one */ + if (timestamp != GST_CLOCK_TIME_NONE) + GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + else { + /* AMR (nb) and AMR-WB both have 20 ms per frame */ + /* FIXME: when we do more than one AMR frame per packet, fix this */ + gint count = basepayload->seqnum - basepayload->seqnum_base; + + GST_BUFFER_TIMESTAMP (outbuf) = count * 20 * GST_MSECOND; + } /* get payload, this is now writable */ payload = gst_rtp_buffer_get_payload (outbuf); |