Initial OMX Support

src/audio.h

@@ -31,3 +31,6 @@ extern AUDIO_RENDERER_CALLBACKS audio_callbacks_sdl;
 extern AUDIO_RENDERER_CALLBACKS audio_callbacks_pulse;
 bool audio_pulse_init();
 #endif
+#ifdef HAVE_PI
+extern AUDIO_RENDERER_CALLBACKS audio_callbacks_omx;
+#endif

src/audio/alsa.c

@@ -28,8 +28,6 @@
 #define MAX_CHANNEL_COUNT 6
 #define FRAME_SIZE 240
 
-const char* audio_device = "sysdefault";
-
 static snd_pcm_t *handle;
 static OpusMSDecoder* decoder;
 static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];

src/audio/omx.c

@@ -0,0 +1,205 @@
+/*
+ * This file is part of Moonlight Embedded.
+ *
+ * Copyright (C) 2015 Iwan Timmer
+ *
+ * Moonlight is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * Moonlight is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with Moonlight; if not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "../audio.h"
+
+#include <stdio.h>
+
+#include <opus_multistream.h>
+#include "bcm_host.h"
+#include "ilclient.h"
+
+#define MAX_CHANNEL_COUNT 6
+#define FRAME_SIZE 240
+
+static OpusMSDecoder* decoder;
+ILCLIENT_T* handle;
+COMPONENT_T* component;
+static OMX_BUFFERHEADERTYPE *buf;
+static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
+static int channelCount;
+
+static void omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
+  int rc, error;
+  OMX_ERRORTYPE err;
+  unsigned char omxMapping[6];
+  char* componentName = "audio_render";
+
+  channelCount = opusConfig->channelCount;
+  /* The supplied mapping array has order: FL-FR-C-LFE-RL-RR
+   * OMX expects the order: FL-FR-LFE-C-RL-RR
+   * We need copy the mapping locally and swap the channels around.
+   */
+  memcpy(omxMapping, opusConfig->mapping, sizeof(omxMapping));
+  if (opusConfig->channelCount > 2) {
+    omxMapping[2] = opusConfig->mapping[3];
+    omxMapping[3] = opusConfig->mapping[2];
+  }
+
+  decoder = opus_multistream_decoder_create(opusConfig->sampleRate,
+                                            opusConfig->channelCount,
+                                            opusConfig->streams,
+                                            opusConfig->coupledStreams,
+                                            omxMapping,
+                                            &rc);
+
+  handle = ilclient_init();
+  if (handle == NULL) {
+  	fprintf(stderr, "IL client init failed\n");
+  	exit(1);
+  }
+
+  if (ilclient_create_component(handle, &component, componentName, ILCLIENT_DISABLE_ALL_PORTS | ILCLIENT_ENABLE_INPUT_BUFFERS) != 0) {
+    fprintf(stderr, "Component create failed\n");
+    exit(1);
+  }
+
+  if (ilclient_change_component_state(component, OMX_StateIdle)!= 0) {
+    fprintf(stderr, "Couldn't change state to Idle\n");
+    exit(1);
+  }
+
+  // must be before we enable buffers
+  OMX_AUDIO_PARAM_PORTFORMATTYPE audioPortFormat;
+  memset(&audioPortFormat, 0, sizeof(OMX_AUDIO_PARAM_PORTFORMATTYPE));
+  audioPortFormat.nSize = sizeof(OMX_AUDIO_PARAM_PORTFORMATTYPE);
+  audioPortFormat.nVersion.nVersion = OMX_VERSION;
+
+  audioPortFormat.nPortIndex = 100;
+
+  OMX_GetParameter(ilclient_get_handle(component), OMX_IndexParamAudioPortFormat, &audioPortFormat);
+
+  audioPortFormat.eEncoding = OMX_AUDIO_CodingPCM;
+  OMX_SetParameter(ilclient_get_handle(component), OMX_IndexParamAudioPortFormat, &audioPortFormat);
+
+  OMX_AUDIO_PARAM_PCMMODETYPE sPCMMode;
+
+  memset(&sPCMMode, 0, sizeof(OMX_AUDIO_PARAM_PCMMODETYPE));
+  sPCMMode.nSize = sizeof(OMX_AUDIO_PARAM_PCMMODETYPE);
+  sPCMMode.nVersion.nVersion = OMX_VERSION;
+  sPCMMode.nPortIndex = 100;
+  sPCMMode.nChannels = channelCount;
+  sPCMMode.eNumData = OMX_NumericalDataSigned;
+  sPCMMode.eEndian = OMX_EndianLittle;
+  sPCMMode.nSamplingRate = opusConfig->sampleRate;
+  sPCMMode.bInterleaved = OMX_TRUE;
+  sPCMMode.nBitPerSample = 16;
+  sPCMMode.ePCMMode = OMX_AUDIO_PCMModeLinear;
+
+  switch(channelCount) {
+  case 1:
+     sPCMMode.eChannelMapping[0] = OMX_AUDIO_ChannelCF;
+     break;
+  case 8:
+     sPCMMode.eChannelMapping[7] = OMX_AUDIO_ChannelRS;
+  case 7:
+     sPCMMode.eChannelMapping[6] = OMX_AUDIO_ChannelLS;
+  case 6:
+     sPCMMode.eChannelMapping[5] = OMX_AUDIO_ChannelRR;
+  case 5:
+     sPCMMode.eChannelMapping[4] = OMX_AUDIO_ChannelLR;
+  case 4:
+     sPCMMode.eChannelMapping[3] = OMX_AUDIO_ChannelLFE;
+  case 3:
+     sPCMMode.eChannelMapping[2] = OMX_AUDIO_ChannelCF;
+  case 2:
+     sPCMMode.eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+     sPCMMode.eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+     break;
+  }
+
+  err = OMX_SetParameter(ilclient_get_handle(component), OMX_IndexParamAudioPcm, &sPCMMode);
+  if(err != OMX_ErrorNone){
+  	fprintf(stderr, "PCM mode unsupported\n");
+  	return;
+  }
+  OMX_CONFIG_BRCMAUDIODESTINATIONTYPE arDest;
+
+  if (audio_device && strlen(audio_device) < sizeof(arDest.sName)) {
+    memset(&arDest, 0, sizeof(OMX_CONFIG_BRCMAUDIODESTINATIONTYPE));
+    arDest.nSize = sizeof(OMX_CONFIG_BRCMAUDIODESTINATIONTYPE);
+    arDest.nVersion.nVersion = OMX_VERSION;
+
+    strcpy((char *)arDest.sName, audio_device);
+
+    err = OMX_SetParameter(ilclient_get_handle(component), OMX_IndexConfigBrcmAudioDestination, &arDest);
+    if (err != OMX_ErrorNone) {
+      fprintf(stderr, "Error on setting audio destination\nomx option must be set to hdmi or local\n");
+      exit(1);
+    }
+  }
+
+  // input port
+  ilclient_enable_port_buffers(component, 100, NULL, NULL, NULL);
+  ilclient_enable_port(component, 100);
+
+  err = ilclient_change_component_state(component, OMX_StateExecuting);
+  if (err < 0) {
+  	fprintf(stderr, "Couldn't change state to Executing\n");
+  	exit(1);
+  }
+}
+
+static void omx_renderer_cleanup() {
+  if (decoder != NULL)
+    opus_multistream_decoder_destroy(decoder);
+  if (handle != NULL) {
+    if((buf = ilclient_get_input_buffer(component, 100, 1)) == NULL){
+      fprintf(stderr, "Can't get audio buffer\n");
+      exit(EXIT_FAILURE);
+    }
+
+    buf->nFilledLen = 0;
+    buf->nFlags = OMX_BUFFERFLAG_TIME_UNKNOWN | OMX_BUFFERFLAG_EOS;
+
+    if(OMX_EmptyThisBuffer(ILC_GET_HANDLE(component), buf) != OMX_ErrorNone){
+      fprintf(stderr, "Can't empty audio buffer\n");
+      return;
+    }
+
+    ilclient_disable_port_buffers(component, 100, NULL, NULL, NULL);
+    ilclient_change_component_state(component, OMX_StateIdle);
+    ilclient_change_component_state(component, OMX_StateLoaded);
+  }
+}
+
+static void omx_renderer_decode_and_play_sample(char* data, int length) {
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
+  if (decodeLen > 0) {
+	buf = ilclient_get_input_buffer(component, 100, 1);
+    buf->nOffset = 0;
+    buf->nFlags = OMX_BUFFERFLAG_TIME_UNKNOWN;
+	int bufLength = decodeLen * sizeof(short) * channelCount;
+    memcpy(buf->pBuffer, pcmBuffer, bufLength);
+    buf->nFilledLen = bufLength;
+    int r = OMX_EmptyThisBuffer(ilclient_get_handle(component), buf);
+    if (r != OMX_ErrorNone) {
+    fprintf(stderr, "Empty buffer error\n");
+    } 
+  } else {
+    printf("Opus error from decode: %d\n", decodeLen);
+  }
+}
+
+AUDIO_RENDERER_CALLBACKS audio_callbacks_omx = {
+  .init = omx_renderer_init,
+  .cleanup = omx_renderer_cleanup,
+  .decodeAndPlaySample = omx_renderer_decode_and_play_sample,
+  .capabilities = CAPABILITY_DIRECT_SUBMIT,
+};

src/config.c

@@ -38,6 +38,7 @@
 
 bool inputAdded = false;
 static bool mapped = true;
+const char* audio_device = "sysdefault";
 
 static struct option long_options[] = {
   {"720", no_argument, NULL, 'a'},

src/main.c

@@ -152,7 +152,7 @@ static void help() {
   printf("\n I/O options\n\n");
   printf("\t-mapping <file>\t\tUse <file> as gamepad mapping configuration file (use before -input)\n");
   printf("\t-input <device>\t\tUse <device> as input. Can be used multiple times\n");
-  printf("\t-audio <device>\t\tUse <device> as ALSA audio output device (default sysdefault)\n");
+  printf("\t-audio <device>\t\tUse <device> as audio output device (default sysdefault)\n");
   printf("\t-forcehw \t\tTry to use video hardware acceleration\n");
   #endif
   printf("\nUse Ctrl+Alt+Shift+Q to exit streaming session\n\n");

src/platform.c

@@ -99,6 +99,10 @@ AUDIO_RENDERER_CALLBACKS* platform_get_audio(enum platform system) {
   case SDL:
     return &audio_callbacks_sdl;
   #endif
+  #ifdef HAVE_PI
+  case PI:
+    return (PAUDIO_RENDERER_CALLBACKS) dlsym(RTLD_DEFAULT, "audio_callbacks_omx");
+  #endif
   default:
     #ifdef HAVE_PULSE
     if (audio_pulse_init())