GUACAMOLE-25: Store input and output audio format.

src/protocols/rdp/audio_input.c

@@ -118,7 +118,7 @@ static int guac_rdp_audio_parse_mimetype(const char* mimetype,
     } while (mimetype != NULL);
 
     /* Mimetype is invalid if rate was not specified */
-    if (rate == -1)
+    if (parsed_rate == -1)
         return 1;
 
     /* Parse success */
@@ -142,25 +142,20 @@ int guac_rdp_audio_handler(guac_user* user, guac_stream* stream,
 
     /* Parse mimetype, abort on parse error */
     if (guac_rdp_audio_parse_mimetype(mimetype, &rate, &channels, &bps)) {
-        guac_user_log(user, GUAC_LOG_WARN, "Denying user audio stream with "
+        guac_user_log(user, GUAC_LOG_WARNING, "Denying user audio stream with "
                 "unsupported mimetype: \"%s\"", mimetype);
         guac_protocol_send_ack(user->socket, stream, "Unsupported audio "
                 "mimetype", GUAC_PROTOCOL_STATUS_CLIENT_BAD_TYPE);
         return 0;
     }
 
-    /* FIXME: Assuming mimetype of "audio/L16;rate=44100,channels=2" */
-    else {
-        guac_user_log(user, GUAC_LOG_DEBUG, "rate=%i, channels=%i, bps=%i",
-                rate, channels, bps);
-    }
-
     /* Init stream data */
     stream->blob_handler = guac_rdp_audio_blob_handler;
     stream->end_handler = guac_rdp_audio_end_handler;
 
     /* Associate stream with audio buffer */
-    guac_rdp_audio_buffer_set_stream(rdp_client->audio_input, user, stream);
+    guac_rdp_audio_buffer_set_stream(rdp_client->audio_input, user, stream,
+            rate, channels, bps);
 
     return 0;
 
@@ -240,13 +235,16 @@ static void guac_rdp_audio_buffer_ack(guac_rdp_audio_buffer* audio_buffer,
 }
 
 void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer,
-        guac_user* user, guac_stream* stream) {
+        guac_user* user, guac_stream* stream, int rate, int channels, int bps) {
 
     pthread_mutex_lock(&(audio_buffer->lock));
 
     /* Associate received stream */
     audio_buffer->user = user;
     audio_buffer->stream = stream;
+    audio_buffer->in_rate = rate;
+    audio_buffer->in_channels = channels;
+    audio_buffer->in_bps = bps;
 
     /* Acknowledge stream creation (if buffer is ready to receive) */
     guac_rdp_audio_buffer_ack(audio_buffer,
@@ -257,13 +255,16 @@ void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer,
 }
 
 void guac_rdp_audio_buffer_begin(guac_rdp_audio_buffer* audio_buffer,
-        int packet_size, guac_rdp_audio_buffer_flush_handler* flush_handler,
+        int rate, int channels, int bps, int packet_size,
-        void* data) {
+        guac_rdp_audio_buffer_flush_handler* flush_handler, void* data) {
 
     pthread_mutex_lock(&(audio_buffer->lock));
 
     /* Reset buffer state to provided values */
     audio_buffer->bytes_written = 0;
+    audio_buffer->out_rate = rate;
+    audio_buffer->out_channels = channels;
+    audio_buffer->out_bps = bps;
     audio_buffer->packet_size = packet_size;
     audio_buffer->flush_handler = flush_handler;
     audio_buffer->data = data;
@@ -285,6 +286,8 @@ void guac_rdp_audio_buffer_write(guac_rdp_audio_buffer* audio_buffer,
 
     pthread_mutex_lock(&(audio_buffer->lock));
 
+    /* FIXME: Assuming mimetype of "audio/L16;rate=44100,channels=2" */
+
     /* Ignore packet if there is no buffer */
     if (audio_buffer->packet_size == 0 || audio_buffer->packet == NULL) {
         pthread_mutex_unlock(&(audio_buffer->lock));

src/protocols/rdp/audio_input.h

@@ -73,6 +73,45 @@ typedef struct guac_rdp_audio_buffer {
      */
     guac_stream* stream;
 
+    /**
+     * The rate of the audio stream being received from the user, if any, in
+     * samples per second. If no stream is yet associated, this value is
+     * undefined.
+     */
+    int in_rate;
+
+    /**
+     * The number of channels included in the audio stream being received from
+     * the user, if any. If no stream is yet associated, this value is
+     * undefined.
+     */
+    int in_channels;
+
+    /**
+     * The size of each sample within the audio stream being received from the
+     * user, if any, in bytes. If no stream is yet associated, this value is
+     * undefined.
+     */
+    int in_bps;
+
+    /**
+     * The rate of the audio stream expected by RDP, if any, in samples per
+     * second. If no stream is yet associated, this value is undefined.
+     */
+    int out_rate;
+
+    /**
+     * The number of channels included in the audio stream expected by RDP, if
+     * any. If no stream is yet associated, this value is undefined.
+     */
+    int out_channels;
+
+    /**
+     * The size of each sample within the audio stream expected by RDP, if any,
+     * in bytes. If no stream is yet associated, this value is undefined.
+     */
+    int out_bps;
+
     /**
      * The size that each audio packet must be, in bytes. The packet buffer
      * within this structure will be at least this size.
@@ -127,9 +166,21 @@ guac_rdp_audio_buffer* guac_rdp_audio_buffer_alloc();
  *
  * @param stream
  *     The guac_stream object representing the audio stream.
+ *
+ * @param rate
+ *     The rate of the audio stream being received from the user, if any, in
+ *     samples per second.
+ *
+ * @param channels
+ *     The number of channels included in the audio stream being received from
+ *     the user, if any.
+ *
+ * @param bps
+ *     The size of each sample within the audio stream being received from the
+ *     user, if any, in bytes.
  */
 void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer,
-        guac_user* user, guac_stream* stream);
+        guac_user* user, guac_stream* stream, int rate, int channels, int bps);
 
 /**
  * Begins handling of audio data received via guac_rdp_audio_buffer_write() and
@@ -140,6 +191,18 @@ void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer,
  * @param audio_buffer
  *     The audio buffer to begin.
  *
+ * @param rate
+ *     The rate of the audio stream expected by RDP, if any, in samples per
+ *     second.
+ *
+ * @param channels
+ *     The number of channels included in the audio stream expected by RDP, if
+ *     any.
+ *
+ * @param bps
+ *     The size of each sample within the audio stream expected by RDP, if any,
+ *     in bytes.
+ *
  * @param packet_size
  *     The number of bytes to include in all audio packets provided to the
  *     given flush_handler.
@@ -152,8 +215,8 @@ void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer,
  *     needs to be flushed.
  */
 void guac_rdp_audio_buffer_begin(guac_rdp_audio_buffer* audio_buffer,
-        int packet_size, guac_rdp_audio_buffer_flush_handler* flush_handler,
+        int rate, int channels, int bps, int packet_size,
-        void* data);
+        guac_rdp_audio_buffer_flush_handler* flush_handler, void* data);
 
 /**
  * Writes the given buffer of audio data to the given audio buffer. A new

src/protocols/rdp/guac_ai/ai_messages.c

@@ -312,7 +312,7 @@ void guac_rdp_ai_process_open(guac_client* client,
 
     /* FIXME: Assuming mimetype of 16-bit 44100 Hz stereo PCM */
     guac_rdp_audio_buffer_begin(audio_buffer, packet_frames * 2 * 2,
-            guac_rdp_ai_flush_packet, channel);
+            44100, 2, 2, guac_rdp_ai_flush_packet, channel);
 
 }