Fix realloc, add convenience buffer to audio stream.

protocols/rdp/Makefile.am

@@ -57,7 +57,10 @@ libguac_client_rdp_la_SOURCES = \
 	src/rdp_keymap_en_us.c \
 	src/default_pointer.c
 
-guac_rdpsnd_la_SOURCES = guac_rdpsnd/messages.c guac_rdpsnd/service.c
+guac_rdpsnd_la_SOURCES =   \
+	src/audio.c            \
+	guac_rdpsnd/messages.c \
+	guac_rdpsnd/service.c
 
 noinst_HEADERS =              \
 	include/audio.h           \

protocols/rdp/guac_rdpsnd/messages.c

@@ -205,6 +205,11 @@ void rdpsnd_process_message_wave(guac_rdpsndPlugin* rdpsnd,
 
     guac_client_log_info(audio->client, "Got sound: %i bytes.", size);
 
+    /* For now, start AND end stream right here */
+    audio_stream_begin(audio);
+    audio_stream_write_pcm(audio, buffer, size);
+    audio_stream_end(audio);
+
 	output_stream = stream_new(8);
 	stream_write_uint8(output_stream, SNDC_WAVECONFIRM);
 	stream_write_uint8(output_stream, 0);

protocols/rdp/include/audio.h

@@ -57,7 +57,7 @@ typedef void audio_encoder_end_handler(audio_stream* audio);
  * Handler which is called when the audio stream is flushed.
  */
 typedef void audio_encoder_write_handler(audio_stream* audio,
-        int* pcm_data, int length);
+        unsigned char* pcm_data, int length);
 
 /**
  * Arbitrary audio codec encoder.
@@ -89,23 +89,38 @@ typedef struct audio_encoder {
 struct audio_stream {
 
     /**
-     * PCM data buffer, 16-bit samples, 2-channel.
+     * PCM data buffer, 16-bit samples, 2-channel, 44100 Hz.
      */
-    int* pcm_data;
+    unsigned char* pcm_data;
 
     /**
-     * Number of samples in buffer.
+     * Number of bytes in buffer.
      */
     int used;
 
     /**
-     * Maximum number of samples in buffer.
+     * Maximum number of bytes in buffer.
      */
     int length;
 
     /**
-     * Arbitrary codec encoder. When the PCM buffer is flushed, PCM data will be sent
-     * to this encoder.
+     * Encoded audio data buffer, as written by the encoder.
+     */
+    unsigned char* encoded_data;
+
+    /**
+     * Number of bytes in the encoded data buffer.
+     */
+    int encoded_data_used;
+
+    /**
+     * Maximum number of bytes in the encoded data buffer.
+     */
+    int encoded_data_length;
+
+    /**
+     * Arbitrary codec encoder. When the PCM buffer is flushed, PCM data will
+     * be sent to this encoder.
      */
     audio_encoder* encoder;
 
@@ -145,12 +160,26 @@ void audio_stream_end(audio_stream* stream);
 /**
  * Writes PCM data to the given audio stream.
  */
-void audio_stream_write_pcm(audio_stream* stream, const int* data, int length);
+void audio_stream_write_pcm(audio_stream* stream,
+        unsigned char* data, int length);
 
 /**
  * Flushes the given audio stream.
  */
 void audio_stream_flush(audio_stream* stream);
 
+/**
+ * Appends arbitrarily-encoded data to the encoded_data buffer
+ * within the given audio stream.
+ */
+void audio_stream_append_data(audio_stream* stream,
+        unsigned char* data, int length);
+
+/**
+ * Clears all data from the encoded_data buffer in the given
+ * audio stream.
+ */
+void audio_stream_clear_data(audio_stream* stream);
+
 #endif
 

protocols/rdp/src/audio.c

@@ -53,8 +53,12 @@ audio_stream* audio_stream_alloc(guac_client* client, audio_encoder* encoder) {
     audio->used = 0;
     audio->length = 0x40000;
 
-    /* Allocate buffer */
-    audio->pcm_data = malloc(sizeof(int) * audio->length);
+    audio->encoded_data_used = 0;
+    audio->encoded_data_length = 0x40000;
+
+    /* Allocate bufferis */
+    audio->pcm_data = malloc(audio->length);
+    audio->encoded_data = malloc(audio->encoded_data_length);
 
     /* Assign encoder */
     audio->encoder = encoder;
@@ -77,14 +81,15 @@ void audio_stream_free(audio_stream* audio) {
     free(audio);
 }
 
-void audio_stream_write_pcm(audio_stream* audio, const int* data, int length) {
+void audio_stream_write_pcm(audio_stream* audio, 
+        unsigned char* data, int length) {
 
     /* Resize audio buffer if necessary */
     if (length > audio->length) {
 
         /* Resize to double provided length */
         audio->length = length*2;
-        audio->pcm_data = realloc(audio, sizeof(int) * audio->length);
+        audio->pcm_data = realloc(audio->pcm_data, audio->length);
 
     }
 
@@ -93,7 +98,7 @@ void audio_stream_write_pcm(audio_stream* audio, const int* data, int length) {
         audio_stream_flush(audio);
 
     /* Append to buffer */
-    memcpy(&(audio->pcm_data[audio->used]), data, sizeof(int) * length);
+    memcpy(&(audio->pcm_data[audio->used]), data, length);
     audio->used += length;
 
 }
@@ -114,3 +119,26 @@ void audio_stream_flush(audio_stream* audio) {
 
 }
 
+void audio_stream_append_data(audio_stream* audio,
+        unsigned char* data, int length) {
+
+    /* Resize audio buffer if necessary */
+    if (audio->encoded_data_used + length > audio->encoded_data_length) {
+
+        /* Increase to double concatenated size to accomodate */
+        audio->encoded_data_length = (audio->encoded_data_length + length)*2;
+        audio->encoded_data = realloc(audio->encoded_data,
+                audio->encoded_data_length);
+
+    }
+
+    /* Append to buffer */
+    memcpy(&(audio->encoded_data[audio->encoded_data_used]), data, length);
+    audio->encoded_data_used += length;
+
+}
+
+void audio_stream_clear_data(audio_stream* audio) {
+    audio->encoded_data_used = 0;
+}
+

protocols/rdp/src/ogg_encoder.c

@@ -54,7 +54,8 @@ void ogg_encoder_end_handler(audio_stream* audio) {
 
 }
 
-void ogg_encoder_write_handler(audio_stream* audio, int* pcm_data, int length) {
+void ogg_encoder_write_handler(audio_stream* audio, 
+        unsigned char* pcm_data, int length) {
 
     guac_client_log_info(audio->client,
             "OGG: Writing data to stream, length=%i",