Merge "rtp_engine: rtcp_report_to_json can overflow the ssrc integer value"
[asterisk/asterisk.git] / main / audiohook.c
index 8a0055e..04a379f 100644 (file)
@@ -156,6 +156,11 @@ int ast_audiohook_destroy(struct ast_audiohook *audiohook)
        return 0;
 }
 
+#define SHOULD_MUTE(hook, dir) \
+       ((ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ) && (dir == AST_AUDIOHOOK_DIRECTION_READ)) || \
+       (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_WRITE) && (dir == AST_AUDIOHOOK_DIRECTION_WRITE)) || \
+       (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE)))
+
 /*! \brief Writes a frame into the audiohook structure
  * \param audiohook Audiohook structure
  * \param direction Direction the audio frame came from
@@ -171,7 +176,6 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
        int our_factory_ms;
        int other_factory_samples;
        int other_factory_ms;
-       int muteme = 0;
 
        /* Update last feeding time to be current */
        *rwtime = ast_tvnow();
@@ -181,7 +185,7 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
        other_factory_samples = ast_slinfactory_available(other_factory);
        other_factory_ms = other_factory_samples / (audiohook->hook_internal_samp_rate / 1000);
 
-       if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
+       if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
                ast_debug(1, "Flushing audiohook %p so it remains in sync\n", audiohook);
                ast_slinfactory_flush(factory);
                ast_slinfactory_flush(other_factory);
@@ -197,17 +201,6 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
                ast_slinfactory_flush(other_factory);
        }
 
-       /* swap frame data for zeros if mute is required */
-       if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) ||
-               (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) ||
-               (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) {
-                       muteme = 1;
-       }
-
-       if (muteme && frame->datalen > 0) {
-               ast_frame_clear(frame);
-       }
-
        /* Write frame out to respective factory */
        ast_slinfactory_feed(factory, frame);
 
@@ -246,8 +239,11 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
                return NULL;
        }
 
-       /* If a volume adjustment needs to be applied apply it */
-       if (vol) {
+       if (SHOULD_MUTE(audiohook, direction)) {
+               /* Swap frame data for zeros if mute is required */
+               ast_frame_clear(&frame);
+       } else if (vol) {
+               /* If a volume adjustment needs to be applied apply it */
                ast_frame_adjust_volume(&frame, vol);
        }
 
@@ -296,8 +292,12 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
        if (usable_read) {
                if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
                        read_buf = buf1;
-                       /* Adjust read volume if need be */
-                       if (audiohook->options.read_volume) {
+
+                       if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ))) {
+                               /* Clear the frame data if we are muting */
+                               memset(buf1, 0, sizeof(buf1));
+                       } else if (audiohook->options.read_volume) {
+                               /* Adjust read volume if need be */
                                adjust_value = abs(audiohook->options.read_volume);
                                for (count = 0; count < samples; count++) {
                                        if (audiohook->options.read_volume > 0) {
@@ -316,8 +316,12 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
        if (usable_write) {
                if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
                        write_buf = buf2;
-                       /* Adjust write volume if need be */
-                       if (audiohook->options.write_volume) {
+
+                       if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE))) {
+                               /* Clear the frame data if we are muting */
+                               memset(buf2, 0, sizeof(buf2));
+                       } else if (audiohook->options.write_volume) {
+                               /* Adjust write volume if need be */
                                adjust_value = abs(audiohook->options.write_volume);
                                for (count = 0; count < samples; count++) {
                                        if (audiohook->options.write_volume > 0) {
@@ -941,6 +945,17 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
        if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
                return frame;
        }
+
+       /* If the translation resulted in an interpolated frame then immediately return as audiohooks
+        * rely on actual media being present to do things.
+        */
+       if (!middle_frame->data.ptr) {
+               if (middle_frame != start_frame) {
+                       ast_frfree(middle_frame);
+               }
+               return start_frame;
+       }
+
        samples = middle_frame->samples;
 
        /*