First import of HTTP Streamer functionality in the daemon used to rec…

…eive AES67 audio streams via HTTP file streaming. The HTTP Streamer can be enabled via the _streamer_enabled_ daemon parameter. When the Streamer is active the daemon starts capturing the configured _Sinks_ up to the maximum number of channels configured by the _streamer_channels_ parameters. The captured PCM samples are split into _streamer_files_num_ files of _streamer_file_duration_ duration (in seconds) for each sink, compressed using AAC LC codec and served via HTTP. The HTTP streamer requires the libfaac-dev package to compile. Please note that since the HTTP Streamer uses the RAVENNA ALSA device for capturing it's not possible to use such device for other audio captures.
bondagit · Jul 6, 2024 · 466f6b4 · 466f6b4
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diff --git a/README.md b/README.md
@@ -38,6 +38,7 @@ The daemon uses the following open source:
 * **cpp-httplib** licensed under the [MIT License](https://github.com/yhirose/cpp-httplib/blob/master/LICENSE)
 * **Avahi common & client libraries** licensed under the [LGPL License](https://github.com/lathiat/avahi/blob/master/LICENSE)
 * **Boost libraries** licensed under the [Boost Software License](https://www.boost.org/LICENSE_1_0.txt)
+* **Freeware Advanced Audio Coder** licensed under the [LGPL License](https://github.com/knik0/faac?tab=License-1-ov-file)
 
 ## Prerequisite ##
 <a name="prerequisite"></a>
@@ -48,6 +49,7 @@ The daemon and the test have been verified starting from **Ubuntu 18.04** distro
 * cmake version >= 3.7
 * boost libraries version >= 1.65
 * Avahi service discovery (if enabled) >= 0.7
+* Freeware Advanced Audio Coder (if streamer enabled) libfaac >= 1.30
 
 The following platforms have been used for testing:
 
@@ -104,9 +106,21 @@ The daemon should work on all Ubuntu starting from 18.04 onward, it's possible t
 ## Devices and interoperability tests ##
 See [Devices and interoperability tests with the AES67 daemon](DEVICES.md)
 
+## HTTP Streamer ##
+The HTTP Streamer was introduced with the daemon version 2.0 and it is used to receive AES67 audio streams via HTTP file streaming.
+
+The HTTP Streamer can be enabled via the _streamer_enabled_ daemon parameter.
+When the Streamer is active the daemon starts capturing the configured _Sinks_ up to the maximum number of channels configured by the _streamer_channels_ parameters.
+The captured PCM samples are split into _streamer_files_num_ files of _streamer_file_duration_ duration (in seconds) for each sink, compressed using AAC LC codec and served via HTTP.
+
+The HTTP streamer requires the libfaac-dev package to compile.
+
+Please note that since the HTTP Streamer uses the RAVENNA ALSA device for capturing it's not possible to use such device for other audio captures.
+
 ## AES67 USB Receiver and Transmitter ##
 See [Use your board as AES67 USB Receiver and Transmitter](USB_GADGET.md)
 
+
 ## Repository content ##
 
 ### [daemon](daemon) directory ###

diff --git a/daemon/CMakeLists.txt b/daemon/CMakeLists.txt
@@ -24,6 +24,9 @@ if (NOT CPP_HTTPLIB_DIR)
     find_path( CPP_HTTPLIB_DIR "httplib.h" REQUIRED)
 endif()
 
+find_library(ALSA_LIBRARY NAMES asound)
+find_library(AAC_LIBRARY NAMES faac)
+
 find_library(AVAHI_LIBRARY-COMMON NAMES avahi-common)
 find_library(AVAHI_LIBRARY-CLIENT NAMES avahi-client)
 find_path(AVAHI_INCLUDE_DIR avahi-client/publish.h)
@@ -34,7 +37,7 @@ find_package(Boost COMPONENTS system thread log program_options REQUIRED)
 include_directories(aes67-daemon ${RAVENNA_ALSA_LKM_DIR}/common ${RAVENNA_ALSA_LKM_DIR}/driver ${CPP_HTTPLIB_DIR} ${Boost_INCLUDE_DIR})
 add_definitions( -DBOOST_LOG_DYN_LINK -DBOOST_LOG_USE_NATIVE_SYSLOG )
 add_compile_options( -Wall )
-set(SOURCES error_code.cpp json.cpp main.cpp session_manager.cpp http_server.cpp config.cpp interface.cpp log.cpp sap.cpp browser.cpp rtsp_client.cpp mdns_client.cpp mdns_server.cpp rtsp_server.cpp utils.cpp)
+set(SOURCES error_code.cpp json.cpp main.cpp session_manager.cpp http_server.cpp config.cpp interface.cpp log.cpp sap.cpp browser.cpp rtsp_client.cpp mdns_client.cpp mdns_server.cpp rtsp_server.cpp utils.cpp streamer.cpp)
 
 if(FAKE_DRIVER)
   MESSAGE(STATUS "FAKE_DRIVER")
@@ -49,7 +52,7 @@ if(ENABLE_TESTS)
     add_subdirectory(tests)
 endif()
 
-target_link_libraries(aes67-daemon ${Boost_LIBRARIES})
+target_link_libraries(aes67-daemon ${Boost_LIBRARIES} ${ALSA_LIBRARY} ${AAC_LIBRARY})
 if(WITH_AVAHI)
   MESSAGE(STATUS "WITH_AVAHI")
   add_definitions(-D_USE_AVAHI_)

diff --git a/daemon/README.md b/daemon/README.md
@@ -148,13 +148,31 @@ In case of failure the server returns a **text/plain** content type with the cat
 * **Body type** application/json    
 * **Body** [RTP Remote Sources params](#rtp-remote-sources)
 
+### Get streamer info for a Sink ###
+* **Description** retrieve the streamer info for the specified Sink
+* **URL** /api/streamer/info/:id
+* **Method** GET
+* **URL Params** id=[integer in the range (0-63)]
+* **Body Type** application/json
+* **Body** [Streamer info params](#streamer-info)
+
+### Get streamer AAC audio file ###
+* **Description** retrieve the AAC audio frames for the specified Sink and file id
+* **URL** /api/streamer/streamer/:sinkId/:fileId
+* **Method** GET
+* **URL Params** sinkId=[integer in the range (0-63)], fileId=[integer in the range (0-*streamer_files_num*)]
+* **HTTP headers** the headers _X-File-Count_, _X-File-Current-Id_, _X-File-Start-Id_ return the current global file count, the current file id and the start file id for the file returned
+* **Body Type** audio/aac
+* **Body** Binary body containing ADTS AAC LC audio frames
+
+
 ## HTTP REST API structures ##
 
 ### JSON Version<a name="version"></a> ###
 
 Example
     {
-      "version:" "bondagit-1.5"
+      "version:" "bondagit-2.0"
     }
 
 where:
@@ -189,7 +207,12 @@ Example
       "node_id": "AES67 daemon d9aca383",
       "custom_node_id": "",
       "ptp_status_script": "./scripts/ptp_status.sh",
-      "auto_sinks_update": true
+      "auto_sinks_update": true,
+      "streamer_enabled": false,
+      "streamer_channels": 8,
+      "streamer_files_num": 6,
+      "streamer_file_duration": 1,
+      "streamer_player_buffer_files_num": 1
     }
 
 where:
@@ -284,6 +307,23 @@ where:
 > JSON string specifying the path to the script executed in background when the PTP slave clock status changes.
 > The PTP clock status is passed as first parameter to the script and it can be *unlocked*, *locking* or *locked*.
 
+> **streamer\_enabled**
+> JSON boolean specifying whether the HTTP Streamer is enabled or disabled.
+> Once activated, the HTTP Streamer starts capturing samples for number of channels specified by *streamer_channels* starting from channel 0, then it splits them into *streamer_files_num* files of a *streamer_file_duration* duration for each configured Sink and it serves them via HTTP.
+
+> **streamer\_channels**
+> JSON number specifying the number of channels captured by the HTTP Streamer starting from channel 0, 8 by default.
+
+> **streamer\_files\_num**
+> JSON number specifying the number of files into which the stream gets split.
+
+> **streamer\_file\_duration**
+> JSON number specifying the maximum duration of each streamer file in seconds.
+
+> **streamer\_player\_buffer\_files\_num**
+> JSON number specifying the player buffer in number of files.
+
+
 ### JSON PTP Config<a name="ptp-config"></a> ###
 
 Example
@@ -658,3 +698,55 @@ where:
 > JSON number specifying the meausured period in seconds between the last source announcements.
 > A remote source is automatically removed if it doesn't get announced for **announce\_period** x 10 seconds.
 
+### JSON Streamer info<a name="streamer-info"></a> ###
+
+Example:
+
+    {
+       "status": 0,
+       "file_duration": 1,
+       "files_num": 8,
+       "player_buffer_files_num": 1,
+       "start_file_id": 3,
+       "current_file_id": 0,
+       "channels": 2,
+       "format": "s16",
+       "rate": 48000
+    }
+
+where:
+
+> **status**
+> JSON number containing the streamer status code.
+> Status is 0 in case the streamer is able to provide the audio samples, othrewise the specific error code is returned.
+> 0 - OK
+> 1 - PTP clock not locked
+> 2 - Channel/s not captured
+> 3 - Buffering
+> 4 - Streamer not enabled
+> 5 - Invalid Sink
+> 6 - Cannot retrieve Sink
+
+> **file_duration_sec**
+> JSON number specifying the duration of each file.
+
+> **files_num**
+> JSON number specifying the number of files. The streamer will use these files as a circular buffer.
+
+> **start_file_id**
+> JSON number specifying the file id to use to start the playback.
+
+> **current_file_id**
+> JSON number specifying the file id that is beeing created by the daemon.
+
+> **player_buffer_files_num**
+> JSON number specifying the number of files to use for buffering.
+
+> **channels**
+> JSON number specifying the number of channels of the stream.
+
+> **format**
+> JSON string specifying the PCM encoding of the AAC compressed stream.
+
+> **rate**
+> JSON number specifying the sample rate of the stream.
diff --git a/daemon/browser.hpp b/daemon/browser.hpp
@@ -100,7 +100,8 @@ class Browser : public MDNSClient {
 
   SAP sap_{config_->get_sap_mcast_addr()};
   IGMP igmp_;
-  std::chrono::time_point<std::chrono::steady_clock> startup_{std::chrono::steady_clock::now()};
+  std::chrono::time_point<std::chrono::steady_clock> startup_{
+      std::chrono::steady_clock::now()};
   uint32_t last_update_{0}; /* seconds from daemon startup */
 };
 

diff --git a/daemon/config.cpp b/daemon/config.cpp
@@ -66,6 +66,15 @@ std::shared_ptr<Config> Config::parse(const std::string& filename,
     config.max_tic_frame_size_ = 1024;
   if (config.sample_rate_ == 0)
     config.sample_rate_ = 48000;
+  if (config.streamer_channels_ < 2 || config.streamer_channels_ > 16)
+    config.streamer_channels_ = 8;
+  if (config.streamer_file_duration_ < 1 || config.streamer_file_duration_ > 4)
+    config.streamer_file_duration_ = 1;
+  if (config.streamer_files_num_ < 4 || config.streamer_files_num_ > 16)
+    config.streamer_files_num_ = 8;
+  if (config.streamer_player_buffer_files_num_ < 1 || config.streamer_player_buffer_files_num_ > 2)
+    config.streamer_player_buffer_files_num_ = 1;
+
   boost::system::error_code ec;
   ip::address_v4::from_string(config.rtp_mcast_base_.c_str(), ec);
   if (ec) {
@@ -126,16 +135,22 @@ bool Config::save(const Config& config) {
         get_max_tic_frame_size() != config.get_max_tic_frame_size() ||
         get_interface_name() != config.get_interface_name();
 
-    daemon_restart_ = driver_restart_ ||
-                      get_http_port() != config.get_http_port() ||
-                      get_rtsp_port() != config.get_rtsp_port() ||
-                      get_http_base_dir() != config.get_http_base_dir() ||
-                      get_rtp_mcast_base() != config.get_rtp_mcast_base() ||
-                      get_sap_mcast_addr() != config.get_sap_mcast_addr() ||
-                      get_rtp_port() != config.get_rtp_port() ||
-                      get_status_file() != config.get_status_file() ||
-                      get_mdns_enabled() != config.get_mdns_enabled() ||
-                      get_custom_node_id() != config.get_custom_node_id();
+    daemon_restart_ =
+        driver_restart_ || get_http_port() != config.get_http_port() ||
+        get_rtsp_port() != config.get_rtsp_port() ||
+        get_http_base_dir() != config.get_http_base_dir() ||
+        get_rtp_mcast_base() != config.get_rtp_mcast_base() ||
+        get_sap_mcast_addr() != config.get_sap_mcast_addr() ||
+        get_rtp_port() != config.get_rtp_port() ||
+        get_status_file() != config.get_status_file() ||
+        get_mdns_enabled() != config.get_mdns_enabled() ||
+        get_custom_node_id() != config.get_custom_node_id() ||
+        get_streamer_channels() != config.get_streamer_channels() ||
+        get_streamer_file_duration() != config.get_streamer_file_duration() ||
+        get_streamer_files_num() != config.get_streamer_files_num() ||
+        get_streamer_player_buffer_files_num() !=
+            config.get_streamer_player_buffer_files_num() ||
+        get_streamer_enabled() != config.get_streamer_enabled();
 
     if (!daemon_restart_)
       *this = config;

diff --git a/daemon/config.hpp b/daemon/config.hpp
@@ -37,6 +37,15 @@ class Config {
   uint16_t get_http_port() const { return http_port_; };
   uint16_t get_rtsp_port() const { return rtsp_port_; };
   const std::string& get_http_base_dir() const { return http_base_dir_; };
+  uint8_t get_streamer_files_num() const { return streamer_files_num_; };
+  uint16_t get_streamer_file_duration() const {
+    return streamer_file_duration_;
+  };
+  uint8_t get_streamer_player_buffer_files_num() const {
+    return streamer_player_buffer_files_num_;
+  };
+  uint8_t get_streamer_channels() const { return streamer_channels_; };
+  bool get_streamer_enabled() const { return streamer_enabled_; };
   int get_log_severity() const { return log_severity_; };
   uint32_t get_playout_delay() const { return playout_delay_; };
   uint32_t get_tic_frame_size_at_1fs() const { return tic_frame_size_at_1fs_; };
@@ -78,6 +87,22 @@ class Config {
   void set_http_base_dir(std::string_view http_base_dir) {
     http_base_dir_ = http_base_dir;
   };
+  void set_streamer_channels(uint8_t streamer_channels) {
+    streamer_channels_ = streamer_channels;
+  };
+  void set_streamer_files_num(uint8_t streamer_files_num) {
+    streamer_files_num_ = streamer_files_num;
+  };
+  void set_streamer_file_duration(uint16_t streamer_file_duration) {
+    streamer_file_duration_ = streamer_file_duration;
+  };
+  void set_streamer_player_buffer_files_num(
+      uint8_t streamer_player_buffer_files_num) {
+    streamer_player_buffer_files_num_ = streamer_player_buffer_files_num;
+  };
+  void set_streamer_enabled(uint8_t streamer_enabled) {
+    streamer_enabled_ = streamer_enabled;
+  };
   void set_log_severity(int log_severity) { log_severity_ = log_severity; };
   void set_playout_delay(uint32_t playout_delay) {
     playout_delay_ = playout_delay;
@@ -137,6 +162,13 @@ class Config {
            lhs.get_http_port() != rhs.get_http_port() ||
            lhs.get_rtsp_port() != rhs.get_rtsp_port() ||
            lhs.get_http_base_dir() != rhs.get_http_base_dir() ||
+           lhs.get_streamer_channels() != rhs.get_streamer_channels() ||
+           lhs.get_streamer_files_num() != rhs.get_streamer_files_num() ||
+           lhs.get_streamer_file_duration() !=
+               rhs.get_streamer_file_duration() ||
+           lhs.get_streamer_player_buffer_files_num() !=
+               rhs.get_streamer_player_buffer_files_num() ||
+           lhs.get_streamer_enabled() != rhs.get_streamer_enabled() ||
            lhs.get_log_severity() != rhs.get_log_severity() ||
            lhs.get_playout_delay() != rhs.get_playout_delay() ||
            lhs.get_tic_frame_size_at_1fs() != rhs.get_tic_frame_size_at_1fs() ||
@@ -166,6 +198,11 @@ class Config {
   uint16_t http_port_{8080};
   uint16_t rtsp_port_{8854};
   std::string http_base_dir_{"../webui/dist"};
+  uint8_t streamer_channels_{8};
+  uint8_t streamer_files_num_{8};
+  uint16_t streamer_file_duration_{1};
+  uint8_t streamer_player_buffer_files_num_{1};
+  bool streamer_enabled_{false};
   int log_severity_{2};
   uint32_t playout_delay_{0};
   uint32_t tic_frame_size_at_1fs_{48};

diff --git a/daemon/daemon.conf b/daemon/daemon.conf
@@ -20,5 +20,10 @@
   "mdns_enabled": true,
   "custom_node_id": "",
   "ptp_status_script": "./scripts/ptp_status.sh",
+  "streamer_channels": 8,
+  "streamer_files_num": 8,
+  "streamer_file_duration": 1,
+  "streamer_player_buffer_files_num": 1,
+  "streamer_enabled": true,
   "auto_sinks_update": true
 }
diff --git a/daemon/error_code.cpp b/daemon/error_code.cpp
@@ -117,6 +117,12 @@ std::string DaemonErrCategory::message(int ev) const {
       return "failed to receive event from driver";
     case DaemonErrc::invalid_driver_response:
       return "unexpected driver command response code";
+    case DaemonErrc::streamer_invalid_ch:
+      return "sink channel not captured";
+    case DaemonErrc::streamer_retry_later:
+      return "not enough samples buffered, retry later";
+    case DaemonErrc::streamer_not_running:
+      return "not running, check PTP lock";
     default:
       return "(unrecognized daemon error)";
   }

diff --git a/daemon/error_code.hpp b/daemon/error_code.hpp
@@ -53,12 +53,15 @@ enum class DaemonErrc {
   cannot_parse_sdp = 45,      // daemon cannot parse SDP
   stream_name_in_use = 46,    // daemon source or sink name in use
   cannot_retrieve_mac = 47,   // daemon cannot retrieve MAC for IP
-  send_invalid_size = 50,     // daemon data size too big for buffer
-  send_u2k_failed = 51,       // daemon failed to send command to driver
-  send_k2u_failed = 52,       // daemon failed to send event response to driver
-  receive_u2k_failed = 53,    // daemon failed to receive response from driver
-  receive_k2u_failed = 54,    // daemon failed to receive event from driver
-  invalid_driver_response = 55  // unexpected driver command response code
+  streamer_invalid_ch = 48,   // daemon streamer sink channel not captured
+  streamer_retry_later = 49,  // daemon streamer not enough samples buffered
+  streamer_not_running = 50,  // daemon streamer not running
+  send_invalid_size = 60,     // daemon data size too big for buffer
+  send_u2k_failed = 61,       // daemon failed to send command to driver
+  send_k2u_failed = 62,       // daemon failed to send event response to driver
+  receive_u2k_failed = 63,    // daemon failed to receive response from driver
+  receive_k2u_failed = 64,    // daemon failed to receive event from driver
+  invalid_driver_response = 65  // unexpected driver command response code
 };
 
 namespace std {