chansrv: sound, add aac

2 files changed, 246 insertions, 2 deletions

diff --git a/sesman/chansrv/Makefile.am b/sesman/chansrv/Makefile.am
index 1df6c762..1e72b4ef 100644
--- a/sesman/chansrv/Makefile.am
+++ b/sesman/chansrv/Makefile.am
@@ -23,6 +23,11 @@ AM_CPPFLAGS += -DXRDP_FUSE $(FUSE_CFLAGS)
 CHANSRV_EXTRA_LIBS += $(FUSE_LIBS)
 endif
 
+if XRDP_FDK_AAC
+AM_CPPFLAGS += -DXRDP_FDK_AAC
+CHANSRV_EXTRA_LIBS += -lfdk-aac
+endif
+
 if XRDP_OPUS
 AM_CPPFLAGS += -DXRDP_OPUS
 CHANSRV_EXTRA_LIBS += -lopus
diff --git a/sesman/chansrv/sound.c b/sesman/chansrv/sound.c
index e9fcb067..5a0db94b 100644
--- a/sesman/chansrv/sound.c
+++ b/sesman/chansrv/sound.c
@@ -35,6 +35,11 @@
 #include "xrdp_sockets.h"
 #include "chansrv_common.h"
 
+#if defined(XRDP_FDK_AAC)
+#include <fdk-aac/aacenc_lib.h>
+static HANDLE_AACENCODER g_fdk_aac_encoder = 0;
+#endif
+
 #if defined(XRDP_OPUS)
 #include <opus/opus.h>
 static OpusEncoder *g_opus_encoder = 0;
@@ -114,6 +119,21 @@ static struct xr_wave_format_ex g_pcm_44100 =
     g_pcm_44100_data /* data */
 };
 
+#if defined(XRDP_FDK_AAC)
+static tui8 g_fdk_aac_44100_data[] = { 0 };
+static struct xr_wave_format_ex g_fdk_aac_44100 =
+{
+    WAVE_FORMAT_AAC_MS,  /* wFormatTag */
+    2,                   /* num of channels */
+    44100,               /* samples per sec */
+    12000,               /* avg bytes per sec */
+    4,                   /* block align */
+    16,                  /* bits per sample */
+    0,                   /* data size */
+    g_fdk_aac_44100_data /* data */
+};
+#endif
+
 #if defined(XRDP_OPUS)
 static tui8 g_opus_44100_data[] = { 0 };
 static struct xr_wave_format_ex g_opus_44100 =
@@ -148,6 +168,9 @@ static struct xr_wave_format_ex *g_wave_outp_formats[] =
 {
     &g_pcm_44100,
     &g_pcm_22050,
+#if defined(XRDP_FDK_AAC)
+    &g_fdk_aac_44100,
+#endif
 #if defined(XRDP_OPUS)
     &g_opus_44100,
 #endif
@@ -157,6 +180,9 @@ static struct xr_wave_format_ex *g_wave_outp_formats[] =
     0
 };
 
+static int g_client_does_fdk_aac = 0;
+static int g_client_fdk_aac_index = 0;
+
 static int g_client_does_opus = 0;
 static int g_client_opus_index = 0;
 
@@ -187,7 +213,7 @@ static struct xr_wave_format_ex g_pcm_inp_22050 =
 static tui8 g_pcm_inp_44100_data[] = { 0 };
 static struct xr_wave_format_ex g_pcm_inp_44100 =
 {
-    1,               /* wFormatTag */
+    WAVE_FORMAT_PCM, /* wFormatTag */
     2,               /* num of channels */
     44100,           /* samples per sec */
     176400,          /* avg bytes per sec */
@@ -370,6 +396,12 @@ sound_process_output_format(int aindex, int wFormatTag, int nChannels,
 
     switch(wFormatTag)
     {
+        case WAVE_FORMAT_AAC_MS:
+            LOG(0, ("wFormatTag, fdk aac"));
+            g_client_does_fdk_aac = 1;
+            g_client_fdk_aac_index = aindex;
+            g_bbuf_size = 4096;
+            break;
         case WAVE_FORMAT_MPEGLAYER3:
             LOG(0, ("wFormatTag, mp3"));
             g_client_does_mp3lame = 1;
@@ -437,6 +469,206 @@ sound_process_output_formats(struct stream *s, int size)
     return 0;
 }
 
+#if defined(XRDP_FDK_AAC)
+
+/*****************************************************************************/
+static int
+sound_wave_compress_fdk_aac(char *data, int data_bytes, int *format_index)
+{
+    int rv;
+    int cdata_bytes;
+    char *cdata;
+
+    AACENC_ERROR error;
+    int aot;
+    int sample_rate;
+    int mode;
+    int bitrate;
+    int afterburner;
+    int channel_order;
+    AACENC_InfoStruct info = { 0 };
+    AACENC_BufDesc in_buf = { 0 };
+    AACENC_BufDesc out_buf = { 0 };
+    AACENC_InArgs in_args = { 0 };
+    AACENC_OutArgs out_args = { 0 };
+    void *in_buffer[1];
+    int in_identifier[1];
+    int in_size[1];
+    int in_elem_size[1];
+    void *out_buffer[1];
+    int out_identifier[1];
+    int out_size[1];
+    int out_elem_size[1];
+
+    rv = data_bytes;
+
+    if (g_client_does_fdk_aac == 0)
+    {
+        return rv;
+    }
+
+    if (g_fdk_aac_encoder == 0)
+    {
+        /* init fdk aac encoder */
+        LOG(0, ("sound_wave_compress_fdk_aac: using fdk aac"));
+
+        error = aacEncOpen(&g_fdk_aac_encoder, 0, 2);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncOpen() failed"));
+            return rv;
+        }
+
+        aot = 2; /* MPEG-4 AAC Low Complexity. */
+        //aot = 129; /* MPEG-2 AAC Low Complexity. */
+        error = aacEncoder_SetParam(g_fdk_aac_encoder, AACENC_AOT, aot);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncoder_SetParam() "
+                    "AACENC_AOT failed"));
+        }
+
+        sample_rate = g_fdk_aac_44100.nSamplesPerSec;
+        error = aacEncoder_SetParam(g_fdk_aac_encoder, AACENC_SAMPLERATE,
+                                    sample_rate);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncoder_SetParam() "
+                    "AACENC_SAMPLERATE failed"));
+        }
+
+        mode = MODE_2;
+        error = aacEncoder_SetParam(g_fdk_aac_encoder,
+                                    AACENC_CHANNELMODE, mode);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncoder_SetParam() "
+                    "AACENC_CHANNELMODE failed"));
+        }
+
+        channel_order = 1; /* WAVE file format channel ordering */
+        error = aacEncoder_SetParam(g_fdk_aac_encoder, AACENC_CHANNELORDER,
+                                    channel_order);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncoder_SetParam() "
+                    "AACENC_CHANNELORDER failed"));
+        }
+
+        /* bytes rate to bit rate */
+        bitrate = g_fdk_aac_44100.nAvgBytesPerSec * 8;
+        error = aacEncoder_SetParam(g_fdk_aac_encoder, AACENC_BITRATE,
+                                    bitrate);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncoder_SetParam() "
+                    "AACENC_BITRATE failed"));
+        }
+
+        error = aacEncoder_SetParam(g_fdk_aac_encoder, AACENC_TRANSMUX, 0);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncoder_SetParam() "
+                    "AACENC_TRANSMUX failed"));
+        }
+
+        afterburner = 1;
+        error = aacEncoder_SetParam(g_fdk_aac_encoder, AACENC_AFTERBURNER,
+                                    afterburner);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncoder_SetParam() "
+                    "AACENC_AFTERBURNER failed"));
+        }
+
+        error = aacEncEncode(g_fdk_aac_encoder, NULL, NULL, NULL, NULL);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: Unable to initialize "
+                    "the encoder"));
+        }
+
+        error = aacEncInfo(g_fdk_aac_encoder, &info);
+        if (error != AACENC_OK)
+        {
+            LOG(0, ("sound_wave_compress_fdk_aac: aacEncInfo failed"));
+        }
+
+        LOG(0, ("sound_wave_compress_fdk_aac:"));
+        LOG(0, ("  AACENC_InfoStruct"));
+        LOG(0, ("    maxOutBufBytes %d", info.maxOutBufBytes));
+        LOG(0, ("    maxAncBytes %d", info.maxAncBytes));
+        LOG(0, ("    inBufFillLevel %d", info.inBufFillLevel));
+        LOG(0, ("    inputChannels %d", info.inputChannels));
+        LOG(0, ("    frameLength %d", info.frameLength));
+        LOG(0, ("    encoderDelay %d", info.encoderDelay));
+        LOG(0, ("    confBuf"));
+        LOG(0, ("    confSize %d", info.confSize));
+    }
+
+    rv = data_bytes;
+    cdata_bytes = data_bytes;
+    cdata = (char *) g_malloc(cdata_bytes, 0);
+    if (data_bytes < g_bbuf_size)
+    {
+        g_memset(data + data_bytes, 0, g_bbuf_size - data_bytes);
+        data_bytes = g_bbuf_size;
+    }
+
+    in_buffer[0] = data;
+    in_identifier[0] = IN_AUDIO_DATA;
+    in_size[0] = data_bytes;
+    in_elem_size[0] = 2;
+
+    in_args.numInSamples = data_bytes / 2;
+    in_buf.numBufs = 1;
+    in_buf.bufs = in_buffer;
+    in_buf.bufferIdentifiers = in_identifier;
+    in_buf.bufSizes = in_size;
+    in_buf.bufElSizes = in_elem_size;
+
+    out_buffer[0] = cdata;
+    out_identifier[0] = OUT_BITSTREAM_DATA;
+    out_size[0] = cdata_bytes;
+    out_elem_size[0] = 1;
+
+    out_buf.numBufs = 1;
+    out_buf.bufs = out_buffer;
+    out_buf.bufferIdentifiers = out_identifier;
+    out_buf.bufSizes = out_size;
+    out_buf.bufElSizes = out_elem_size;
+
+    error = aacEncEncode(g_fdk_aac_encoder, &in_buf, &out_buf,
+                         &in_args, &out_args);
+    if (error == AACENC_OK)
+    {
+        cdata_bytes = out_args.numOutBytes;
+        LOG(10, ("sound_wave_compress_fdk_aac: aacEncEncode ok "
+                 "cdata_bytes %d", cdata_bytes));
+        *format_index = g_client_fdk_aac_index;
+        g_memcpy(data, cdata, cdata_bytes);
+        rv = cdata_bytes;
+    }
+    else
+    {
+        LOG(0, ("sound_wave_compress_fdk_aac: aacEncEncode failed"));
+    }
+    g_free(cdata);
+
+    return rv;
+}
+
+#else
+
+/*****************************************************************************/
+static int
+sound_wave_compress_fdk_aac(char *data, int data_bytes, int *format_index)
+{
+    return data_bytes;
+}
+
+#endif
+
 #if defined(XRDP_OPUS)
 
 /*****************************************************************************/
@@ -602,7 +834,11 @@ sound_wave_compress_mp3lame(char *data, int data_bytes, int *format_index)
 static int
 sound_wave_compress(char *data, int data_bytes, int *format_index)
 {
-    if (g_client_does_opus)
+    if (g_client_does_fdk_aac)
+    {
+        return sound_wave_compress_fdk_aac(data, data_bytes, format_index);
+    }
+    else if (g_client_does_opus)
     {
         return sound_wave_compress_opus(data, data_bytes, format_index);
     }
@@ -982,6 +1218,9 @@ sound_init(void)
     /* save data from sound_server_source */
     fifo_init(&g_in_fifo, 100);
 
+    g_client_does_fdk_aac = 0;
+    g_client_fdk_aac_index = 0;
+
     g_client_does_opus = 0;
     g_client_opus_index = 0;