Fix memory safety, misleading error messages, logging correctness, and React hook bugs

include/video/mp4_writer_internal.h

@@ -66,7 +66,7 @@ bool is_audio_codec_compatible_with_mp4(enum AVCodecID codec_id, const char **co
  * @param transcoded_params Output parameter to store the transcoded codec parameters
  * @return 0 on success, negative on error
  */
-int transcode_mulaw_to_aac(const AVCodecParameters *codec_params,
+int transcode_pcm_to_aac(const AVCodecParameters *codec_params,
                            const AVRational *time_base,
                            const char *stream_name,
                            AVCodecParameters **transcoded_params);

src/video/go2rtc/go2rtc_process.c

@@ -405,6 +405,7 @@ bool go2rtc_process_init(const char *binary_path, const char *config_dir, int ap
     if (!g_config_path) {
         log_error("Memory allocation failed for config path");
         free(g_config_dir);
+        g_config_dir = NULL;
         return false;
     }
 
@@ -473,7 +474,7 @@ bool go2rtc_process_generate_config(const char *config_path, int api_port) {
     }
     FILE *config_file = fdopen(config_fd, "w");
     if (!config_file) {
-        log_error("Failed to open go2rtc config file for writing: %s", config_path);
+        log_error("Failed to create file stream for go2rtc config file: %s", config_path);
         close(config_fd);
         return false;
     }
@@ -1137,10 +1138,10 @@ bool go2rtc_process_start(int api_port) {
         }
 
         // Redirect stdout and stderr to log files
-        char log_path[1024]; // Use a reasonable fixed size instead of PATH_MAX
+        char log_path[PATH_MAX]; // Use PATH_MAX to accommodate full filesystem paths
 
         // Extract directory from g_config->log_file
-        char log_dir[1024] = {0};
+        char log_dir[PATH_MAX] = {0};
         if (g_config.log_file[0] != '\0') {
             strncpy(log_dir, g_config.log_file, sizeof(log_dir) - 1);
 
@@ -1161,11 +1162,11 @@ bool go2rtc_process_start(int api_port) {
         }
 
         // Log the path we're using for the log file
-        log_info("Using go2rtc log file: %s", log_path);
+        fprintf(stderr, "Using go2rtc log file: %s\n", log_path);
 
         int log_fd = open(log_path, O_WRONLY | O_CREAT | O_APPEND, 0644);
         if (log_fd == -1) {
-            log_error("Failed to open log file: %s", log_path);
+            fprintf(stderr, "Failed to open log file: %s\n", log_path);
             exit(EXIT_FAILURE);
         }
 

src/video/hls/hls_unified_thread.c

@@ -7,7 +7,7 @@
  *
  * This file implements the unified thread approach for HLS streaming.
  *
- * CRITICAL FIX (2025-04-11): Fixed segmentation fault issues related to thread safety
+ * CRITICAL FIX: Fixed segmentation fault issues related to thread safety
  * in the writer cleanup process. The main issue was that the hls_writer was being accessed
  * after it had been freed by another thread. The fix uses atomic operations to ensure
  * thread-safe access to the writer pointer.
@@ -2982,7 +2982,7 @@ static void ffmpeg_buffer_cleanup(void) {
 
     // CRITICAL FIX: Skip direct av_freep(NULL) call during shutdown or hard deletion
     // This is the most likely cause of segmentation faults
-    if (!is_shutdown_initiated() && !is_stream_stopping(NULL)) {
+    if (!is_shutdown_initiated()) {
         // Use a safer approach to release memory
         log_debug("Using safer approach to release FFmpeg memory");
 
@@ -2993,7 +2993,7 @@ static void ffmpeg_buffer_cleanup(void) {
             av_free(dummy);
         }
     } else {
-        log_info("Skipping av_freep(NULL) during shutdown or stream deletion to prevent segmentation fault");
+        log_info("Skipping av_freep(NULL) during shutdown to prevent segmentation fault");
     }
 
     // Call FFmpeg's internal memory cleanup functions - safely

src/video/mp4_writer_utils.c

@@ -185,7 +185,7 @@ static int init_audio_transcoder(const char *stream_name,
     strncpy(audio_transcoder_stream_names[slot], stream_name, MAX_STREAM_NAME - 1);
     audio_transcoder_stream_names[slot][MAX_STREAM_NAME - 1] = '\0';
 
-    log_info("Successfully initialized audio transcoder from μ-law to AAC for %s", stream_name);
+    log_info("Successfully initialized audio transcoder from PCM to AAC for %s", stream_name);
 
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(59, 37, 100)
     log_info("Sample rate: %d, Channels: %d, Bit rate: %ld",
@@ -395,7 +395,7 @@ int transcode_audio_packet(const char *stream_name,
  * @param transcoded_params Output parameter to store the transcoded codec parameters
  * @return 0 on success, negative on error
  */
-int transcode_mulaw_to_aac(const AVCodecParameters *codec_params,
+int transcode_pcm_to_aac(const AVCodecParameters *codec_params,
                                  const AVRational *time_base,
                                  const char *stream_name,
                                  AVCodecParameters **transcoded_params) {
@@ -456,7 +456,7 @@ int transcode_mulaw_to_aac(const AVCodecParameters *codec_params,
     // Open decoder
     ret = avcodec_open2(decoder_ctx, decoder, NULL);
     if (ret < 0) {
-        log_ffmpeg_error(ret, "Failed to open μ-law decoder");
+        log_ffmpeg_error(ret, "Failed to open PCM decoder");
         goto cleanup;
     }
 
@@ -505,7 +505,7 @@ int transcode_mulaw_to_aac(const AVCodecParameters *codec_params,
         goto cleanup;
     }
 
-    log_info("Successfully configured transcoding from μ-law to AAC for %s",
+    log_info("Successfully configured transcoding from PCM to AAC for %s",
             stream_name ? stream_name : "unknown");
 
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(59, 37, 100)
@@ -871,7 +871,7 @@ int mp4_writer_initialize(mp4_writer_t *writer, const AVPacket *pkt, const AVStr
 
                 // Try to transcode PCM to AAC
                 AVCodecParameters *transcoded_params = NULL;
-                int transcode_ret = transcode_mulaw_to_aac(input_stream->codecpar,
+                int transcode_ret = transcode_pcm_to_aac(input_stream->codecpar,
                                                          &input_stream->time_base,
                                                          writer->stream_name,
                                                          &transcoded_params);
@@ -1290,7 +1290,7 @@ int mp4_writer_add_audio_stream(mp4_writer_t *writer, const AVCodecParameters *c
 
             // Try to transcode PCM to AAC
             AVCodecParameters *transcoded_params = NULL;
-            int transcode_ret = transcode_mulaw_to_aac(codec_params, &safe_time_base,
+            int transcode_ret = transcode_pcm_to_aac(codec_params, &safe_time_base,
                                                      writer->stream_name, &transcoded_params);
 
             if (transcode_ret >= 0 && transcoded_params) {

src/video/onvif_device_management.c

@@ -87,7 +87,8 @@ static char* create_security_header(const char *username, const char *password,
     memcpy(concatenated, nonce_bytes, nonce_len);
     // Raw byte copies for SHA-1 input: intermediate parts are not C strings
     memcpy((void *)(concatenated + nonce_len), created, created_len); // NOLINT(bugprone-not-null-terminated-result)
-    memcpy((void *)(concatenated + nonce_len + created_len), password, password_len + 1);
+    // Exclude the null terminator: WS-Security spec requires raw bytes only
+    memcpy((void *)(concatenated + nonce_len + created_len), password, password_len);
 
     // Calculate SHA1 digest
     mbedtls_sha1((unsigned char*)concatenated, nonce_len + created_len + password_len, digest);

web/js/components/preact/WebRTCView.jsx

@@ -184,8 +184,11 @@ export function WebRTCView() {
     setIsLoading(isLoadingStreams);
   }, [isLoadingStreams]);
 
-  // Process streams data when it's loaded or when the selected stream changes.
+  // Process streams data when it's loaded.
   useEffect(() => {
+    // If a stream is already selected (e.g. by the user or URL), don't override it here.
+    if (selectedStream) return;
+
     if (streamsData && Array.isArray(streamsData)) {
       // Process the streams data
       const processStreams = async () => {
@@ -226,9 +229,9 @@ export function WebRTCView() {
 
       processStreams();
     }
-    // Note: This effect now explicitly depends on selectedStream.
-    // It is written to be idempotent and will only update selectedStream when necessary.
-  }, [streamsData, selectedStream]);
+    // Note: This effect now explicitly depends on streamsData and autoGrid.
+    // It will only update selectedStream when none is already selected.
+  }, [streamsData, autoGrid]);
 
   // Sync layout/page/stream to URL — only meaningful once streams are loaded.
   useEffect(() => {