audiohook.c: Improve frame pairing logic to avoid MixMonitor breakage with mixed...

This patch adjusts the read/write synchronization logic in audiohook_read_frame_both()
to better handle calls where participants use different codecs or sample sizes
(e.g., alaw vs G.722). The previous hard threshold of 2 * samples caused MixMonitor
recordings to break or stutter when frames were not aligned between both directions.

The new logic uses a more tolerant limit (1.5 * samples), which prevents audio tearing
without causing excessive buffer overruns. This fix specifically addresses issues
with MixMonitor when recording directly on a channel in a bridge using mixed codecs.

Reported-by: Michal Hajek <michal.hajek@daktela.com>
Resolves: #1276
Resolves: #1279

diff --git a/main/audiohook.c b/main/audiohook.c
index 86136c22dd016b568ec5e68e78f751fb801a3ef7..aabe8629c89086896b094cd268e28f2bb50f3719 100644 (file)
--- a/main/audiohook.c
+++ b/main/audiohook.c
@@ -308,14 +308,23 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
         *    2 * samples) according to actual needs, for example, setting it to (ast_tvdiff_ms(ast_tvnow(),
         *    audiohook->write_time) < (samples/8)*4) && (ast_slinfactory_available(&audiohook->read_factory)
         *    < 4 * samples).
+        *
+        *    Update:
+        *       Increased time and sample thresholds allow for better handling of asymmetric streams
+        *       (e.g., mixed codecs like alaw and G.722) and high RTT conditions.
+        *       This avoids premature frame reads when one direction is delayed, which can cause
+        *       audio tearing or broken recordings.
+        *       Specifically addresses issues with MixMonitor when recording directly on a channel
+        *       that is part of a bridge with different sample rates or codecs.
+        *       A slight overrun in recording duration is acceptable in exchange for audio stability.
         */
-       if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2) && (ast_slinfactory_available(&audiohook->read_factory) < 2 * samples)) {
+       if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*4) && (ast_slinfactory_available(&audiohook->read_factory) < 4 * samples)) {
                ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
                return NULL;
        }
 
        /* As shown in the above comment. */
-       if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2) && (ast_slinfactory_available(&audiohook->write_factory) < 2 * samples)) {
+       if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*4) && (ast_slinfactory_available(&audiohook->write_factory) < 4 * samples)) {
                ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
                return NULL;
        }