Merge 'develop' for release 2.2.0

Merge 'develop' for release 2.2.0:
- refactor align forward-stream logic with improvements from the original mod_audio_stream repo (#24)
- refactor api command cli helper (#24)
- fix both safety issues and performance improvements in stream_frame (#24) 
- fix avoid per-frame heap allocations of support buffers #24
- update docs references for project rename to mod_openai_realtime

Author: dariopellegrino00

CMakeLists.txt

@@ -2,7 +2,7 @@ cmake_minimum_required(VERSION 3.18)
 project(mod_openai_audio_stream
         VERSION 1.0.0
         DESCRIPTION "Audio streaming module for FreeSWITCH."
-        HOMEPAGE_URL "https://github.com/VoiSmart/mod_openai_audio_stream") 
+        HOMEPAGE_URL "https://github.com/VoiSmart/mod_openai_realtime") 
 
 include(GNUInstallDirs)
 
@@ -34,7 +34,13 @@ if(CMAKE_BUILD_TYPE MATCHES "Release")
     set_target_properties(ixwebsocket PROPERTIES LINK_FLAGS_RELEASE "-s -w") #-static-libgcc -static-libstdc++
 endif()
 
-add_library(mod_openai_audio_stream SHARED mod_openai_audio_stream.c mod_openai_audio_stream.h openai_audio_streamer_glue.h openai_audio_streamer_glue.cpp buffer/ringbuffer.c base64.cpp)
+add_library(mod_openai_audio_stream SHARED
+    mod_openai_audio_stream.c
+    mod_openai_audio_stream.h
+    openai_audio_streamer_glue.h
+    openai_audio_streamer_glue.cpp
+    base64.cpp
+)
 
 set_property(TARGET mod_openai_audio_stream PROPERTY POSITION_INDEPENDENT_CODE ON)
 

README.md

@@ -1,14 +1,18 @@
-# mod_openai_audio_stream
+# mod_openai_realtime 
 
-![Build](https://github.com/VoiSmart/mod_openai_audio_stream/actions/workflows/build.yml/badge.svg?branch=main)
-![Code-Checks](https://github.com/VoiSmart/mod_openai_audio_stream/actions/workflows/code-checks.yml/badge.svg?branch=main)
+![Build](https://github.com/VoiSmart/mod_openai_realtime/actions/workflows/build.yml/badge.svg?branch=main)
+![Code-Checks](https://github.com/VoiSmart/mod_openai_realtime/actions/workflows/code-checks.yml/badge.svg?branch=main)
 [![License: MIT](https://img.shields.io/badge/license-MIT-blue?style=flat)](LICENSE)
 
-**mod_openai_audio_stream** is a FreeSWITCH module that streams L16 audio from a channel to an OpenAI Realtime WebSocket endpoint. The stream follows OpenAI's Realtime API specification and enables real-time audio playback directly in the channel.  
+**mod_openai_realtime** is a FreeSWITCH module that streams L16 audio from a channel to an OpenAI Realtime WebSocket endpoint. The stream follows OpenAI's Realtime API specification and enables real-time audio playback directly in the channel.  
+
+> [!WARNING]
+> This is a standalone fork of `mod_audio_stream`, not affiliated with the original project.
+> Legacy naming (`mod_openai_audio_stream`) is retained for backward compatibility but will be updated in a future major release.
 
 It is a fork of [mod_audio_stream](https://github.com/amigniter/mod_audio_stream), specifically adapted for streaming audio to OpenAI's Realtime API and playing the responses back to the user via FreeSWITCH and WebSocket.  
 
-The goal of **mod_openai_audio_stream** is to provide a simple, lightweight, yet effective module for streaming audio and receiving responses directly from OpenAI’s Realtime WebSocket into the call through FreeSWITCH. It uses [ixwebsocket](https://machinezone.github.io/IXWebSocket/), a C++ WebSocket library compiled as a static library.  
+The goal of **mod_openai_realtime** is to provide a simple, lightweight, yet effective module for streaming audio and receiving responses directly from OpenAI’s Realtime WebSocket into the call through FreeSWITCH. It uses [ixwebsocket](https://machinezone.github.io/IXWebSocket/), a C++ WebSocket library compiled as a static library.  
 
 
 ## Important Notes 
@@ -76,7 +80,7 @@ The following is **a simple dialplan example** that demonstrates how to use the
 
 * Make sure to replace `sk-xxxxxxxxxxxxxxxxxx` with your actual OpenAI API key.
 * The dialplan answers the call and starts streaming audio to OpenAI's Realtime API using `uuid_openai_audio_stream`, so you can try it out and see the OpenAI events in the FreeSWITCH console within the `mod_openai_audio_stream::json` events and other module events.
-* The playback action with `silence_stream://-1//` is needed for audio playback to work properly. For more details, check issue [#16](https://github.com/VoiSmart/mod_openai_audio_stream/issues/16).
+* The playback action with `silence_stream://-1//` is needed for audio playback to work properly. For more details, check issue [#16](https://github.com/VoiSmart/mod_openai_realtime/issues/16).
 
 #### Next steps
 
@@ -85,7 +89,7 @@ The **getting started** example is a basic demonstration of how to use the modul
 This way you can build more complex applications **allowing for function calls, updating instructions**, and other interactions with OpenAI's Realtime API. Check out the [OpenAI Realtime documentation](https://platform.openai.com/docs/guides/realtime) and [API reference](https://platform.openai.com/docs/api-reference/realtime) for more details on how to structure your requests and handle responses.
 
 ### Channel variables
-The following channel variables can be used to fine tune websocket connection and also configure mod_openai_audio_stream logging:
+The following channel variables can be used to fine-tune websocket connection and also configure mod_openai_realtime logging:
 
 | Variable                               | Description                                             | Default |
 | -------------------------------------- | ------------------------------------------------------- | ------- |

buffer/ringbuffer.c

@@ -170,12 +170,12 @@ static switch_status_t send_json(switch_core_session_t *session, char* json) {
 #define STREAM_API_SYNTAX \
 "USAGE:\n" \
 "--------------------------------------------------------------------------------\n" \
-"uuid_openai_audio_stream <uuid> [start | stop | send_json | pause | resume |\n" \
-"                                 mute | unmute]\n" \
-"                                [wss-url | path | user | openai | all | base64json]\n" \
-"                                [mono | mixed | stereo]\n" \
-"                                [8000 | 16000 | 24000]\n" \
-"                                [mute_user]\n" \
+"uuid_openai_audio_stream <uuid> start <wss-url> <mono | mixed | stereo>\n" \
+"                        [8k | 16k | 24k | <other rate>] [mute_user]\n" \
+"                        where <rate> = 8k|16k|24k or any multiple of 8000 (default: 8k)\n" \
+"uuid_openai_audio_stream <uuid> [stop | pause | resume]\n" \
+"uuid_openai_audio_stream <uuid> [mute | unmute] [user | openai | all]\n" \
+"uuid_openai_audio_stream <uuid> send_json <base64json>\n" \
 "--------------------------------------------------------------------------------\n"
 
 typedef enum {

buffer/ringbuffer.h

@@ -4,7 +4,6 @@
 #include <switch.h>
 #include <limits.h>
 #include <speex/speex_resampler.h>
-#include "buffer/ringbuffer.h"
 
 #define MY_BUG_NAME "audio_stream"
 #define MAX_SESSION_ID (256)
@@ -33,11 +32,10 @@ struct private_data {
     int user_audio_muted:1;
     int openai_audio_muted:1;
     int close_requested:1;
-    RingBuffer *buffer;
     switch_buffer_t *sbuffer;
-    uint8_t *data;
     int rtp_packets;
     switch_buffer_t *playback_buffer;
+    void *stream_buffers;
 };
 
 typedef struct private_data private_t;