diff --git a/src/protocols/rdp/audio_input.c b/src/protocols/rdp/audio_input.c
index 9e88fb5c..0eb757d8 100644
--- a/src/protocols/rdp/audio_input.c
+++ b/src/protocols/rdp/audio_input.c
@@ -250,28 +250,49 @@ void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer,
     guac_rdp_audio_buffer_ack(audio_buffer,
             "OK", GUAC_PROTOCOL_STATUS_SUCCESS);
 
+    guac_user_log(user, GUAC_LOG_DEBUG, "User is requesting to provide audio "
+            "input as %i-channel, %i Hz PCM audio at %i bytes/sample.",
+            audio_buffer->in_format.channels,
+            audio_buffer->in_format.rate,
+            audio_buffer->in_format.bps);
+
+    pthread_mutex_unlock(&(audio_buffer->lock));
+
+}
+
+void guac_rdp_audio_buffer_set_output(guac_rdp_audio_buffer* audio_buffer,
+        int rate, int channels, int bps) {
+
+    pthread_mutex_lock(&(audio_buffer->lock));
+
+    /* Set output format */
+    audio_buffer->out_format.rate = rate;
+    audio_buffer->out_format.channels = channels;
+    audio_buffer->out_format.bps = bps;
+
     pthread_mutex_unlock(&(audio_buffer->lock));
 
 }
 
 void guac_rdp_audio_buffer_begin(guac_rdp_audio_buffer* audio_buffer,
-        int rate, int channels, int bps, int packet_size,
-        guac_rdp_audio_buffer_flush_handler* flush_handler, void* data) {
+        int packet_frames, 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_format.rate = rate;
-    audio_buffer->out_format.channels = channels;
-    audio_buffer->out_format.bps = bps;
-    audio_buffer->packet_size = packet_size;
     audio_buffer->flush_handler = flush_handler;
     audio_buffer->data = data;
 
+    /* Calculate size of each packet in bytes */
+    audio_buffer->packet_size = packet_frames
+                              * audio_buffer->out_format.channels
+                              * audio_buffer->out_format.bps;
+
     /* Allocate new buffer */
     free(audio_buffer->packet);
-    audio_buffer->packet = malloc(packet_size);
+    audio_buffer->packet = malloc(audio_buffer->packet_size);
 
     /* Acknowledge stream creation (if stream is ready to receive) */
     guac_rdp_audio_buffer_ack(audio_buffer,
diff --git a/src/protocols/rdp/audio_input.h b/src/protocols/rdp/audio_input.h
index 0970671b..87eee9a9 100644
--- a/src/protocols/rdp/audio_input.h
+++ b/src/protocols/rdp/audio_input.h
@@ -180,29 +180,43 @@ guac_rdp_audio_buffer* guac_rdp_audio_buffer_alloc();
 void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer,
         guac_user* user, guac_stream* stream, int rate, int channels, int bps);
 
+/**
+ * Defines the output format that should be used by the audio buffer when
+ * flushing packets of audio data received via guac_rdp_audio_buffer_write().
+ * As this format determines how the underlying packet buffer will be
+ * allocated, this function MUST be called prior to the call to
+ * guac_rdp_audio_buffer_begin().
+ *
+ * @param audio_buffer
+ *     The audio buffer to set the output format of.
+ *
+ * @param rate
+ *     The rate of the audio stream expected by RDP, in samples per second.
+ *
+ * @param channels
+ *     The number of channels included in the audio stream expected by RDP.
+ *
+ * @param bps
+ *     The size of each sample within the audio stream expected by RDP, in
+ *     bytes.
+ */
+void guac_rdp_audio_buffer_set_output(guac_rdp_audio_buffer* audio_buffer,
+        int rate, int channels, int bps);
+
 /**
  * Begins handling of audio data received via guac_rdp_audio_buffer_write() and
- * allocates the necessary underlying packet buffer. Audio packets of exactly
- * packet_size bytes will be flushed as available using the provided
- * flush_handler.
+ * allocates the necessary underlying packet buffer. Audio packets having
+ * exactly packet_frames frames will be flushed as available using the provided
+ * flush_handler. An audio frame is a set of single samples, one sample per
+ * channel. The guac_rdp_audio_buffer_set_output() function MUST have
+ * been invoked first.
  *
  * @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
+ * @param packet_frames
+ *     The exact number of frames (a set of samples, one for each channel)
+ *     which MUST be included in all audio packets provided to the
  *     given flush_handler.
  *
  * @param flush_handler
@@ -213,8 +227,8 @@ 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 rate, int channels, int bps, int packet_size,
-        guac_rdp_audio_buffer_flush_handler* flush_handler, void* data);
+        int packet_frames, guac_rdp_audio_buffer_flush_handler* flush_handler,
+        void* data);
 
 /**
  * Writes the given buffer of audio data to the given audio buffer. A new
diff --git a/src/protocols/rdp/guac_ai/ai_messages.c b/src/protocols/rdp/guac_ai/ai_messages.c
index eeb3e287..c3465878 100644
--- a/src/protocols/rdp/guac_ai/ai_messages.c
+++ b/src/protocols/rdp/guac_ai/ai_messages.c
@@ -251,6 +251,9 @@ void guac_rdp_ai_process_version(guac_client* client,
 void guac_rdp_ai_process_formats(guac_client* client,
         IWTSVirtualChannel* channel, wStream* stream) {
 
+    guac_rdp_client* rdp_client = (guac_rdp_client*) client->data;
+    guac_rdp_audio_buffer* audio_buffer = rdp_client->audio_input;
+
     UINT32 num_formats;
     Stream_Read_UINT32(stream, num_formats); /* NumFormats */
     Stream_Seek_UINT32(stream); /* cbSizeFormatsPacket (MUST BE IGNORED) */
@@ -265,6 +268,10 @@ void guac_rdp_ai_process_formats(guac_client* client,
         if (format.tag != GUAC_RDP_WAVE_FORMAT_PCM)
             continue;
 
+        /* Set output format of internal audio buffer to match RDP server */
+        guac_rdp_audio_buffer_set_output(audio_buffer, format.rate,
+                format.channels, format.bps / 8);
+
         /* Accept single format */
         guac_rdp_ai_send_incoming_data(channel);
         guac_rdp_ai_send_formats(channel, &format, 1);
@@ -301,18 +308,19 @@ void guac_rdp_ai_process_open(guac_client* client,
     Stream_Read_UINT32(stream, packet_frames); /* FramesPerPacket */
     Stream_Read_UINT32(stream, initial_format); /* InitialFormat */
 
-    /* STUB */
-    guac_client_log(client, GUAC_LOG_DEBUG, "AUDIO_INPUT: open: "
-            "packet_frames=%i, initial_format=%i",
-            packet_frames, initial_format);
+    guac_client_log(client, GUAC_LOG_DEBUG, "RDP server is accepting audio "
+            "input as %i-channel, %i Hz PCM audio at %i bytes/sample.",
+            audio_buffer->out_format.channels,
+            audio_buffer->out_format.rate,
+            audio_buffer->out_format.bps);
 
     /* Success */
     guac_rdp_ai_send_formatchange(channel, initial_format);
     guac_rdp_ai_send_open_reply(channel, 0);
 
-    /* FIXME: Assuming mimetype of 16-bit 44100 Hz stereo PCM */
-    guac_rdp_audio_buffer_begin(audio_buffer, packet_frames * 2 * 2,
-            44100, 2, 2, guac_rdp_ai_flush_packet, channel);
+    /* Begin receiving audio data */
+    guac_rdp_audio_buffer_begin(audio_buffer, packet_frames,
+            guac_rdp_ai_flush_packet, channel);
 
 }