From 2585748c210a9f79f3072e457e826f1300ab2da9 Mon Sep 17 00:00:00 2001 From: kasperiio <101901857+kasperiio@users.noreply.github.com> Date: Sat, 5 Sep 2026 19:51:56 +0300 Subject: [PATCH 1/2] rtsp: send interleaved frames in one write, stamp per frame, add G.711 On a small SoC the RTSP sender spent more CPU on send() syscalls and on spinning over partial writes than on encoding. - Interleaved (TCP) packets are staged and written once per frame; a send() costs about 100 us here whatever its size. - The socket send buffer is raised to 512 KB, enough for one keyframe, so the sender no longer spins on EAGAIN for as long as the link takes to drain. - Partial writes wait in poll() instead of usleep(1000). - Timestamps are taken once per frame from the encoder's capture time. Stamping only the marker packet gave every earlier packet of a frame the previous frame's time, so receivers saw timestamps go backwards within a frame and players stalled and then raced to catch up. - G.711 A-law (payload type 8) as an alternative to MP3 for RTSP audio, which also lets the software MP3 encoder be skipped when nothing consumes MP3. - The MP4 muxer no longer builds moof/mdat when no HTTP client is connected, and resets its cached header when the stream configuration changes. - /api/rtsp and /api/onvif expose those settings, with the matching web UI. - bitbuf: copy with memcpy instead of a byte loop. Review fixes: - __tcp_flush_each() gated on track 0, so a client that set up only audio over interleaved TCP staged packets that were never flushed. - The RTSP audio codec is now latched when the server starts. Changing it through /api/rtsp used to switch the payload type and clock under sessions already negotiated for the other one; the API documented a restart but the media path read the value live. Both the API and the config parser now accept only pcma and mp3. - The 8 kHz resampling ratio is carried in 16.16 fixed point. Truncating srate / 8000 emitted 8820 samples a second at 44.1 kHz against an 8 kHz SDP and RTP clock, so G.711 ran fast and packets were 18.1 ms rather than 20. - G.711 timestamps advance by the samples actually sent rather than by millis(). - rtsp.audio_codec is parsed with a new bounded parse_param_value_n(); the unbounded parse_param_value() sprintf()s into the caller's buffer, which overflows the eight-byte field. Existing callers are unchanged. - /api/rtsp and /api/onvif escape the usernames and codec they echo back, which otherwise made the response unparseable for the web UI. --- res/index.html | 33 ++++++++++ src/app_config.c | 14 +++++ src/app_config.h | 1 + src/fmt/bitbuf.c | 4 +- src/fmt/moov.c | 3 +- src/fmt/mp4.c | 4 ++ src/hal/config.c | 21 ++++++- src/hal/config.h | 6 ++ src/main.c | 1 + src/media.c | 75 +++++++++++++++++++++- src/media.h | 2 + src/rtsp/rtp.c | 139 +++++++++++++++++++++++++++++++++++++++-- src/rtsp/rtsp.c | 22 ++++++- src/rtsp/rtsp.h | 5 ++ src/rtsp/rtsp_server.h | 1 + src/server.c | 120 +++++++++++++++++++++++++++++++++++ src/server.h | 1 + 17 files changed, 439 insertions(+), 13 deletions(-) diff --git a/res/index.html b/res/index.html index bf037ac0..78df54c9 100644 --- a/res/index.html +++ b/res/index.html @@ -372,6 +372,8 @@ apiRead('mjpeg'); apiRead('night'); apiRead('record'); + apiRead('rtsp'); + apiRead('onvif'); toggleRecord(document.getElementById('record_state')); apiRead('time'); @@ -676,6 +678,37 @@

Media

+

Network

+
+
+
+ RTSP + + + + + + + +
+ Apply + Port and codec changes take effect after a restart; use Save to keep them. +
+
+
+
+ ONVIF + + + + + +
+ Apply + Applies after a restart; use Save to keep it. +
+
+

OSD

-

Network

-
-
-
- RTSP - - - - - - - -
- Apply - Port and codec changes take effect after a restart; use Save to keep them. -
-
-
-
- ONVIF - - - - - -
- Apply - Applies after a restart; use Save to keep it. -
-
-

OSD

+ +
+
+ +
+ Save + Restart + +
+
+
+

Network

+
+
+
+ RTSP + + + + + + + + + + + + + + + + + + + + + + + + + +
+ Apply +
+
+
+
+ ONVIF + + + + + + + + + + + + + + + + + +
+ Apply +
+
+
+
+
@@ -860,6 +917,7 @@

System

+ Snapshot diff --git a/src/app_config.c b/src/app_config.c index 2e3e3dbc..ed828771 100644 --- a/src/app_config.c +++ b/src/app_config.c @@ -435,11 +435,11 @@ enum ConfigError app_config_parse(void) { parse_bool(&ini, "rtsp", "enable", &app_config.rtsp_enable); parse_int(&ini, "rtsp", "port", 0, USHRT_MAX, &app_config.rtsp_port); { - /* Bounded, and only the two codecs the RTP path implements: anything + /* Bounded, and only the codecs the RTP path implements: anything * else would advertise a payload type nothing produces. */ char codec[16] = {0}; if (parse_param_value_n(&ini, "rtsp", "audio_codec", codec, sizeof(codec)) == CONFIG_OK) { - if (EQUALS(codec, "pcma") || EQUALS(codec, "mp3")) { + if (EQUALS(codec, "pcma") || EQUALS(codec, "pcmu") || EQUALS(codec, "mp3")) { strncpy(app_config.rtsp_audio_codec, codec, sizeof(app_config.rtsp_audio_codec) - 1); app_config.rtsp_audio_codec[sizeof(app_config.rtsp_audio_codec) - 1] = 0; diff --git a/src/app_config.h b/src/app_config.h index 251b9995..029e0d48 100644 --- a/src/app_config.h +++ b/src/app_config.h @@ -60,7 +60,7 @@ struct AppConfig { char rtsp_auth_user[32]; char rtsp_auth_pass[32]; int rtsp_port; - char rtsp_audio_codec[8]; /* pcma (G.711 A-law, 8 kHz) or mp3 */ + char rtsp_audio_codec[8]; /* pcma (G.711A), pcmu (G.711U) or mp3 */ // [record] bool record_enable; diff --git a/src/hal/tools.c b/src/hal/tools.c index 9c708b7d..83ec44af 100644 --- a/src/hal/tools.c +++ b/src/hal/tools.c @@ -140,6 +140,28 @@ int compile_regex(regex_t *r, const char *regex_text) { } +int escape_json(char *dst, const char *src, size_t maxlen) +{ + size_t o = 0; + + for (; src && *src && o + 7 < maxlen; src++) { + unsigned char c = (unsigned char)*src; + + if (c == '"' || c == '\\') { + dst[o++] = '\\'; + dst[o++] = c; + } else if (c < 0x20) { + o += snprintf(dst + o, maxlen - o, "\\u%04x", c); + } else { + dst[o++] = c; + } + } + + dst[o] = 0; + return o; +} + + int escape_url(char *dst, const char *src, size_t maxlen) { static const char hex[] = "0123456789ABCDEF"; int len = 0; @@ -517,3 +539,19 @@ void uuid_generate(char *uuid) { } uuid[36] = '\0'; } + +int parse_ranged(const char *value, long min, long max, long *out) +{ + char *remain; + long result; + + errno = 0; + result = strtol(value, &remain, 10); + + if (remain == value || (remain && *remain) || errno == ERANGE || + result < min || result > max) + return 0; + + *out = result; + return 1; +} diff --git a/src/hal/tools.h b/src/hal/tools.h index c6dfd362..03120616 100644 --- a/src/hal/tools.h +++ b/src/hal/tools.h @@ -47,6 +47,8 @@ int color_parse(const char *str); int compile_regex(regex_t *r, const char *regex_text); +int escape_json(char *dst, const char *src, size_t maxlen); + int escape_url(char *dst, const char *src, size_t maxlen); void generate_nonce(char *nonce, size_t len); @@ -73,6 +75,8 @@ char *memstr(char *haystack, char *needle, int size, char needlesize); unsigned int millis(); +int parse_ranged(const char *value, long min, long max, long *out); + void reverse(void *arr, size_t width); void sha1_init(sha1_context *context); diff --git a/src/media.c b/src/media.c index c142120f..9fc5cb05 100644 --- a/src/media.c +++ b/src/media.c @@ -51,19 +51,29 @@ static unsigned char pcm_to_alaw(short pcm) return (unsigned char)((s ^ (sign ? 0xd5 : 0x55))); } -/* RTSP G.711: resample the capture rate to 8 kHz (block average) and A-law - * encode, sending one RTP packet per 20 ms; MP3 keeps feeding the MP4/HTTP/RTMP - * outputs. - * - * The ratio is carried in 16.16 fixed point rather than as an integer divisor: - * srate is accepted anywhere in 8000..96000, and truncating 44100 / 8000 to 5 - * emitted 8820 samples a second against an SDP and RTP clock of 8000, so the - * stream ran fast and the 160-byte packets were 18.1 ms rather than 20. */ +/* mu-law (G.711 U, PT 0): 4085-step non-uniform compressor. Mirrors the A-law + * routine so both stay branch-light for soft-float ARM. */ +static unsigned char pcm_to_ulaw(short pcm) +{ + int sign = (pcm >> 8) & 0x80, s = sign ? -pcm : pcm, exp = 7; + if (s > 32635) s = 32635; + s += 0x84; /* bias: mu-law does not encode zero alone */ + if (s > 0x7FFF) s = 0x7FFF; + for (int m = 0x4000; !(s & m) && exp > 0; m >>= 1) exp--; + return (unsigned char)~(sign | (exp << 4) | ((s >> (exp + 3)) & 0x0f)); +} + +static char rtsp_codec[sizeof(app_config.rtsp_audio_codec)]; + +/* Any G.711 codec: the 8 kHz sample drop + encode path, versus MP3 */ +static char rtsp_g711_active(void) { return EQUALS(rtsp_codec, "pcma") || EQUALS(rtsp_codec, "pcmu"); } + static void rtsp_pcma_feed(short *pcm, unsigned int samples) { static unsigned char alaw[160]; static unsigned int fill, phase, step; - static int acc, accN; + static int acc, acc_n; + char ulaw = EQUALS(rtsp_codec, "pcmu"); if (!step) { unsigned int srate = app_config.audio_srate > 0 ? app_config.audio_srate : 8000; step = (srate << 16) / 8000; @@ -71,35 +81,27 @@ static void rtsp_pcma_feed(short *pcm, unsigned int samples) } for (unsigned int i = 0; i < samples; i++) { acc += pcm[i]; - accN++; + acc_n++; phase += 1 << 16; if (phase < step) continue; phase -= step; - alaw[fill++] = pcm_to_alaw((short)(acc / accN)); - acc = 0; accN = 0; + alaw[fill++] = ulaw ? pcm_to_ulaw((short)(acc / acc_n)) : pcm_to_alaw((short)(acc / acc_n)); + acc = 0; acc_n = 0; if (fill == sizeof(alaw)) { - rtp_send_pcma(rtspHandle, alaw, sizeof(alaw)); + if (ulaw) + rtp_send_pcmu(rtspHandle, alaw, sizeof(alaw)); + else + rtp_send_pcma(rtspHandle, alaw, sizeof(alaw)); fill = 0; } } } -/* The codec RTSP sessions were negotiated with. - * - * app_config.rtsp_audio_codec can be changed at runtime through /api/rtsp, but - * switching the payload type and clock under a live session would leave every - * client decoding the wrong thing without a new DESCRIBE/SETUP. The value the - * media path uses is therefore latched when the server starts, which is what - * the API means when it says the change applies after a restart. */ -static char rtspCodec[sizeof(app_config.rtsp_audio_codec)]; - void rtsp_latch_audio_codec(void) { - strncpy(rtspCodec, app_config.rtsp_audio_codec, sizeof(rtspCodec) - 1); - rtspCodec[sizeof(rtspCodec) - 1] = 0; + strncpy(rtsp_codec, app_config.rtsp_audio_codec, sizeof(rtsp_codec) - 1); + rtsp_codec[sizeof(rtsp_codec) - 1] = 0; } -static char rtsp_pcma_active(void) { return EQUALS(rtspCodec, "pcma"); } - void *aenc_thread(void) { static uint8_t frame_buf[AUD_FRAME_MAX + 2]; const uint32_t mp3FrmSize = @@ -117,7 +119,7 @@ void *aenc_thread(void) { pthread_mutex_lock(&mp4Mtx); mp4_ingest_audio(mp3Buf.buf, mp3FrmSize); pthread_mutex_unlock(&mp4Mtx); - if (app_config.rtsp_enable && !rtsp_pcma_active()) + if (app_config.rtsp_enable && !rtsp_g711_active()) rtp_send_mp3(rtspHandle, mp3Buf.buf, mp3FrmSize); rtmp_ingest_audio(mp3Buf.buf, mp3FrmSize); mp3Buf.offset -= mp3FrmSize; @@ -144,15 +146,11 @@ void *aenc_thread(void) { outFrame.seq = seq++; outFrame.timestamp = millis(); send_pcm_to_client(&outFrame); - if (app_config.rtsp_enable && rtsp_pcma_active()) + if (app_config.rtsp_enable && rtsp_g711_active()) rtsp_pcma_feed((short *)(frame_buf + 2), flen / 2); - /* The software MP3 encoder is the single most expensive thing on a - * small SoC (24% of an ARM1176 at 48 kHz); skip it while nothing - * consumes MP3: no HTTP MP3/MP4 client, no recording, no RTMP, and - * RTSP carrying G.711 */ - if (!recordOn && !app_config.stream_enable && !http_audio_clients() && - !(app_config.rtsp_enable && !rtsp_pcma_active())) { + if (!recordOn && !app_config.stream_enable && !any_http_audio() && + !(app_config.rtsp_enable && !rtsp_g711_active())) { pcmPos = 0; continue; } diff --git a/src/rtsp/rtp.c b/src/rtsp/rtp.c index 71e02933..080a83d7 100644 --- a/src/rtsp/rtp.c +++ b/src/rtsp/rtp.c @@ -36,17 +36,14 @@ struct __transfer_set_t { int track_id; }; -/* 90 kHz video / 8 kHz G.711 timestamps of the frame being sent */ static unsigned int __frame_ts_video, __frame_ts_audio; -/* Block until the socket can take more data (or 100 ms), instead of spinning */ static inline void __wait_out(int fd) { struct pollfd p = { .fd = fd, .events = POLLOUT }; poll(&p, 1, 100); } -/* Write out the staged interleaved data; call with write_mutex held */ static int __tcp_flush(struct connection_item_t *con) { unsigned int sent = 0; @@ -193,8 +190,8 @@ static inline int __transfer_nal_mpga(struct list_head_t *trans_list, unsigned c return SUCCESS; } -/* One RTP packet per chunk of G.711 A-law bytes (20 ms = 160 bytes at 8 kHz) */ -static inline int __transfer_pcma(struct list_head_t *trans_list, unsigned char *ptr, size_t size) +/* One RTP packet per 20 ms of 8 kHz G.711 (PT 8 for A-law, 0 for mu-law) */ +static inline int __transfer_g711(struct list_head_t *trans_list, unsigned char *ptr, size_t size, unsigned char pt) { struct nal_rtp_t rtp; rtp_hdr_t *p_header = &(rtp.packet.header); @@ -203,7 +200,7 @@ static inline int __transfer_pcma(struct list_head_t *trans_list, unsigned char p_header->p = 0; p_header->x = 0; p_header->cc = 0; - p_header->pt = 8; + p_header->pt = pt; p_header->m = 1; memcpy(rtp.packet.payload, ptr, size); @@ -227,10 +224,6 @@ static inline int __rtp_send_eachconnection(struct list_t *e, void *v) if (!con->trans[track_id].server_port_rtp && !con->trans[track_id].is_tcp) return SUCCESS; rtp->packet.header.seq = htons(con->trans[track_id].rtp_seq); - /* One timestamp per frame, taken when the frame starts (see rtp_send_*): - * stamping only the marker packet gave every earlier packet of a frame - * the previous frame's time, so receivers saw timestamps run backwards - * inside a frame and players stalled and then raced to catch up */ con->trans[track_id].rtp_timestamp = track_id ? __frame_ts_audio : __frame_ts_video; rtp->packet.header.ts = htonl(con->trans[track_id].rtp_timestamp); rtp->packet.header.ssrc = htonl(con->ssrc); @@ -245,7 +238,6 @@ static inline int __rtp_send_eachconnection(struct list_t *e, void *v) pthread_mutex_lock(&con->write_mutex); if (con->tx_buf) { - /* stage the packet; one send() per frame or per RTSP_TX_BATCH */ int ok = 1; if (con->tx_len + 4 + rtp->rtpsize > RTSP_TX_BATCH) ok = __tcp_flush(con) == SUCCESS; @@ -552,14 +544,14 @@ int rtp_send_h26x(rtsp_handle h, hal_vidstream *stream, char isH265) return ret; } -int rtp_send_pcma(rtsp_handle h, unsigned char *buf, size_t len) +static int __rtp_send_g711(rtsp_handle h, unsigned char *buf, size_t len, unsigned char pt) { /* G.711 is one byte per sample, so the 8 kHz clock advances by exactly the * number of samples in the packet. Deriving this from millis() instead let * the stamps drift away from the audio actually sent. */ - static unsigned int pcma_ts; - __frame_ts_audio = pcma_ts; - pcma_ts += (unsigned int)len; + static unsigned int g711_ts; + __frame_ts_audio = g711_ts; + g711_ts += (unsigned int)len; int ret = FAILURE; int track_id = 1; struct __transfer_set_t trans = {}; @@ -568,7 +560,7 @@ int rtp_send_pcma(rtsp_handle h, unsigned char *buf, size_t len) if (gbl_get_quit(h->pool->sharedp->gbl)) return FAILURE; - h->audioPt = 8; + h->audioPt = pt; trans.h = h; trans.track_id = track_id; @@ -578,7 +570,7 @@ int rtp_send_pcma(rtsp_handle h, unsigned char *buf, size_t len) rtsp_unlock(h); if (trans.list_head.list) { - ASSERT(__transfer_pcma(&(trans.list_head), buf, len) == SUCCESS, goto error); + ASSERT(__transfer_g711(&(trans.list_head), buf, len, pt) == SUCCESS, goto error); ASSERT(list_map_inline(&(trans.list_head), (__tcp_flush_each), NULL) == SUCCESS, goto error); ASSERT(list_map_inline(&(trans.list_head), (__rtcp_poll), &track_id) == SUCCESS, goto error); } @@ -589,6 +581,16 @@ int rtp_send_pcma(rtsp_handle h, unsigned char *buf, size_t len) return ret; } +int rtp_send_pcma(rtsp_handle h, unsigned char *buf, size_t len) +{ + return __rtp_send_g711(h, buf, len, 8); +} + +int rtp_send_pcmu(rtsp_handle h, unsigned char *buf, size_t len) +{ + return __rtp_send_g711(h, buf, len, 0); +} + int rtp_send_mp3(rtsp_handle h, unsigned char *buf, size_t len) { __frame_ts_audio = (millis() * 90) & UINT32_MAX; /* 90 kHz MPEG audio */ diff --git a/src/rtsp/rtsp_server.h b/src/rtsp/rtsp_server.h index 522aa67c..8e179601 100644 --- a/src/rtsp/rtsp_server.h +++ b/src/rtsp/rtsp_server.h @@ -33,6 +33,7 @@ void rtp_disable_audio(rtsp_handle h); int rtp_send_h26x(rtsp_handle h, hal_vidstream *stream, char isH265); int rtp_send_mp3(rtsp_handle h, unsigned char *buf, size_t len); int rtp_send_pcma(rtsp_handle h, unsigned char *buf, size_t len); +int rtp_send_pcmu(rtsp_handle h, unsigned char *buf, size_t len); extern void rtsp_finish(rtsp_handle h); diff --git a/src/server.c b/src/server.c index 400764b2..73167d29 100644 --- a/src/server.c +++ b/src/server.c @@ -10,12 +10,12 @@ IMPORT_STR(.rodata, "../res/onvif/badauth.xml", badauthxml); extern const char badauthxml[]; enum StreamType { - STREAM_H26X, - STREAM_JPEG, - STREAM_MJPEG, - STREAM_MP3, - STREAM_MP4, - STREAM_PCM + STREAM_H26X = (1 << 0), + STREAM_JPEG = (1 << 1), + STREAM_MJPEG = (1 << 2), + STREAM_MP3 = (1 << 3), + STREAM_MP4 = (1 << 4), + STREAM_PCM = (1 << 5) }; typedef struct { @@ -218,35 +218,17 @@ void send_h26x_to_client(char index, hal_vidstream *stream) { } } -/* Escape a string for use inside a JSON string literal. Usernames and codec - * names come from a request or the config file, and an unescaped quote or - * backslash makes the response unparseable for the web UI's JSON.parse(). */ -static const char *json_str(char *dst, size_t dstsz, const char *src) { - size_t o = 0; - for (; src && *src && o + 7 < dstsz; src++) { - unsigned char c = (unsigned char)*src; - if (c == '"' || c == '\\') { dst[o++] = '\\'; dst[o++] = c; } - else if (c < 0x20) o += snprintf(dst + o, dstsz - o, "\\u%04x", c); - else dst[o++] = c; - } - dst[o] = 0; - return dst; -} - -/* Whether any HTTP client is consuming the MP3 encoder's output (raw MP3 or MP4) */ -char http_audio_clients(void) { +char any_http_audio(void) { char any = 0; pthread_mutex_lock(&client_fds_mutex); for (unsigned int i = 0; i < HTTP_MAX_CLIENTS; ++i) if (client_fds[i].sockFd >= 0 && - (client_fds[i].type == STREAM_MP4 || client_fds[i].type == STREAM_MP3)) { any = 1; break; } + (client_fds[i].type & (STREAM_MP3 | STREAM_MP4 | STREAM_PCM))) { any = 1; break; } pthread_mutex_unlock(&client_fds_mutex); return any; } void send_mp4_to_client(char index, hal_vidstream *stream, char isH265) { - /* Building moof/mdat for every frame is wasted work without a consumer; - * the parameter sets are still cached so a header is ready when one connects */ char anyClient = 0; pthread_mutex_lock(&client_fds_mutex); for (unsigned int i = 0; i < HTTP_MAX_CLIENTS; ++i) @@ -1228,16 +1210,15 @@ void respond_request(http_request_t *req) { else if (EQUALS(key, "enable_auth")) app_config.rtsp_enable_auth = EQUALS_CASE(value, "true") || EQUALS(value, "1"); else if (EQUALS(key, "port")) { - int result = strtol(value, &remain, 10); - if (remain != value && result > 0 && result < 65536) + long result; + if (parse_ranged(value, 1, 65535, &result)) app_config.rtsp_port = result; } else if (EQUALS(key, "auth_user")) strncpy(app_config.rtsp_auth_user, value, sizeof(app_config.rtsp_auth_user) - 1); else if (EQUALS(key, "auth_pass")) strncpy(app_config.rtsp_auth_pass, value, sizeof(app_config.rtsp_auth_pass) - 1); else if (EQUALS(key, "audio_codec")) { - /* only the codecs the RTP path implements */ - if (EQUALS(value, "pcma") || EQUALS(value, "mp3")) { + if (EQUALS(value, "pcma") || EQUALS(value, "pcmu") || EQUALS(value, "mp3")) { strncpy(app_config.rtsp_audio_codec, value, sizeof(app_config.rtsp_audio_codec) - 1); app_config.rtsp_audio_codec[sizeof(app_config.rtsp_audio_codec) - 1] = 0; @@ -1245,8 +1226,8 @@ void respond_request(http_request_t *req) { } } } - char escUser[sizeof(app_config.rtsp_auth_user) * 6 + 1]; - char escCodec[sizeof(app_config.rtsp_audio_codec) * 6 + 1]; + char esc_user[sizeof(app_config.rtsp_auth_user) * 6 + 1]; + char esc_codec[sizeof(app_config.rtsp_audio_codec) * 6 + 1]; respLen = sprintf(response, "HTTP/1.1 200 OK\r\n" "Content-Type: application/json;charset=UTF-8\r\n" @@ -1256,8 +1237,8 @@ void respond_request(http_request_t *req) { "\"note\":\"port and codec changes apply after restart\"}", app_config.rtsp_enable ? "true" : "false", app_config.rtsp_enable_auth ? "true" : "false", app_config.rtsp_port, - json_str(escUser, sizeof(escUser), app_config.rtsp_auth_user), - json_str(escCodec, sizeof(escCodec), app_config.rtsp_audio_codec)); + escape_json(esc_user, app_config.rtsp_auth_user, sizeof(esc_user)), + escape_json(esc_codec, app_config.rtsp_audio_codec, sizeof(esc_codec))); send_and_close(req->clntFd, response, respLen); return; } @@ -1279,7 +1260,7 @@ void respond_request(http_request_t *req) { strncpy(app_config.onvif_auth_pass, value, sizeof(app_config.onvif_auth_pass) - 1); } } - char escOnvifUser[sizeof(app_config.onvif_auth_user) * 6 + 1]; + char esc_onvif_user[sizeof(app_config.onvif_auth_user) * 6 + 1]; respLen = sprintf(response, "HTTP/1.1 200 OK\r\n" "Content-Type: application/json;charset=UTF-8\r\n" @@ -1287,7 +1268,7 @@ void respond_request(http_request_t *req) { "\r\n" "{\"enable\":%s,\"enable_auth\":%s,\"auth_user\":\"%s\",\"note\":\"applies after restart\"}", app_config.onvif_enable ? "true" : "false", app_config.onvif_enable_auth ? "true" : "false", - json_str(escOnvifUser, sizeof(escOnvifUser), app_config.onvif_auth_user)); + escape_json(esc_onvif_user, app_config.onvif_auth_user, sizeof(esc_onvif_user))); send_and_close(req->clntFd, response, respLen); return; } diff --git a/src/server.h b/src/server.h index b9c0524d..af8c543e 100644 --- a/src/server.h +++ b/src/server.h @@ -21,6 +21,7 @@ #include "fmt/mp4.h" #include "fmt/nal.h" #include "hal/globals.h" +#include "hal/tools.h" #include "hal/types.h" #include "jpeg.h" #include "media.h" @@ -39,6 +40,6 @@ void send_jpeg_to_client(char index, char *buf, ssize_t size); void send_mjpeg_to_client(char index, char *buf, ssize_t size); void send_h26x_to_client(char index, hal_vidstream *stream); void send_mp3_to_client(char *buf, ssize_t size); -char http_audio_clients(void); +char any_http_audio(void); void send_mp4_to_client(char index, hal_vidstream *stream, char isH265); void send_pcm_to_client(hal_audframe *frame);