mirror of
https://github.com/xenia-project/FFmpeg.git
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rtpdec: K&R formatting and spelling cosmetics
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This commit is contained in:
Martin Storsjö
Diego Biurrun
parent
ba0c898120
commit
5d471b73d2
@ -25,49 +25,46 @@
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#include "libavcodec/get_bits.h"
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#include "avformat.h"
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#include "mpegts.h"
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#include "url.h"
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#include "network.h"
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#include "url.h"
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#include "rtpdec.h"
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#include "rtpdec_formats.h"
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//#define DEBUG
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/* TODO: - add RTCP statistics reporting (should be optional).
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- add support for h263/mpeg4 packetized output : IDEA: send a
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buffer to 'rtp_write_packet' contains all the packets for ONE
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frame. Each packet should have a four byte header containing
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the length in big endian format (same trick as
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'ffio_open_dyn_packet_buf')
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*/
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/* TODO:
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* - add RTCP statistics reporting (should be optional).
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*
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* - add support for H.263/MPEG-4 packetized output: IDEA: send a
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* buffer to 'rtp_write_packet' contains all the packets for ONE
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* frame. Each packet should have a four byte header containing
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* the length in big-endian format (same trick as
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* 'ffio_open_dyn_packet_buf').
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*/
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static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
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.enc_name = "X-MP3-draft-00",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_MP3ADU,
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.enc_name = "X-MP3-draft-00",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_MP3ADU,
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};
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static RTPDynamicProtocolHandler speex_dynamic_handler = {
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.enc_name = "speex",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_SPEEX,
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.enc_name = "speex",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_SPEEX,
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};
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static RTPDynamicProtocolHandler opus_dynamic_handler = {
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.enc_name = "opus",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_OPUS,
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.enc_name = "opus",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = AV_CODEC_ID_OPUS,
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};
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/* statistics functions */
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static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
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static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler = NULL;
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void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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{
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handler->next= RTPFirstDynamicPayloadHandler;
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RTPFirstDynamicPayloadHandler= handler;
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handler->next = RTPFirstDynamicPayloadHandler;
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RTPFirstDynamicPayloadHandler = handler;
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}
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void av_register_rtp_dynamic_payload_handlers(void)
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@ -108,7 +105,7 @@ void av_register_rtp_dynamic_payload_handlers(void)
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}
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RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
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enum AVMediaType codec_type)
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enum AVMediaType codec_type)
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{
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RTPDynamicProtocolHandler *handler;
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for (handler = RTPFirstDynamicPayloadHandler;
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@ -120,7 +117,7 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
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}
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RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
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enum AVMediaType codec_type)
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enum AVMediaType codec_type)
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{
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RTPDynamicProtocolHandler *handler;
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for (handler = RTPFirstDynamicPayloadHandler;
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@ -131,7 +128,8 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
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return NULL;
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}
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
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int len)
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{
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int payload_len;
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while (len >= 4) {
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@ -140,11 +138,12 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
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switch (buf[1]) {
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case RTCP_SR:
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if (payload_len < 20) {
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av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
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av_log(NULL, AV_LOG_ERROR,
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"Invalid length for RTCP SR packet\n");
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return AVERROR_INVALIDDATA;
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}
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s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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s->last_rtcp_timestamp = AV_RB32(buf + 16);
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if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
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s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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@ -164,7 +163,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
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return -1;
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}
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#define RTP_SEQ_MOD (1<<16)
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#define RTP_SEQ_MOD (1 << 16)
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static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
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{
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@ -174,8 +173,9 @@ static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
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}
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/*
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* called whenever there is a large jump in sequence numbers, or when they get out of probation...
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*/
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* Called whenever there is a large jump in sequence numbers,
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* or when they get out of probation...
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*/
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static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
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{
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s->max_seq = seq;
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@ -189,9 +189,7 @@ static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
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s->transit = 0;
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}
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/*
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* returns 1 if we should handle this packet.
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*/
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/* Returns 1 if we should handle this packet. */
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static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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{
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uint16_t udelta = seq - s->max_seq;
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@ -199,7 +197,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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const int MAX_MISORDER = 100;
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const int MIN_SEQUENTIAL = 2;
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/* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
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/* source not valid until MIN_SEQUENTIAL packets with sequence
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* seq. numbers have been received */
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if (s->probation) {
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if (seq == s->max_seq + 1) {
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s->probation--;
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@ -211,7 +210,7 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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}
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} else {
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s->probation = MIN_SEQUENTIAL - 1;
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s->max_seq = seq;
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s->max_seq = seq;
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}
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} else if (udelta < MAX_DROPOUT) {
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// in order, with permissible gap
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@ -223,7 +222,8 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
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// sequence made a large jump...
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if (seq == s->bad_seq) {
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// two sequential packets-- assume that the other side restarted without telling us; just resync.
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/* two sequential packets -- assume that the other side
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* restarted without telling us; just resync. */
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rtp_init_sequence(s, seq);
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} else {
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s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
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@ -256,7 +256,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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return -1;
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/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
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/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
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/* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
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s->octet_count += count;
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rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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RTCP_TX_RATIO_DEN;
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@ -277,15 +277,15 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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avio_wb32(pb, s->ssrc); // server SSRC
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// some placeholders we should really fill...
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// RFC 1889/p64
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extended_max = stats->cycles + stats->max_seq;
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expected = extended_max - stats->base_seq + 1;
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lost = expected - stats->received;
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lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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expected_interval = expected - stats->expected_prior;
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extended_max = stats->cycles + stats->max_seq;
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expected = extended_max - stats->base_seq + 1;
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lost = expected - stats->received;
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lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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expected_interval = expected - stats->expected_prior;
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stats->expected_prior = expected;
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received_interval = stats->received - stats->received_prior;
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received_interval = stats->received - stats->received_prior;
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stats->received_prior = stats->received;
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lost_interval = expected_interval - received_interval;
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lost_interval = expected_interval - received_interval;
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if (expected_interval == 0 || lost_interval <= 0)
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fraction = 0;
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else
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@ -301,7 +301,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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avio_wb32(pb, 0); /* last SR timestamp */
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avio_wb32(pb, 0); /* delay since last SR */
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} else {
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uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
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uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
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uint32_t delay_since_last = ntp_time - s->last_rtcp_ntp_time;
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avio_wb32(pb, middle_32_bits); /* last SR timestamp */
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@ -318,23 +318,22 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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avio_w8(pb, len);
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avio_write(pb, s->hostname, len);
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// padding
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for (len = (6 + len) % 4; len % 4; len++) {
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for (len = (6 + len) % 4; len % 4; len++)
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avio_w8(pb, 0);
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}
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avio_flush(pb);
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len = avio_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf) {
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int av_unused result;
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av_dlog(s->ic, "sending %d bytes of RR\n", len);
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result= ffurl_write(s->rtp_ctx, buf, len);
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result = ffurl_write(s->rtp_ctx, buf, len);
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av_dlog(s->ic, "result from ffurl_write: %d\n", result);
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av_free(buf);
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}
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return 0;
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}
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void ff_rtp_send_punch_packets(URLContext* rtp_handle)
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void ff_rtp_send_punch_packets(URLContext *rtp_handle)
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{
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AVIOContext *pb;
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uint8_t *buf;
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@ -372,25 +371,26 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle)
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av_free(buf);
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}
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/**
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* open a new RTP parse context for stream 'st'. 'st' can be NULL for
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* MPEG2TS streams to indicate that they should be demuxed inside the
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* MPEG2-TS streams to indicate that they should be demuxed inside the
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* rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
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*/
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RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
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RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
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URLContext *rtpc, int payload_type,
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int queue_size)
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{
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RTPDemuxContext *s;
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s = av_mallocz(sizeof(RTPDemuxContext));
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if (!s)
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return NULL;
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s->payload_type = payload_type;
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s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
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s->payload_type = payload_type;
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s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
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s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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s->ic = s1;
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s->st = st;
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s->queue_size = queue_size;
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s->ic = s1;
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s->st = st;
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s->queue_size = queue_size;
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rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
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if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
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s->ts = ff_mpegts_parse_open(s->ic);
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@ -399,7 +399,7 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext
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return NULL;
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}
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} else if (st) {
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switch(st->codec->codec_id) {
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switch (st->codec->codec_id) {
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case AV_CODEC_ID_MPEG1VIDEO:
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case AV_CODEC_ID_MPEG2VIDEO:
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case AV_CODEC_ID_MP2:
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@ -432,11 +432,12 @@ void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
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RTPDynamicProtocolHandler *handler)
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{
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s->dynamic_protocol_context = ctx;
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s->parse_packet = handler->parse_packet;
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s->parse_packet = handler->parse_packet;
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}
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/**
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* This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
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* This was the second switch in rtp_parse packet.
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* Normalizes time, if required, sets stream_index, etc.
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*/
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static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
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{
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@ -452,7 +453,9 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
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/* compute pts from timestamp with received ntp_time */
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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/* convert to the PTS timebase */
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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);
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addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
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s->st->time_base.den,
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(uint64_t) s->st->time_base.num << 32);
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pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
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delta_timestamp;
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return;
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@ -460,13 +463,15 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
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if (!s->base_timestamp)
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s->base_timestamp = timestamp;
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/* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
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/* assume that the difference is INT32_MIN < x < INT32_MAX,
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* but allow the first timestamp to exceed INT32_MAX */
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if (!s->timestamp)
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s->unwrapped_timestamp += timestamp;
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else
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s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
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s->timestamp = timestamp;
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pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
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pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
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s->base_timestamp;
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}
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static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
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@ -477,15 +482,15 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
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int ext;
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AVStream *st;
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uint32_t timestamp;
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int rv= 0;
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int rv = 0;
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ext = buf[0] & 0x10;
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ext = buf[0] & 0x10;
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payload_type = buf[1] & 0x7f;
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if (buf[1] & 0x80)
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flags |= RTP_FLAG_MARKER;
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seq = AV_RB16(buf + 2);
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seq = AV_RB16(buf + 2);
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timestamp = AV_RB32(buf + 4);
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ssrc = AV_RB32(buf + 8);
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ssrc = AV_RB32(buf + 8);
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/* store the ssrc in the RTPDemuxContext */
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s->ssrc = ssrc;
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@ -495,9 +500,9 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
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st = s->st;
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// only do something with this if all the rtp checks pass...
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if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
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{
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av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
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if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
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av_log(st ? st->codec : NULL, AV_LOG_ERROR,
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"RTP: PT=%02x: bad cseq %04x expected=%04x\n",
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payload_type, seq, ((s->seq + 1) & 0xffff));
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return -1;
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}
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@ -509,8 +514,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
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}
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s->seq = seq;
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len -= 12;
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buf += 12;
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len -= 12;
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buf += 12;
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/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
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if (ext) {
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@ -528,7 +533,7 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
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}
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if (!st) {
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/* specific MPEG2TS demux support */
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/* specific MPEG2-TS demux support */
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ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
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/* The only error that can be returned from ff_mpegts_parse_packet
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* is "no more data to return from the provided buffer", so return
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@ -546,14 +551,15 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
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rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
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s->st, pkt, ×tamp, buf, len, flags);
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} 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;
|
||||
}
|
||||
|
||||
Loading…
Reference in New Issue
Block a user