From 5d471b73d20616f5ac701ff62e5de49465cda264 Mon Sep 17 00:00:00 2001 From: Martin Storsjö Date: Fri, 7 Dec 2012 15:50:17 +0200 Subject: rtpdec: K&R formatting and spelling cosmetics Signed-off-by: Diego Biurrun --- libavformat/rtpdec.c | 212 ++++++++++++++++++++++++++------------------------- 1 file changed, 110 insertions(+), 102 deletions(-) (limited to 'libavformat/rtpdec.c') diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index a305dd6957..38ce7f62d4 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; } -- cgit v1.2.3