Initial working microphone redirection for android.

Added debug messages to alsa and pulse microphone backends.

channels/audin/client/alsa/audin_alsa.c

@@ -165,7 +165,12 @@ static BOOL audin_alsa_thread_receive(AudinALSADevice* alsa, BYTE* src, int size
 			if (WaitForSingleObject(alsa->stopEvent, 0) == WAIT_OBJECT_0)
 				break;
 			else
+			{
+				DEBUG_DVC("encoded %d [%d] to %d [%X]", alsa->buffer_frames,
+						tbytes_per_frame, encoded_size,
+						alsa->wformat);
 				ret = alsa->receive(encoded_data, encoded_size, alsa->user_data);
+			}
 
 			alsa->buffer_frames = 0;
 

channels/audin/client/opensles/audin_opensl_es.c

@@ -53,7 +53,6 @@ typedef struct _AudinOpenSLESDevice
 
 	UINT32 bytes_per_channel;
 
-	UINT32 wformat;
 	UINT32 format;
 	UINT32 block_size;
 
@@ -84,7 +83,7 @@ static void* audin_opensles_thread_func(void* arg)
 	assert(opensles->stopEvent);
 	assert(opensles->stream);
 
-	buffer.v = calloc(sizeof(short), opensles->frames_per_packet);
+	buffer.v = calloc(opensles->bytes_per_channel, opensles->frames_per_packet);
 	ZeroMemory(buffer.v, opensles->frames_per_packet);
 	freerdp_dsp_context_reset_adpcm(opensles->dsp_context);
 
@@ -98,23 +97,22 @@ static void* audin_opensles_thread_func(void* arg)
 		if (rc < 0)
 		{
 			DEBUG_WARN("android_RecIn %d", rc);
+			continue;
 		}
 
-		DEBUG_DVC("Got %d frames from microphone", opensles->frames_per_packet);
 		if (opensles->format == WAVE_FORMAT_ADPCM)
 		{
 			opensles->dsp_context->encode_ms_adpcm(opensles->dsp_context,
-				buffer.b, opensles->frames_per_packet * sizeof(short),
+				buffer.b, rc * opensles->bytes_per_channel,
 				opensles->channels, opensles->block_size);
 
 			encoded_data = opensles->dsp_context->adpcm_buffer;
 			encoded_size = opensles->dsp_context->adpcm_size;
-
 		}
 		else if (opensles->format == WAVE_FORMAT_DVI_ADPCM)
 		{
 			opensles->dsp_context->encode_ima_adpcm(opensles->dsp_context,
-				buffer.b, opensles->frames_per_packet * sizeof(short),
+				buffer.b, rc * opensles->bytes_per_channel,
 				opensles->channels, opensles->block_size);
 
 			encoded_data = opensles->dsp_context->adpcm_buffer;
@@ -123,9 +121,11 @@ static void* audin_opensles_thread_func(void* arg)
 		else
 		{
 			encoded_data = buffer.v;
-			encoded_size = opensles->frames_per_packet;
+			encoded_size = rc * opensles->bytes_per_channel;
 		}
 
+		DEBUG_DVC("encoded %d [%d] to %d [%X]", rc,
+				opensles->bytes_per_channel, encoded_size, opensles->format);
 		rc = opensles->receive(encoded_data, encoded_size, opensles->user_data);
 		if (!rc)
 			break;
@@ -172,19 +172,17 @@ static BOOL audin_opensles_format_supported(IAudinDevice* device, audinFormat* f
 
 	switch (format->wFormatTag)
 	{
-		/*
+		case WAVE_FORMAT_PCM: /* PCM */
-		case WAVE_FORMAT_PCM:
 			if (format->cbSize == 0 &&
 				(format->nSamplesPerSec <= 48000) &&
 				(format->wBitsPerSample == 8 || format->wBitsPerSample == 16) &&
-				(format->nChannels == 1 || format->nChannels == 2))
+				(format->nChannels >= 1 && format->nChannels <= 2))
 			{
 				return TRUE;
 			}
 			break;
-*/
+//		case WAVE_FORMAT_ADPCM: /* IMA ADPCM */
-		case WAVE_FORMAT_ADPCM:
+		case WAVE_FORMAT_DVI_ADPCM: 
-//		case WAVE_FORMAT_DVI_ADPCM:
 			if ((format->nSamplesPerSec <= 48000) &&
 				(format->wBitsPerSample == 4) &&
 				(format->nChannels == 1 || format->nChannels == 2))
@@ -192,6 +190,11 @@ static BOOL audin_opensles_format_supported(IAudinDevice* device, audinFormat* f
 				return TRUE;
 			}
 			break;
+		default:
+			DEBUG_DVC("Encoding '%s' [%08X] not supported",
+				rdpsnd_get_audio_tag_string(format->wFormatTag),
+				format->wFormatTag); 
+			break;
 	}
 
 	return FALSE;
@@ -220,31 +223,18 @@ static void audin_opensles_set_format(IAudinDevice* device,
 			switch (format->wBitsPerSample)
 			{
 				case 4:
-					opensles->format = WAVE_FORMAT_ADPCM;
 					opensles->bytes_per_channel = 1;
 					break;
 				case 8:
-					opensles->format = WAVE_FORMAT_PCM;
 					opensles->bytes_per_channel = 1;
 					break;
 				case 16:
-					opensles->format = WAVE_FORMAT_ADPCM;
 					opensles->bytes_per_channel = 2;
 					break;
 			}
 			break;
 		case WAVE_FORMAT_ADPCM:
-			opensles->format = WAVE_FORMAT_ADPCM;
-			opensles->bytes_per_channel = 2;
-			bs = (format->nBlockAlign - 4 * format->nChannels) * 4;
-	
-			opensles->frames_per_packet =
-				(FramesPerPacket * format->nChannels * 2 /
-				bs + 1) * bs / (format->nChannels * 2);
-			break;
-	
 		case WAVE_FORMAT_DVI_ADPCM:
-			opensles->format = WAVE_FORMAT_DVI_ADPCM;
 			opensles->bytes_per_channel = 2;
 			bs = (format->nBlockAlign - 4 * format->nChannels) * 4;
 	
@@ -252,8 +242,14 @@ static void audin_opensles_set_format(IAudinDevice* device,
 				(FramesPerPacket * format->nChannels * 2 /
 				bs + 1) * bs / (format->nChannels * 2);
 			break;
+		case WAVE_FORMAT_ALAW:
+		case WAVE_FORMAT_MULAW:
+			opensles->frames_per_packet = FramesPerPacket;
+			break;
+
 		default:
-			DEBUG_WARN("Unsupported fromat %08X requested, ignoring.",
+			DEBUG_WARN("Encoding '%s' [%08X] not supported",
+				rdpsnd_get_audio_tag_string(format->wFormatTag),
 				format->wFormatTag); 
 			return;
 	}
@@ -261,17 +257,11 @@ static void audin_opensles_set_format(IAudinDevice* device,
 	opensles->rate = format->nSamplesPerSec;
 	opensles->channels = format->nChannels;
 
-	opensles->wformat = format->wFormatTag;
+	opensles->format = format->wFormatTag;
 	opensles->block_size = format->nBlockAlign;
 
-	if (opensles->stream)
+	DEBUG_DVC("aligned frames_per_packet=%d, block_size=%d",
-	{
+			opensles->frames_per_packet, opensles->block_size);
-		android_CloseRecDevice(opensles->stream);
-
-		opensles->stream = android_OpenRecDevice(opensles->device_name,
-				opensles->rate, opensles->channels,
-				opensles->frames_per_packet);
-	}
 }
 
 static void audin_opensles_open(IAudinDevice* device, AudinReceive receive,
@@ -291,7 +281,9 @@ static void audin_opensles_open(IAudinDevice* device, AudinReceive receive,
 	opensles->stream = android_OpenRecDevice(
 			opensles->device_name,
 			opensles->rate,
-			opensles->channels, opensles->frames_per_packet);
+			opensles->channels,
+			opensles->frames_per_packet,
+			opensles->bytes_per_channel * 8);
 	assert(opensles->stream);
 
 	opensles->receive = receive;
@@ -314,7 +306,10 @@ static void audin_opensles_close(IAudinDevice* device)
 	/* The function may have been called out of order,
 	 * ignore duplicate requests. */
 	if (!opensles->stopEvent)
+	{
+		DEBUG_WARN("[ERROR] function called without matching open.");
 		return;
+	}
 
 	assert(opensles->stopEvent);
 	assert(opensles->thread);

channels/audin/client/opensles/opensl_io.c

@@ -134,9 +134,27 @@ static SLresult openSLRecOpen(OPENSL_STREAM *p){
     else
 			speakers = SL_SPEAKER_FRONT_CENTER;
     SLDataLocator_AndroidSimpleBufferQueue loc_bq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
-    SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM, channels, sr,
+    SLDataFormat_PCM format_pcm;
-				   SL_PCMSAMPLEFORMAT_FIXED_16, SL_PCMSAMPLEFORMAT_FIXED_16,
+	
-				   speakers, SL_BYTEORDER_LITTLEENDIAN};
+		format_pcm.formatType = SL_DATAFORMAT_PCM;
+		format_pcm.numChannels = channels;
+		format_pcm.samplesPerSec = sr;
+		format_pcm.channelMask = speakers;
+		format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 
+		
+		if (16 == p->bits_per_sample)
+		{
+			format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+			format_pcm.containerSize = 16;
+		}
+		else if (8 == p->bits_per_sample)
+		{
+			format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_8;
+			format_pcm.containerSize = 8;
+		}
+		else
+			assert(0);
+
     SLDataSink audioSnk = {&loc_bq, &format_pcm};
 
     // create audio recorder
@@ -211,7 +229,7 @@ static void openSLDestroyEngine(OPENSL_STREAM *p)
 
 // open the android audio device for input 
 OPENSL_STREAM *android_OpenRecDevice(char *name, int sr, int inchannels,
-		int bufferframes)
+		int bufferframes, int bits_per_sample)
 {
   
   OPENSL_STREAM *p;
@@ -221,18 +239,26 @@ OPENSL_STREAM *android_OpenRecDevice(char *name, int sr, int inchannels,
   p->inchannels = inchannels;
   p->sr = sr;
 	p->queue = Queue_New(TRUE, -1, -1);
-
-  if(openSLCreateEngine(p) != SL_RESULT_SUCCESS) {
-    android_CloseRecDevice(p);
-    return NULL;
-  }
-
-  if(openSLRecOpen(p) != SL_RESULT_SUCCESS) {
-    android_CloseRecDevice(p);
-    return NULL;
-  } 
-
 	p->buffersize = bufferframes;
+	p->bits_per_sample = bits_per_sample;
+
+	if ((p->bits_per_sample != 8) && (p->bits_per_sample != 16))
+	{
+    android_CloseRecDevice(p);
+		return NULL;
+	}
+	
+  if(openSLCreateEngine(p) != SL_RESULT_SUCCESS)
+	{
+    android_CloseRecDevice(p);
+    return NULL;
+  }
+
+  if(openSLRecOpen(p) != SL_RESULT_SUCCESS)
+	{
+    android_CloseRecDevice(p);
+    return NULL;
+  } 
 
   return p;
 }
@@ -245,12 +271,16 @@ void android_CloseRecDevice(OPENSL_STREAM *p)
   if (p == NULL)
     return;
 
+	if (p->queue)
+	{
 		while (Queue_Count(p->queue) > 0)
 		{
 			queue_element *e = Queue_Dequeue(p->queue);
 			free(e->data);
 			free(e);
 		}
+		Queue_Free(p->queue);
+	}
 
 	if (p->next)
 	{
@@ -258,7 +288,6 @@ void android_CloseRecDevice(OPENSL_STREAM *p)
 		free(p->next);
 	}
 
-	Queue_Free(p->queue);
   openSLDestroyEngine(p);
 
   free(p);
@@ -276,9 +305,9 @@ void bqRecorderCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
 	assert(p);
 	assert(p->queue);
 
-	DEBUG_DVC("Signalled");
+	printf("Signalled");
 	e = calloc(1, sizeof(queue_element));
-	e->data = calloc(p->buffersize, sizeof(short));
+	e->data = calloc(p->buffersize, p->bits_per_sample / 8);
 	e->size = p->buffersize;
 
 	if (p->next)
@@ -306,19 +335,21 @@ int android_RecIn(OPENSL_STREAM *p,short *buffer,int size)
 		bqRecorderCallback(p->recorderBufferQueue, p);
 	}
 
+	WaitForSingleObject(p->queue->event, INFINITE);
 	e = Queue_Dequeue(p->queue);
 	if (!e)
+	{
+		DEBUG_WARN("[ERROR] got e=%p from queue", e);
 		return -1;
+	}
 
-	rc = e->size;
+	rc = (e->size < size) ? e->size : size;
 	assert(p->buffersize == e->size);
 
-	memcpy(buffer, e->data, size > e->size ? e->size : size);
+	memcpy(buffer, e->data, rc * sizeof(short));
 	free(e->data);
 	free(e);
 
-	DEBUG_DVC("yay!");
+  return rc;
-
-  return size;
 }
 

channels/audin/client/opensles/opensl_io.h

@@ -64,6 +64,7 @@ typedef struct opensl_stream {
   unsigned int inchannels;
   unsigned int sr;
 	unsigned int buffersize;
+	unsigned int bits_per_sample;
 
 	wQueue *queue;
 	queue_element *next;
@@ -74,7 +75,7 @@ typedef struct opensl_stream {
   in frames. Returns a handle to the OpenSL stream
   */
   OPENSL_STREAM* android_OpenRecDevice(char *name, int sr, int inchannels,
-			int bufferframes);
+			int bufferframes, int bits_per_sample);
   /* 
   Close the audio device 
   */

channels/audin/client/pulse/audin_pulse.c

@@ -324,8 +324,6 @@ static void audin_pulse_stream_request_callback(pa_stream* stream, size_t length
 					pulse->sample_spec.channels, pulse->block_size);
 				encoded_data = pulse->dsp_context->adpcm_buffer;
 				encoded_size = pulse->dsp_context->adpcm_size;
-				DEBUG_DVC("encoded %d to %d",
-					pulse->buffer_frames * pulse->bytes_per_frame, encoded_size);
 			}
 			else
 			{
@@ -333,6 +331,9 @@ static void audin_pulse_stream_request_callback(pa_stream* stream, size_t length
 				encoded_size = pulse->buffer_frames * pulse->bytes_per_frame;
 			}
 
+			DEBUG_DVC("encoded %d [%d] to %d [%X]",
+				pulse->buffer_frames, pulse->bytes_per_frame, encoded_size,
+				pulse->format);
 			ret = pulse->receive(encoded_data, encoded_size, pulse->user_data);
 			pulse->buffer_frames = 0;
 			if (!ret)