Merge "rtp_engine: rtcp_report_to_json can overflow the ssrc integer value"
[asterisk/asterisk.git] / main / audiohook.c
index ba1ecd4..04a379f 100644 (file)
@@ -29,8 +29,6 @@
 
 #include "asterisk.h"
 
-ASTERISK_REGISTER_FILE()
-
 #include <signal.h>
 
 #include "asterisk/channel.h"
@@ -45,6 +43,7 @@ ASTERISK_REGISTER_FILE()
 
 #define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */
 #define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE 100 /*!< When small queue is enabled, this is the maximum amount of audio that can remain queued at a time. */
+#define AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE 500 /*!< Otheriwise we still don't want the queue to grow indefinitely */
 
 #define DEFAULT_INTERNAL_SAMPLE_RATE 8000
 
@@ -157,6 +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
@@ -172,7 +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();
@@ -182,7 +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);
@@ -192,17 +195,10 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
                ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
                ast_slinfactory_flush(factory);
                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);
+       } else if ((our_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE)) {
+               ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
+               ast_slinfactory_flush(factory);
+               ast_slinfactory_flush(other_factory);
        }
 
        /* Write frame out to respective factory */
@@ -243,8 +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);
        }
 
@@ -253,11 +252,16 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
 
 static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples, struct ast_frame **read_reference, struct ast_frame **write_reference)
 {
-       int i = 0, usable_read, usable_write;
-       short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
+       int count;
+       int usable_read;
+       int usable_write;
+       short adjust_value;
+       short buf1[samples];
+       short buf2[samples];
+       short *read_buf = NULL;
+       short *write_buf = NULL;
        struct ast_frame frame = {
                .frametype = AST_FRAME_VOICE,
-               .data.ptr = NULL,
                .datalen = sizeof(buf1),
                .samples = samples,
        };
@@ -288,10 +292,13 @@ 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) {
-                               int count = 0;
-                               short adjust_value = abs(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) {
                                                ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
@@ -309,10 +316,13 @@ 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) {
-                               int count = 0;
-                               short adjust_value = abs(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) {
                                                ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
@@ -326,34 +336,32 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
                ast_debug(1, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
        }
 
+       frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
+
        /* Basically we figure out which buffer to use... and if mixing can be done here */
        if (read_buf && read_reference) {
-               frame.data.ptr = buf1;
+               frame.data.ptr = read_buf;
                *read_reference = ast_frdup(&frame);
        }
        if (write_buf && write_reference) {
-               frame.data.ptr = buf2;
+               frame.data.ptr = write_buf;
                *write_reference = ast_frdup(&frame);
        }
 
-       if (read_buf && write_buf) {
-               for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++) {
-                       ast_slinear_saturated_add(data1, data2);
+       /* Make the correct buffer part of the built frame, so it gets duplicated. */
+       if (read_buf) {
+               frame.data.ptr = read_buf;
+               if (write_buf) {
+                       for (count = 0; count < samples; count++) {
+                               ast_slinear_saturated_add(read_buf++, write_buf++);
+                       }
                }
-               final_buf = buf1;
-       } else if (read_buf) {
-               final_buf = buf1;
        } else if (write_buf) {
-               final_buf = buf2;
+               frame.data.ptr = write_buf;
        } else {
                return NULL;
        }
 
-       /* Make the final buffer part of the frame, so it gets duplicated fine */
-       frame.data.ptr = final_buf;
-
-       frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
-
        /* Yahoo, a combined copy of the audio! */
        return ast_frdup(&frame);
 }
@@ -823,13 +831,20 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l
                return new_frame;
        }
 
-       if (ast_format_cmp(frame->subclass.format, in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
+       if (!in_translate->format ||
+               ast_format_cmp(frame->subclass.format, in_translate->format) != AST_FORMAT_CMP_EQUAL) {
+               struct ast_trans_pvt *new_trans;
+
+               new_trans = ast_translator_build_path(slin, frame->subclass.format);
+               if (!new_trans) {
+                       return NULL;
+               }
+
                if (in_translate->trans_pvt) {
                        ast_translator_free_path(in_translate->trans_pvt);
                }
-               if (!(in_translate->trans_pvt = ast_translator_build_path(slin, frame->subclass.format))) {
-                       return NULL;
-               }
+               in_translate->trans_pvt = new_trans;
+
                ao2_replace(in_translate->format, frame->subclass.format);
        }
 
@@ -930,6 +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;
 
        /*