From 73468176691dd0956f5e8c5bd359326ebdcabcfd Mon Sep 17 00:00:00 2001
From: Michael Jumper <zhangmaike@users.sourceforge.net>
Date: Mon, 29 Oct 2012 10:51:56 -0700
Subject: [PATCH] Refactored audio API.

---
 protocols/rdp/guac_rdpsnd/messages.c |  6 ++++--
 protocols/rdp/include/audio.h        | 27 +++++++++++++++++++--------
 protocols/rdp/src/audio.c            | 16 ++++++++++------
 3 files changed, 33 insertions(+), 16 deletions(-)

diff --git a/protocols/rdp/guac_rdpsnd/messages.c b/protocols/rdp/guac_rdpsnd/messages.c
index d8d595d5..c4eb0651 100644
--- a/protocols/rdp/guac_rdpsnd/messages.c
+++ b/protocols/rdp/guac_rdpsnd/messages.c
@@ -181,6 +181,9 @@ void guac_rdpsnd_process_message_wave_info(guac_rdpsndPlugin* rdpsnd, audio_stre
 	rdpsnd->waveDataSize = BodySize - 8;
 	rdpsnd->expectingWave = true;
 
+    audio_stream_begin(audio, 22050, 2, 16); /* FIXME: Hard-coding rates */
+    audio_stream_write_pcm(audio, rdpsnd->waveData, 4);
+
 }
 
 /* header is not removed from data in this function */
@@ -205,8 +208,7 @@ 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);
+    /* Write rest of audio packet */
     audio_stream_write_pcm(audio, buffer, size);
     audio_stream_end(audio);
 
diff --git a/protocols/rdp/include/audio.h b/protocols/rdp/include/audio.h
index dfa2e47c..d5dd8eb1 100644
--- a/protocols/rdp/include/audio.h
+++ b/protocols/rdp/include/audio.h
@@ -134,6 +134,23 @@ struct audio_stream {
      */
     guac_stream* stream;
 
+    /**
+     * The number of samples per second of PCM data sent to this stream.
+     */
+    int rate;
+
+    /**
+     * The number of audio channels per sample of PCM data. Legal values are
+     * 1 or 2.
+     */
+    int channels;
+
+    /**
+     * The number of bits per sample per channel for PCM data. Legal values are
+     * 8 or 16.
+     */
+    int bps;
+
 };
 
 /**
@@ -150,7 +167,7 @@ void audio_stream_free(audio_stream* stream);
 /**
  * Begins a new audio stream.
  */
-void audio_stream_begin(audio_stream* stream);
+void audio_stream_begin(audio_stream* stream, int rate, int channels, int bps);
 
 /**
  * Ends the current audio stream.
@@ -172,14 +189,8 @@ 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,
+void audio_stream_write_encoded(audio_stream* audio,
         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
 
diff --git a/protocols/rdp/src/audio.c b/protocols/rdp/src/audio.c
index 131c1ca4..74d6f723 100644
--- a/protocols/rdp/src/audio.c
+++ b/protocols/rdp/src/audio.c
@@ -67,8 +67,16 @@ audio_stream* audio_stream_alloc(guac_client* client, audio_encoder* encoder) {
     return audio;
 }
 
-void audio_stream_begin(audio_stream* audio) {
+void audio_stream_begin(audio_stream* audio, int rate, int channels, int bps) {
+
+    /* Load PCM properties */
+    audio->rate = rate;
+    audio->channels = channels;
+    audio->bps = bps;
+
+    /* Call handler */
     audio->encoder->begin_handler(audio);
+
 }
 
 void audio_stream_end(audio_stream* audio) {
@@ -119,7 +127,7 @@ void audio_stream_flush(audio_stream* audio) {
 
 }
 
-void audio_stream_append_data(audio_stream* audio,
+void audio_stream_write_encoded(audio_stream* audio,
         unsigned char* data, int length) {
 
     /* Resize audio buffer if necessary */
@@ -138,7 +146,3 @@ void audio_stream_append_data(audio_stream* audio,
 
 }
 
-void audio_stream_clear_data(audio_stream* audio) {
-    audio->encoded_data_used = 0;
-}
-