[FFmpeg-cvslog] rtpdec: K&R formatting and spelling cosmetics
Martin Storsjö
git at videolan.org
Mon Dec 10 01:31:31 CET 2012
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Fri Dec 7 15:50:17 2012 +0200| [5d471b73d20616f5ac701ff62e5de49465cda264] | committer: Diego Biurrun
rtpdec: K&R formatting and spelling cosmetics
Signed-off-by: Diego Biurrun <diego at biurrun.de>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=5d471b73d20616f5ac701ff62e5de49465cda264
---
libavformat/rtpdec.c | 212 ++++++++++++++++++++++++++------------------------
1 file changed, 110 insertions(+), 102 deletions(-)
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index a305dd6..38ce7f6 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -25,49 +25,46 @@
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
-#include "url.h"
-
#include "network.h"
-
+#include "url.h"
#include "rtpdec.h"
#include "rtpdec_formats.h"
-//#define DEBUG
-
-/* TODO: - add RTCP statistics reporting (should be optional).
-
- - add support for h263/mpeg4 packetized output : IDEA: send a
- buffer to 'rtp_write_packet' contains all the packets for ONE
- frame. Each packet should have a four byte header containing
- the length in big endian format (same trick as
- 'ffio_open_dyn_packet_buf')
-*/
+/* TODO:
+ * - add RTCP statistics reporting (should be optional).
+ *
+ * - add support for H.263/MPEG-4 packetized output: IDEA: send a
+ * buffer to 'rtp_write_packet' contains all the packets for ONE
+ * frame. Each packet should have a four byte header containing
+ * the length in big-endian format (same trick as
+ * 'ffio_open_dyn_packet_buf').
+ */
static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
- .enc_name = "X-MP3-draft-00",
- .codec_type = AVMEDIA_TYPE_AUDIO,
- .codec_id = AV_CODEC_ID_MP3ADU,
+ .enc_name = "X-MP3-draft-00",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_MP3ADU,
};
static RTPDynamicProtocolHandler speex_dynamic_handler = {
- .enc_name = "speex",
- .codec_type = AVMEDIA_TYPE_AUDIO,
- .codec_id = AV_CODEC_ID_SPEEX,
+ .enc_name = "speex",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_SPEEX,
};
static RTPDynamicProtocolHandler opus_dynamic_handler = {
- .enc_name = "opus",
- .codec_type = AVMEDIA_TYPE_AUDIO,
- .codec_id = AV_CODEC_ID_OPUS,
+ .enc_name = "opus",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_OPUS,
};
/* statistics functions */
-static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
+static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler = NULL;
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
- handler->next= RTPFirstDynamicPayloadHandler;
- RTPFirstDynamicPayloadHandler= handler;
+ handler->next = RTPFirstDynamicPayloadHandler;
+ RTPFirstDynamicPayloadHandler = handler;
}
void av_register_rtp_dynamic_payload_handlers(void)
@@ -108,7 +105,7 @@ void av_register_rtp_dynamic_payload_handlers(void)
}
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
- enum AVMediaType codec_type)
+ enum AVMediaType codec_type)
{
RTPDynamicProtocolHandler *handler;
for (handler = RTPFirstDynamicPayloadHandler;
@@ -120,7 +117,7 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
}
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
- enum AVMediaType codec_type)
+ enum AVMediaType codec_type)
{
RTPDynamicProtocolHandler *handler;
for (handler = RTPFirstDynamicPayloadHandler;
@@ -131,7 +128,8 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
return NULL;
}
-static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
+static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
+ int len)
{
int payload_len;
while (len >= 4) {
@@ -140,11 +138,12 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
switch (buf[1]) {
case RTCP_SR:
if (payload_len < 20) {
- av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
+ av_log(NULL, AV_LOG_ERROR,
+ "Invalid length for RTCP SR packet\n");
return AVERROR_INVALIDDATA;
}
- s->last_rtcp_ntp_time = AV_RB64(buf + 8);
+ s->last_rtcp_ntp_time = AV_RB64(buf + 8);
s->last_rtcp_timestamp = AV_RB32(buf + 16);
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
@@ -164,7 +163,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
return -1;
}
-#define RTP_SEQ_MOD (1<<16)
+#define RTP_SEQ_MOD (1 << 16)
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
{
@@ -174,8 +173,9 @@ static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
}
/*
-* called whenever there is a large jump in sequence numbers, or when they get out of probation...
-*/
+ * Called whenever there is a large jump in sequence numbers,
+ * or when they get out of probation...
+ */
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
s->max_seq = seq;
@@ -189,9 +189,7 @@ static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
s->transit = 0;
}
-/*
-* returns 1 if we should handle this packet.
-*/
+/* Returns 1 if we should handle this packet. */
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
uint16_t udelta = seq - s->max_seq;
@@ -199,7 +197,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
const int MAX_MISORDER = 100;
const int MIN_SEQUENTIAL = 2;
- /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
+ /* source not valid until MIN_SEQUENTIAL packets with sequence
+ * seq. numbers have been received */
if (s->probation) {
if (seq == s->max_seq + 1) {
s->probation--;
@@ -211,7 +210,7 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
}
} else {
s->probation = MIN_SEQUENTIAL - 1;
- s->max_seq = seq;
+ s->max_seq = seq;
}
} else if (udelta < MAX_DROPOUT) {
// in order, with permissible gap
@@ -223,7 +222,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
// sequence made a large jump...
if (seq == s->bad_seq) {
- // two sequential packets-- assume that the other side restarted without telling us; just resync.
+ /* two sequential packets -- assume that the other side
+ * restarted without telling us; just resync. */
rtp_init_sequence(s, seq);
} else {
s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
@@ -256,7 +256,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
return -1;
/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
- /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
+ /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
s->octet_count += count;
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
@@ -277,15 +277,15 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
avio_wb32(pb, s->ssrc); // server SSRC
// some placeholders we should really fill...
// RFC 1889/p64
- extended_max = stats->cycles + stats->max_seq;
- expected = extended_max - stats->base_seq + 1;
- lost = expected - stats->received;
- lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
- expected_interval = expected - stats->expected_prior;
+ extended_max = stats->cycles + stats->max_seq;
+ expected = extended_max - stats->base_seq + 1;
+ lost = expected - stats->received;
+ lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
+ expected_interval = expected - stats->expected_prior;
stats->expected_prior = expected;
- received_interval = stats->received - stats->received_prior;
+ received_interval = stats->received - stats->received_prior;
stats->received_prior = stats->received;
- lost_interval = expected_interval - received_interval;
+ lost_interval = expected_interval - received_interval;
if (expected_interval == 0 || lost_interval <= 0)
fraction = 0;
else
@@ -301,7 +301,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
avio_wb32(pb, 0); /* last SR timestamp */
avio_wb32(pb, 0); /* delay since last SR */
} else {
- uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
+ uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
uint32_t delay_since_last = ntp_time - s->last_rtcp_ntp_time;
avio_wb32(pb, middle_32_bits); /* last SR timestamp */
@@ -318,23 +318,22 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
avio_w8(pb, len);
avio_write(pb, s->hostname, len);
// padding
- for (len = (6 + len) % 4; len % 4; len++) {
+ for (len = (6 + len) % 4; len % 4; len++)
avio_w8(pb, 0);
- }
avio_flush(pb);
len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
int av_unused result;
av_dlog(s->ic, "sending %d bytes of RR\n", len);
- result= ffurl_write(s->rtp_ctx, buf, len);
+ result = ffurl_write(s->rtp_ctx, buf, len);
av_dlog(s->ic, "result from ffurl_write: %d\n", result);
av_free(buf);
}
return 0;
}
-void ff_rtp_send_punch_packets(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext *rtp_handle)
{
AVIOContext *pb;
uint8_t *buf;
@@ -372,25 +371,26 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle)
av_free(buf);
}
-
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
- * MPEG2TS streams to indicate that they should be demuxed inside the
+ * MPEG2-TS streams to indicate that they should be demuxed inside the
* rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
*/
-RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
+ URLContext *rtpc, int payload_type,
+ int queue_size)
{
RTPDemuxContext *s;
s = av_mallocz(sizeof(RTPDemuxContext));
if (!s)
return NULL;
- s->payload_type = payload_type;
- s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
+ s->payload_type = payload_type;
+ s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
- s->ic = s1;
- s->st = st;
- s->queue_size = queue_size;
+ s->ic = s1;
+ s->st = st;
+ s->queue_size = queue_size;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
s->ts = ff_mpegts_parse_open(s->ic);
@@ -399,7 +399,7 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext
return NULL;
}
} else if (st) {
- switch(st->codec->codec_id) {
+ switch (st->codec->codec_id) {
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
case AV_CODEC_ID_MP2:
@@ -432,11 +432,12 @@ void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
RTPDynamicProtocolHandler *handler)
{
s->dynamic_protocol_context = ctx;
- s->parse_packet = handler->parse_packet;
+ s->parse_packet = handler->parse_packet;
}
/**
- * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
+ * This was the second switch in rtp_parse packet.
+ * Normalizes time, if required, sets stream_index, etc.
*/
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
@@ -452,7 +453,9 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
/* compute pts from timestamp with received ntp_time */
delta_timestamp = timestamp - s->last_rtcp_timestamp;
/* convert to the PTS timebase */
- addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
+ addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
+ s->st->time_base.den,
+ (uint64_t) s->st->time_base.num << 32);
pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
delta_timestamp;
return;
@@ -460,13 +463,15 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
if (!s->base_timestamp)
s->base_timestamp = timestamp;
- /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
+ /* assume that the difference is INT32_MIN < x < INT32_MAX,
+ * but allow the first timestamp to exceed INT32_MAX */
if (!s->timestamp)
s->unwrapped_timestamp += timestamp;
else
s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
s->timestamp = timestamp;
- pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
+ pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
+ s->base_timestamp;
}
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
@@ -477,15 +482,15 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
int ext;
AVStream *st;
uint32_t timestamp;
- int rv= 0;
+ int rv = 0;
- ext = buf[0] & 0x10;
+ ext = buf[0] & 0x10;
payload_type = buf[1] & 0x7f;
if (buf[1] & 0x80)
flags |= RTP_FLAG_MARKER;
- seq = AV_RB16(buf + 2);
+ seq = AV_RB16(buf + 2);
timestamp = AV_RB32(buf + 4);
- ssrc = AV_RB32(buf + 8);
+ ssrc = AV_RB32(buf + 8);
/* store the ssrc in the RTPDemuxContext */
s->ssrc = ssrc;
@@ -495,9 +500,9 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
st = s->st;
// only do something with this if all the rtp checks pass...
- if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
- {
- av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
+ if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
+ av_log(st ? st->codec : NULL, AV_LOG_ERROR,
+ "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
return -1;
}
@@ -509,8 +514,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
}
s->seq = seq;
- len -= 12;
- buf += 12;
+ len -= 12;
+ buf += 12;
/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
if (ext) {
@@ -528,7 +533,7 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
}
if (!st) {
- /* specific MPEG2TS demux support */
+ /* specific MPEG2-TS demux support */
ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
/* The only error that can be returned from ff_mpegts_parse_packet
* is "no more data to return from the provided buffer", so return
@@ -546,14 +551,15 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, ×tamp, buf, len, flags);
} else {
- // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
- switch(st->codec->codec_id) {
+ /* At this point, the RTP header has been stripped;
+ * This is ASSUMING that there is only 1 CSRC, which isn't wise. */
+ switch (st->codec->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
- /* better than nothing: skip mpeg audio RTP header */
+ /* better than nothing: skip MPEG audio RTP header */
if (len <= 4)
return -1;
- h = AV_RB32(buf);
+ h = AV_RB32(buf);
len -= 4;
buf += 4;
av_new_packet(pkt, len);
@@ -561,14 +567,14 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
- /* better than nothing: skip mpeg video RTP header */
+ /* better than nothing: skip MPEG video RTP header */
if (len <= 4)
return -1;
- h = AV_RB32(buf);
+ h = AV_RB32(buf);
buf += 4;
len -= 4;
if (h & (1 << 26)) {
- /* mpeg2 */
+ /* MPEG-2 */
if (len <= 4)
return -1;
buf += 4;
@@ -607,7 +613,7 @@ void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
{
- uint16_t seq = AV_RB16(buf + 2);
+ uint16_t seq = AV_RB16(buf + 2);
RTPPacket *cur = s->queue, *prev = NULL, *packet;
/* Find the correct place in the queue to insert the packet */
@@ -616,17 +622,17 @@ static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
if (diff < 0)
break;
prev = cur;
- cur = cur->next;
+ cur = cur->next;
}
packet = av_mallocz(sizeof(*packet));
if (!packet)
return;
packet->recvtime = av_gettime();
- packet->seq = seq;
- packet->len = len;
- packet->buf = buf;
- packet->next = cur;
+ packet->seq = seq;
+ packet->len = len;
+ packet->buf = buf;
+ packet->next = cur;
if (prev)
prev->next = packet;
else
@@ -657,7 +663,7 @@ static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
"RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
/* Parse the first packet in the queue, and dequeue it */
- rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
+ rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
next = s->queue->next;
av_free(s->queue->buf);
av_free(s->queue);
@@ -669,10 +675,10 @@ static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **bufptr, int len)
{
- uint8_t* buf = bufptr ? *bufptr : NULL;
+ uint8_t *buf = bufptr ? *bufptr : NULL;
int ret, flags = 0;
uint32_t timestamp;
- int rv= 0;
+ int rv = 0;
if (!buf) {
/* If parsing of the previous packet actually returned 0 or an error,
@@ -681,12 +687,12 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
if (s->prev_ret <= 0)
return rtp_parse_queued_packet(s, pkt);
/* return the next packets, if any */
- if(s->st && s->parse_packet) {
+ if (s->st && s->parse_packet) {
/* timestamp should be overwritten by parse_packet, if not,
* the packet is left with pts == AV_NOPTS_VALUE */
timestamp = RTP_NOTS_VALUE;
- rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
- s->st, pkt, ×tamp, NULL, 0, flags);
+ rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
+ s->st, pkt, ×tamp, NULL, 0, flags);
finalize_packet(s, pkt, timestamp);
return rv;
} else {
@@ -694,7 +700,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
if (s->read_buf_index >= s->read_buf_size)
return AVERROR(EAGAIN);
ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
- s->read_buf_size - s->read_buf_index);
+ s->read_buf_size - s->read_buf_index);
if (ret < 0)
return AVERROR(EAGAIN);
s->read_buf_index += ret;
@@ -786,14 +792,16 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
}
// remove protocol identifier
- while (*p && *p == ' ') p++; // strip spaces
- while (*p && *p != ' ') p++; // eat protocol identifier
- while (*p && *p == ' ') p++; // strip trailing spaces
+ while (*p && *p == ' ')
+ p++; // strip spaces
+ while (*p && *p != ' ')
+ p++; // eat protocol identifier
+ while (*p && *p == ' ')
+ p++; // strip trailing spaces
while (ff_rtsp_next_attr_and_value(&p,
attr, sizeof(attr),
value, value_size)) {
-
res = parse_fmtp(stream, data, attr, value);
if (res < 0 && res != AVERROR_PATCHWELCOME) {
av_free(value);
@@ -808,9 +816,9 @@ int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
{
av_init_packet(pkt);
- pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
+ pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
pkt->stream_index = stream_idx;
pkt->destruct = av_destruct_packet;
- *dyn_buf = NULL;
+ *dyn_buf = NULL;
return pkt->size;
}
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