diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-08-09 19:09:39 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-08-09 19:31:56 +0200 |
commit | 9f088a1ed4d34f0cf4244a4b80960af9e8f23dfc (patch) | |
tree | 049c8db5ee42c462bfe3d999146a224cff29bf03 /libavformat | |
parent | e1a983e6010930ab742bede275de1ccf921485b7 (diff) | |
parent | f69f4036f8cc3b673864dce01d2714fd5e49e8da (diff) | |
download | ffmpeg-9f088a1ed4d34f0cf4244a4b80960af9e8f23dfc.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
mpegvideo: reduce excessive inlining of mpeg_motion()
mpegvideo: convert mpegvideo_common.h to a .c file
build: factor out mpegvideo.o dependencies to CONFIG_MPEGVIDEO
Move MASK_ABS macro to libavcodec/mathops.h
x86: move MANGLE() and related macros to libavutil/x86/asm.h
x86: rename libavutil/x86_cpu.h to libavutil/x86/asm.h
aacdec: Don't fall back to the old output configuration when no old configuration is present.
rtmp: Add message tracking
rtsp: Support mpegts in raw udp packets
rtsp: Support receiving plain data over UDP without any RTP encapsulation
rtpdec: Remove an unused include
rtpenc: Remove an av_abort() that depends on user-supplied data
vsrc_movie: discourage its use with avconv.
avconv: allow no input files.
avconv: prevent invalid reads in transcode_init()
avconv: rename OutputStream.is_past_recording_time to finished.
Conflicts:
configure
doc/filters.texi
ffmpeg.c
ffmpeg.h
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/mpegvideo.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat')
-rw-r--r-- | libavformat/rtmpproto.c | 298 | ||||
-rw-r--r-- | libavformat/rtpdec_h264.c | 1 | ||||
-rw-r--r-- | libavformat/rtpenc.c | 21 | ||||
-rw-r--r-- | libavformat/rtsp.c | 51 | ||||
-rw-r--r-- | libavformat/rtsp.h | 8 | ||||
-rw-r--r-- | libavformat/version.h | 2 |
6 files changed, 224 insertions, 157 deletions
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c index 4120aa6b21..e48c740a2e 100644 --- a/libavformat/rtmpproto.c +++ b/libavformat/rtmpproto.c @@ -52,15 +52,17 @@ typedef enum { STATE_START, ///< client has not done anything yet STATE_HANDSHAKED, ///< client has performed handshake - STATE_RELEASING, ///< client releasing stream before publish it (for output) STATE_FCPUBLISH, ///< client FCPublishing stream (for output) - STATE_CONNECTING, ///< client connected to server successfully - STATE_READY, ///< client has sent all needed commands and waits for server reply STATE_PLAYING, ///< client has started receiving multimedia data from server STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output) STATE_STOPPED, ///< the broadcast has been stopped } ClientState; +typedef struct TrackedMethod { + char *name; + int id; +} TrackedMethod; + /** protocol handler context */ typedef struct RTMPContext { const AVClass *class; @@ -86,7 +88,6 @@ typedef struct RTMPContext { uint8_t flv_header[11]; ///< partial incoming flv packet header int flv_header_bytes; ///< number of initialized bytes in flv_header int nb_invokes; ///< keeps track of invoke messages - int create_stream_invoke; ///< invoke id for the create stream command char* tcurl; ///< url of the target stream char* flashver; ///< version of the flash plugin char* swfurl; ///< url of the swf player @@ -96,6 +97,9 @@ typedef struct RTMPContext { int client_buffer_time; ///< client buffer time in ms int flush_interval; ///< number of packets flushed in the same request (RTMPT only) int encrypted; ///< use an encrypted connection (RTMPE only) + TrackedMethod*tracked_methods; ///< tracked methods buffer + int nb_tracked_methods; ///< number of tracked methods + int tracked_methods_size; ///< size of the tracked methods buffer } RTMPContext; #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing @@ -121,6 +125,72 @@ static const uint8_t rtmp_server_key[] = { 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE }; +static int add_tracked_method(RTMPContext *rt, const char *name, int id) +{ + void *ptr; + + if (rt->nb_tracked_methods + 1 > rt->tracked_methods_size) { + rt->tracked_methods_size = (rt->nb_tracked_methods + 1) * 2; + ptr = av_realloc(rt->tracked_methods, + rt->tracked_methods_size * sizeof(*rt->tracked_methods)); + if (!ptr) + return AVERROR(ENOMEM); + rt->tracked_methods = ptr; + } + + rt->tracked_methods[rt->nb_tracked_methods].name = av_strdup(name); + if (!rt->tracked_methods[rt->nb_tracked_methods].name) + return AVERROR(ENOMEM); + rt->tracked_methods[rt->nb_tracked_methods].id = id; + rt->nb_tracked_methods++; + + return 0; +} + +static void del_tracked_method(RTMPContext *rt, int index) +{ + memmove(&rt->tracked_methods[index], &rt->tracked_methods[index + 1], + sizeof(*rt->tracked_methods) * (rt->nb_tracked_methods - index - 1)); + rt->nb_tracked_methods--; +} + +static void free_tracked_methods(RTMPContext *rt) +{ + int i; + + for (i = 0; i < rt->nb_tracked_methods; i ++) + av_free(rt->tracked_methods[i].name); + av_free(rt->tracked_methods); +} + +static int rtmp_send_packet(RTMPContext *rt, RTMPPacket *pkt, int track) +{ + int ret; + + if (pkt->type == RTMP_PT_INVOKE && track) { + GetByteContext gbc; + char name[128]; + double pkt_id; + int len; + + bytestream2_init(&gbc, pkt->data, pkt->data_size); + if ((ret = ff_amf_read_string(&gbc, name, sizeof(name), &len)) < 0) + goto fail; + + if ((ret = ff_amf_read_number(&gbc, &pkt_id)) < 0) + goto fail; + + if ((ret = add_tracked_method(rt, name, pkt_id)) < 0) + goto fail; + } + + ret = ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, + rt->prev_pkt[1]); +fail: + ff_rtmp_packet_destroy(pkt); + return ret; +} + static int rtmp_write_amf_data(URLContext *s, char *param, uint8_t **p) { char *field, *value; @@ -269,11 +339,7 @@ static int gen_connect(URLContext *s, RTMPContext *rt) pkt.data_size = p - pkt.data; - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 1); } /** @@ -297,11 +363,7 @@ static int gen_release_stream(URLContext *s, RTMPContext *rt) ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -325,11 +387,7 @@ static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt) ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -353,11 +411,7 @@ static int gen_fcunpublish_stream(URLContext *s, RTMPContext *rt) ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -380,13 +434,8 @@ static int gen_create_stream(URLContext *s, RTMPContext *rt) ff_amf_write_string(&p, "createStream"); ff_amf_write_number(&p, ++rt->nb_invokes); ff_amf_write_null(&p); - rt->create_stream_invoke = rt->nb_invokes; - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 1); } @@ -412,11 +461,7 @@ static int gen_delete_stream(URLContext *s, RTMPContext *rt) ff_amf_write_null(&p); ff_amf_write_number(&p, rt->main_channel_id); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -437,11 +482,7 @@ static int gen_buffer_time(URLContext *s, RTMPContext *rt) bytestream_put_be32(&p, rt->main_channel_id); bytestream_put_be32(&p, rt->client_buffer_time); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -469,11 +510,7 @@ static int gen_play(URLContext *s, RTMPContext *rt) ff_amf_write_string(&p, rt->playpath); ff_amf_write_number(&p, rt->live); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 1); } /** @@ -500,11 +537,7 @@ static int gen_publish(URLContext *s, RTMPContext *rt) ff_amf_write_string(&p, rt->playpath); ff_amf_write_string(&p, "live"); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 1); } /** @@ -529,11 +562,8 @@ static int gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt) p = pkt.data; bytestream_put_be16(&p, 7); bytestream_put_be32(&p, AV_RB32(ppkt->data+2)); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -551,11 +581,8 @@ static int gen_server_bw(URLContext *s, RTMPContext *rt) p = pkt.data; bytestream_put_be32(&p, rt->server_bw); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -576,11 +603,7 @@ static int gen_check_bw(URLContext *s, RTMPContext *rt) ff_amf_write_number(&p, RTMP_NOTIFICATION); ff_amf_write_null(&p); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 0); } /** @@ -598,11 +621,8 @@ static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts) p = pkt.data; bytestream_put_be32(&p, rt->bytes_read); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - return ret; + return rtmp_send_packet(rt, &pkt, 0); } static int gen_fcsubscribe_stream(URLContext *s, RTMPContext *rt, @@ -622,11 +642,7 @@ static int gen_fcsubscribe_stream(URLContext *s, RTMPContext *rt, ff_amf_write_null(&p); ff_amf_write_string(&p, subscribe); - ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, - rt->prev_pkt[1]); - ff_rtmp_packet_destroy(&pkt); - - return ret; + return rtmp_send_packet(rt, &pkt, 1); } int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap, @@ -1010,7 +1026,8 @@ static int handle_invoke(URLContext *s, RTMPPacket *pkt) RTMPContext *rt = s->priv_data; int i, t; const uint8_t *data_end = pkt->data + pkt->data_size; - int ret; + char *tracked_method = NULL; + int ret = 0; //TODO: check for the messages sent for wrong state? if (!memcmp(pkt->data, "\002\000\006_error", 9)) { @@ -1021,68 +1038,72 @@ static int handle_invoke(URLContext *s, RTMPPacket *pkt) av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr); return -1; } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) { - switch (rt->state) { - case STATE_HANDSHAKED: - if (!rt->is_input) { - if ((ret = gen_release_stream(s, rt)) < 0) - return ret; - if ((ret = gen_fcpublish_stream(s, rt)) < 0) - return ret; - rt->state = STATE_RELEASING; - } else { - if ((ret = gen_server_bw(s, rt)) < 0) - return ret; - rt->state = STATE_CONNECTING; - } - if ((ret = gen_create_stream(s, rt)) < 0) - return ret; - - if (rt->is_input) { - /* Send the FCSubscribe command when the name of live - * stream is defined by the user or if it's a live stream. */ - if (rt->subscribe) { - if ((ret = gen_fcsubscribe_stream(s, rt, - rt->subscribe)) < 0) - return ret; - } else if (rt->live == -1) { - if ((ret = gen_fcsubscribe_stream(s, rt, - rt->playpath)) < 0) - return ret; - } - } - break; - case STATE_FCPUBLISH: - rt->state = STATE_CONNECTING; - break; - case STATE_RELEASING: - rt->state = STATE_FCPUBLISH; - /* hack for Wowza Media Server, it does not send result for - * releaseStream and FCPublish calls */ - if (!pkt->data[10]) { - int pkt_id = av_int2double(AV_RB64(pkt->data + 11)); - if (pkt_id == rt->create_stream_invoke) - rt->state = STATE_CONNECTING; - } - if (rt->state != STATE_CONNECTING) - break; - case STATE_CONNECTING: - //extract a number from the result - if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) { - av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n"); - } else { - rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21)); - } - if (rt->is_input) { - if ((ret = gen_play(s, rt)) < 0) - return ret; - if ((ret = gen_buffer_time(s, rt)) < 0) - return ret; - } else { - if ((ret = gen_publish(s, rt)) < 0) - return ret; + GetByteContext gbc; + double pkt_id; + + bytestream2_init(&gbc, pkt->data + 10, pkt->data_size); + if ((ret = ff_amf_read_number(&gbc, &pkt_id)) < 0) + return ret; + + for (i = 0; i < rt->nb_tracked_methods; i++) { + if (rt->tracked_methods[i].id != pkt_id) + continue; + + tracked_method = rt->tracked_methods[i].name; + del_tracked_method(rt, i); + break; + } + + if (!tracked_method) { + /* Ignore this reply when the current method is not tracked. */ + return 0; + } + + if (!memcmp(tracked_method, "connect", 7)) { + if (!rt->is_input) { + if ((ret = gen_release_stream(s, rt)) < 0) + goto invoke_fail; + + if ((ret = gen_fcpublish_stream(s, rt)) < 0) + goto invoke_fail; + } else { + if ((ret = gen_server_bw(s, rt)) < 0) + goto invoke_fail; + } + + if ((ret = gen_create_stream(s, rt)) < 0) + goto invoke_fail; + + if (rt->is_input) { + /* Send the FCSubscribe command when the name of live + * stream is defined by the user or if it's a live stream. */ + if (rt->subscribe) { + if ((ret = gen_fcsubscribe_stream(s, rt, + rt->subscribe)) < 0) + goto invoke_fail; + } else if (rt->live == -1) { + if ((ret = gen_fcsubscribe_stream(s, rt, + rt->playpath)) < 0) + goto invoke_fail; } - rt->state = STATE_READY; - break; + } + } else if (!memcmp(tracked_method, "createStream", 12)) { + //extract a number from the result + if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) { + av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n"); + } else { + rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21)); + } + + if (!rt->is_input) { + if ((ret = gen_publish(s, rt)) < 0) + goto invoke_fail; + } else { + if ((ret = gen_play(s, rt)) < 0) + goto invoke_fail; + if ((ret = gen_buffer_time(s, rt)) < 0) + goto invoke_fail; + } } } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) { const uint8_t* ptr = pkt->data + 11; @@ -1113,7 +1134,9 @@ static int handle_invoke(URLContext *s, RTMPPacket *pkt) return ret; } - return 0; +invoke_fail: + av_free(tracked_method); + return ret; } /** @@ -1283,6 +1306,7 @@ static int rtmp_close(URLContext *h) if (rt->state > STATE_HANDSHAKED) ret = gen_delete_stream(h, rt); + free_tracked_methods(rt); av_freep(&rt->flv_data); ffurl_close(rt->stream); return ret; @@ -1570,10 +1594,8 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size) if (rt->flv_off == rt->flv_size) { rt->skip_bytes = 4; - if ((ret = ff_rtmp_packet_write(rt->stream, &rt->out_pkt, - rt->chunk_size, rt->prev_pkt[1])) < 0) + if ((ret = rtmp_send_packet(rt, &rt->out_pkt, 0)) < 0) return ret; - ff_rtmp_packet_destroy(&rt->out_pkt); rt->flv_size = 0; rt->flv_off = 0; rt->flv_header_bytes = 0; diff --git a/libavformat/rtpdec_h264.c b/libavformat/rtpdec_h264.c index f8a2272203..61e038afeb 100644 --- a/libavformat/rtpdec_h264.c +++ b/libavformat/rtpdec_h264.c @@ -37,7 +37,6 @@ #include "libavutil/avstring.h" #include "libavcodec/get_bits.h" #include "avformat.h" -#include "mpegts.h" #include "network.h" #include <assert.h> diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 6af23f760a..7cd7034fbf 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -281,8 +281,8 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) /* send an integer number of samples and compute time stamp and fill the rtp send buffer before sending. */ -static void rtp_send_samples(AVFormatContext *s1, - const uint8_t *buf1, int size, int sample_size_bits) +static int rtp_send_samples(AVFormatContext *s1, + const uint8_t *buf1, int size, int sample_size_bits) { RTPMuxContext *s = s1->priv_data; int len, max_packet_size, n; @@ -292,7 +292,7 @@ static void rtp_send_samples(AVFormatContext *s1, max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size; /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */ if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0) - av_abort(); + return AVERROR(EINVAL); n = 0; while (size > 0) { s->buf_ptr = s->buf; @@ -307,6 +307,7 @@ static void rtp_send_samples(AVFormatContext *s1, ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); n += (s->buf_ptr - s->buf); } + return 0; } static void rtp_send_mpegaudio(AVFormatContext *s1, @@ -461,25 +462,21 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) case AV_CODEC_ID_PCM_ALAW: case AV_CODEC_ID_PCM_U8: case AV_CODEC_ID_PCM_S8: - rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); - break; + return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); case AV_CODEC_ID_PCM_U16BE: case AV_CODEC_ID_PCM_U16LE: case AV_CODEC_ID_PCM_S16BE: case AV_CODEC_ID_PCM_S16LE: - rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels); - break; + return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels); case AV_CODEC_ID_ADPCM_G722: /* The actual sample size is half a byte per sample, but since the * stream clock rate is 8000 Hz while the sample rate is 16000 Hz, * the correct parameter for send_samples_bits is 8 bits per stream * clock. */ - rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); - break; + return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); case AV_CODEC_ID_ADPCM_G726: - rtp_send_samples(s1, pkt->data, size, - st->codec->bits_per_coded_sample * st->codec->channels); - break; + return rtp_send_samples(s1, pkt->data, size, + st->codec->bits_per_coded_sample * st->codec->channels); case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: rtp_send_mpegaudio(s1, pkt->data, size); diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 633ec39ecb..b5e6858e00 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -46,6 +46,7 @@ #include "rtpenc_chain.h" #include "url.h" #include "rtpenc.h" +#include "mpegts.h" //#define DEBUG @@ -370,7 +371,9 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, get_word(buf1, sizeof(buf1), &p); /* port */ rtsp_st->sdp_port = atoi(buf1); - get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */ + get_word(buf1, sizeof(buf1), &p); /* protocol */ + if (!strcmp(buf1, "udp")) + rt->transport = RTSP_TRANSPORT_RAW; /* XXX: handle list of formats */ get_word(buf1, sizeof(buf1), &p); /* format list */ @@ -378,6 +381,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) { /* no corresponding stream */ + if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC) + rt->ts = ff_mpegts_parse_open(s); } else if (rt->server_type == RTSP_SERVER_WMS && codec_type == AVMEDIA_TYPE_DATA) { /* RTX stream, a stream that carries all the other actual @@ -563,7 +568,7 @@ void ff_rtsp_undo_setup(AVFormatContext *s) avformat_free_context(rtpctx); } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) ff_rdt_parse_close(rtsp_st->transport_priv); - else if (CONFIG_RTPDEC) + else if (rt->transport == RTSP_TRANSPORT_RAW && CONFIG_RTPDEC) ff_rtp_parse_close(rtsp_st->transport_priv); } rtsp_st->transport_priv = NULL; @@ -594,6 +599,8 @@ void ff_rtsp_close_streams(AVFormatContext *s) if (rt->asf_ctx) { avformat_close_input(&rt->asf_ctx); } + if (rt->ts && CONFIG_RTPDEC) + ff_mpegts_parse_close(rt->ts); av_free(rt->p); av_free(rt->recvbuf); } @@ -617,6 +624,8 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) rtsp_st->rtp_handle = NULL; if (ret < 0) return ret; + } else if (rt->transport == RTSP_TRANSPORT_RAW) { + return 0; // Don't need to open any parser here } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index, rtsp_st->dynamic_protocol_context, @@ -629,7 +638,7 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) if (!rtsp_st->transport_priv) { return AVERROR(ENOMEM); - } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) { + } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) { if (rtsp_st->dynamic_handler) { ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv, rtsp_st->dynamic_protocol_context, @@ -698,6 +707,15 @@ static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p) get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p); profile[0] = '\0'; th->transport = RTSP_TRANSPORT_RDT; + } else if (!av_strcasecmp(transport_protocol, "raw")) { + get_word_sep(profile, sizeof(profile), "/;,", &p); + lower_transport[0] = '\0'; + /* raw/raw/<protocol> */ + if (*p == '/') { + get_word_sep(lower_transport, sizeof(lower_transport), + ";,", &p); + } + th->transport = RTSP_TRANSPORT_RAW; } if (!av_strcasecmp(lower_transport, "TCP")) th->lower_transport = RTSP_LOWER_TRANSPORT_TCP; @@ -1187,6 +1205,8 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, if (rt->transport == RTSP_TRANSPORT_RDT) trans_pref = "x-pn-tng"; + else if (rt->transport == RTSP_TRANSPORT_RAW) + trans_pref = "RAW/RAW"; else trans_pref = "RTP/AVP"; @@ -1753,8 +1773,15 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) if (rt->cur_transport_priv) { if (rt->transport == RTSP_TRANSPORT_RDT) { ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); - } else + } else if (rt->transport == RTSP_TRANSPORT_RTP) { ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); + } else if (rt->ts && CONFIG_RTPDEC) { + ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos); + if (ret >= 0) { + rt->recvbuf_pos += ret; + ret = rt->recvbuf_pos < rt->recvbuf_len; + } + } if (ret == 0) { rt->cur_transport_priv = NULL; return 0; @@ -1817,7 +1844,7 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) return AVERROR_EOF; if (rt->transport == RTSP_TRANSPORT_RDT) { ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); - } else { + } else if (rt->transport == RTSP_TRANSPORT_RTP) { ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); if (ret < 0) { /* Either bad packet, or a RTCP packet. Check if the @@ -1856,6 +1883,20 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) return AVERROR_EOF; } } + } else if (rt->ts && CONFIG_RTPDEC) { + ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len); + if (ret >= 0) { + if (ret < len) { + rt->recvbuf_len = len; + rt->recvbuf_pos = ret; + rt->cur_transport_priv = rt->ts; + return 1; + } else { + ret = 0; + } + } + } else { + return AVERROR_INVALIDDATA; } end: if (ret < 0) diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 6d9680161e..e55073c430 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -52,6 +52,7 @@ enum RTSPLowerTransport { enum RTSPTransport { RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ + RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */ RTSP_TRANSPORT_NB }; @@ -310,6 +311,13 @@ typedef struct RTSPState { * other cases, this is a copy of AVFormatContext->filename. */ char control_uri[1024]; + /** The following are used for parsing raw mpegts in udp */ + //@{ + struct MpegTSContext *ts; + int recvbuf_pos; + int recvbuf_len; + //@} + /** Additional output handle, used when input and output are done * separately, eg for HTTP tunneling. */ URLContext *rtsp_hd_out; diff --git a/libavformat/version.h b/libavformat/version.h index 53d9a06520..750b7652e2 100644 --- a/libavformat/version.h +++ b/libavformat/version.h @@ -31,7 +31,7 @@ #define LIBAVFORMAT_VERSION_MAJOR 54 #define LIBAVFORMAT_VERSION_MINOR 22 -#define LIBAVFORMAT_VERSION_MICRO 103 +#define LIBAVFORMAT_VERSION_MICRO 104 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \ LIBAVFORMAT_VERSION_MINOR, \ |