Milestone 5: deliver embedded RDP sessions and lifecycle hardening

third_party/FreeRDP/channels/rdpsnd/client/opensles/CMakeLists.txt

@@ -0,0 +1,29 @@
+# FreeRDP: A Remote Desktop Protocol Implementation
+# FreeRDP cmake build script
+#
+# Copyright 2013 Armin Novak <armin.novak@gmail.com>
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+define_channel_client_subsystem("rdpsnd" "opensles" "")
+
+find_package(OpenSLES REQUIRED)
+
+set(${MODULE_PREFIX}_SRCS opensl_io.c rdpsnd_opensles.c)
+
+set(${MODULE_PREFIX}_LIBS winpr freerdp ${OpenSLES_LIBRARIES})
+
+include_directories(..)
+include_directories(SYSTEM ${OpenSLES_INCLUDE_DIRS})
+
+add_channel_client_subsystem_library(${MODULE_PREFIX} ${MODULE_NAME} ${CHANNEL_NAME} "" TRUE "")

third_party/FreeRDP/channels/rdpsnd/client/opensles/opensl_io.c

@@ -0,0 +1,422 @@
+/*
+opensl_io.c:
+Android OpenSL input/output module
+Copyright (c) 2012, Victor Lazzarini
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are met:
+    * Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+    * Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+    * Neither the name of the <organization> nor the
+      names of its contributors may be used to endorse or promote products
+      derived from this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+DISCLAIMED. IN NO EVENT SHALL <COPYRIGHT HOLDER> BE LIABLE FOR ANY
+DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <winpr/assert.h>
+
+#include "rdpsnd_main.h"
+#include "opensl_io.h"
+#define CONV16BIT 32768
+#define CONVMYFLT (1. / 32768.)
+
+static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void* context);
+
+// creates the OpenSL ES audio engine
+static SLresult openSLCreateEngine(OPENSL_STREAM* p)
+{
+	SLresult result;
+	// create engine
+	result = slCreateEngine(&(p->engineObject), 0, nullptr, 0, nullptr, nullptr);
+	DEBUG_SND("engineObject=%p", (void*)p->engineObject);
+
+	if (result != SL_RESULT_SUCCESS)
+		goto engine_end;
+
+	// realize the engine
+	result = (*p->engineObject)->Realize(p->engineObject, SL_BOOLEAN_FALSE);
+	DEBUG_SND("Realize=%" PRIu32 "", result);
+
+	if (result != SL_RESULT_SUCCESS)
+		goto engine_end;
+
+	// get the engine interface, which is needed in order to create other objects
+	result = (*p->engineObject)->GetInterface(p->engineObject, SL_IID_ENGINE, &(p->engineEngine));
+	DEBUG_SND("engineEngine=%p", (void*)p->engineEngine);
+
+	if (result != SL_RESULT_SUCCESS)
+		goto engine_end;
+
+engine_end:
+	return result;
+}
+
+// opens the OpenSL ES device for output
+static SLresult openSLPlayOpen(OPENSL_STREAM* p)
+{
+	SLresult result;
+	SLuint32 sr = p->sr;
+	SLuint32 channels = p->outchannels;
+	WINPR_ASSERT(p->engineObject);
+	WINPR_ASSERT(p->engineEngine);
+
+	if (channels)
+	{
+		// configure audio source
+		SLDataLocator_AndroidSimpleBufferQueue loc_bufq = { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
+			                                                p->queuesize };
+
+		switch (sr)
+		{
+			case 8000:
+				sr = SL_SAMPLINGRATE_8;
+				break;
+
+			case 11025:
+				sr = SL_SAMPLINGRATE_11_025;
+				break;
+
+			case 16000:
+				sr = SL_SAMPLINGRATE_16;
+				break;
+
+			case 22050:
+				sr = SL_SAMPLINGRATE_22_05;
+				break;
+
+			case 24000:
+				sr = SL_SAMPLINGRATE_24;
+				break;
+
+			case 32000:
+				sr = SL_SAMPLINGRATE_32;
+				break;
+
+			case 44100:
+				sr = SL_SAMPLINGRATE_44_1;
+				break;
+
+			case 48000:
+				sr = SL_SAMPLINGRATE_48;
+				break;
+
+			case 64000:
+				sr = SL_SAMPLINGRATE_64;
+				break;
+
+			case 88200:
+				sr = SL_SAMPLINGRATE_88_2;
+				break;
+
+			case 96000:
+				sr = SL_SAMPLINGRATE_96;
+				break;
+
+			case 192000:
+				sr = SL_SAMPLINGRATE_192;
+				break;
+
+			default:
+				return -1;
+		}
+
+		const SLInterfaceID ids[] = { SL_IID_VOLUME };
+		const SLboolean req[] = { SL_BOOLEAN_FALSE };
+		result = (*p->engineEngine)
+		             ->CreateOutputMix(p->engineEngine, &(p->outputMixObject), 1, ids, req);
+		DEBUG_SND("engineEngine=%p", (void*)p->engineEngine);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		// realize the output mix
+		result = (*p->outputMixObject)->Realize(p->outputMixObject, SL_BOOLEAN_FALSE);
+		DEBUG_SND("Realize=%" PRIu32 "", result);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		int speakers;
+
+		if (channels > 1)
+			speakers = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+		else
+			speakers = SL_SPEAKER_FRONT_CENTER;
+
+		SLDataFormat_PCM format_pcm = { SL_DATAFORMAT_PCM,
+			                            channels,
+			                            sr,
+			                            SL_PCMSAMPLEFORMAT_FIXED_16,
+			                            SL_PCMSAMPLEFORMAT_FIXED_16,
+			                            speakers,
+			                            SL_BYTEORDER_LITTLEENDIAN };
+		SLDataSource audioSrc = { &loc_bufq, &format_pcm };
+		// configure audio sink
+		SLDataLocator_OutputMix loc_outmix = { SL_DATALOCATOR_OUTPUTMIX, p->outputMixObject };
+		SLDataSink audioSnk = { &loc_outmix, nullptr };
+		// create audio player
+		const SLInterfaceID ids1[] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_VOLUME };
+		const SLboolean req1[] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
+		result = (*p->engineEngine)
+		             ->CreateAudioPlayer(p->engineEngine, &(p->bqPlayerObject), &audioSrc,
+		                                 &audioSnk, 2, ids1, req1);
+		DEBUG_SND("bqPlayerObject=%p", (void*)p->bqPlayerObject);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		// realize the player
+		result = (*p->bqPlayerObject)->Realize(p->bqPlayerObject, SL_BOOLEAN_FALSE);
+		DEBUG_SND("Realize=%" PRIu32 "", result);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		// get the play interface
+		result =
+		    (*p->bqPlayerObject)->GetInterface(p->bqPlayerObject, SL_IID_PLAY, &(p->bqPlayerPlay));
+		DEBUG_SND("bqPlayerPlay=%p", (void*)p->bqPlayerPlay);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		// get the volume interface
+		result = (*p->bqPlayerObject)
+		             ->GetInterface(p->bqPlayerObject, SL_IID_VOLUME, &(p->bqPlayerVolume));
+		DEBUG_SND("bqPlayerVolume=%p", (void*)p->bqPlayerVolume);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		// get the buffer queue interface
+		result = (*p->bqPlayerObject)
+		             ->GetInterface(p->bqPlayerObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+		                            &(p->bqPlayerBufferQueue));
+		DEBUG_SND("bqPlayerBufferQueue=%p", (void*)p->bqPlayerBufferQueue);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		// register callback on the buffer queue
+		result = (*p->bqPlayerBufferQueue)
+		             ->RegisterCallback(p->bqPlayerBufferQueue, bqPlayerCallback, p);
+		DEBUG_SND("bqPlayerCallback=%p", (void*)p->bqPlayerCallback);
+		WINPR_ASSERT(!result);
+
+		if (result != SL_RESULT_SUCCESS)
+			goto end_openaudio;
+
+		// set the player's state to playing
+		result = (*p->bqPlayerPlay)->SetPlayState(p->bqPlayerPlay, SL_PLAYSTATE_PLAYING);
+		DEBUG_SND("SetPlayState=%" PRIu32 "", result);
+		WINPR_ASSERT(!result);
+	end_openaudio:
+		WINPR_ASSERT(!result);
+		return result;
+	}
+
+	return SL_RESULT_SUCCESS;
+}
+
+// close the OpenSL IO and destroy the audio engine
+static void openSLDestroyEngine(OPENSL_STREAM* p)
+{
+	// destroy buffer queue audio player object, and invalidate all associated interfaces
+	if (p->bqPlayerObject != nullptr)
+	{
+		(*p->bqPlayerObject)->Destroy(p->bqPlayerObject);
+		p->bqPlayerObject = nullptr;
+		p->bqPlayerVolume = nullptr;
+		p->bqPlayerPlay = nullptr;
+		p->bqPlayerBufferQueue = nullptr;
+		p->bqPlayerEffectSend = nullptr;
+	}
+
+	// destroy output mix object, and invalidate all associated interfaces
+	if (p->outputMixObject != nullptr)
+	{
+		(*p->outputMixObject)->Destroy(p->outputMixObject);
+		p->outputMixObject = nullptr;
+	}
+
+	// destroy engine object, and invalidate all associated interfaces
+	if (p->engineObject != nullptr)
+	{
+		(*p->engineObject)->Destroy(p->engineObject);
+		p->engineObject = nullptr;
+		p->engineEngine = nullptr;
+	}
+}
+
+// open the android audio device for and/or output
+OPENSL_STREAM* android_OpenAudioDevice(int sr, int outchannels, int bufferframes)
+{
+	OPENSL_STREAM* p;
+	p = (OPENSL_STREAM*)calloc(1, sizeof(OPENSL_STREAM));
+
+	if (!p)
+		return nullptr;
+
+	p->queuesize = bufferframes;
+	p->outchannels = outchannels;
+	p->sr = sr;
+
+	if (openSLCreateEngine(p) != SL_RESULT_SUCCESS)
+	{
+		android_CloseAudioDevice(p);
+		return nullptr;
+	}
+
+	if (openSLPlayOpen(p) != SL_RESULT_SUCCESS)
+	{
+		android_CloseAudioDevice(p);
+		return nullptr;
+	}
+
+	p->queue = Queue_New(TRUE, -1, -1);
+
+	if (!p->queue)
+	{
+		android_CloseAudioDevice(p);
+		return nullptr;
+	}
+
+	return p;
+}
+
+// close the android audio device
+void android_CloseAudioDevice(OPENSL_STREAM* p)
+{
+	if (p == nullptr)
+		return;
+
+	openSLDestroyEngine(p);
+
+	if (p->queue)
+		Queue_Free(p->queue);
+
+	free(p);
+}
+
+// this callback handler is called every time a buffer finishes playing
+static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void* context)
+{
+	OPENSL_STREAM* p = (OPENSL_STREAM*)context;
+	WINPR_ASSERT(p);
+	WINPR_ASSERT(p->queue);
+	void* data = Queue_Dequeue(p->queue);
+	free(data);
+}
+
+// puts a buffer of size samples to the device
+int android_AudioOut(OPENSL_STREAM* p, const short* buffer, int size)
+{
+	HANDLE ev;
+	WINPR_ASSERT(p);
+	WINPR_ASSERT(buffer);
+	WINPR_ASSERT(size > 0);
+
+	ev = Queue_Event(p->queue);
+	/* Assure, that the queue is not full. */
+	if (p->queuesize <= Queue_Count(p->queue) && WaitForSingleObject(ev, INFINITE) == WAIT_FAILED)
+	{
+		DEBUG_SND("WaitForSingleObject failed!");
+		return -1;
+	}
+
+	void* data = calloc(size, sizeof(short));
+
+	if (!data)
+	{
+		DEBUG_SND("unable to allocate a buffer");
+		return -1;
+	}
+
+	memcpy(data, buffer, size * sizeof(short));
+	Queue_Enqueue(p->queue, data);
+	(*p->bqPlayerBufferQueue)->Enqueue(p->bqPlayerBufferQueue, data, sizeof(short) * size);
+	return size;
+}
+
+int android_GetOutputMute(OPENSL_STREAM* p)
+{
+	SLboolean mute;
+	WINPR_ASSERT(p);
+	WINPR_ASSERT(p->bqPlayerVolume);
+	SLresult rc = (*p->bqPlayerVolume)->GetMute(p->bqPlayerVolume, &mute);
+
+	if (SL_RESULT_SUCCESS != rc)
+		return SL_BOOLEAN_FALSE;
+
+	return mute;
+}
+
+BOOL android_SetOutputMute(OPENSL_STREAM* p, BOOL _mute)
+{
+	SLboolean mute = _mute;
+	WINPR_ASSERT(p);
+	WINPR_ASSERT(p->bqPlayerVolume);
+	SLresult rc = (*p->bqPlayerVolume)->SetMute(p->bqPlayerVolume, mute);
+
+	if (SL_RESULT_SUCCESS != rc)
+		return FALSE;
+
+	return TRUE;
+}
+
+int android_GetOutputVolume(OPENSL_STREAM* p)
+{
+	SLmillibel level;
+	WINPR_ASSERT(p);
+	WINPR_ASSERT(p->bqPlayerVolume);
+	SLresult rc = (*p->bqPlayerVolume)->GetVolumeLevel(p->bqPlayerVolume, &level);
+
+	if (SL_RESULT_SUCCESS != rc)
+		return 0;
+
+	return level;
+}
+
+int android_GetOutputVolumeMax(OPENSL_STREAM* p)
+{
+	SLmillibel level;
+	WINPR_ASSERT(p);
+	WINPR_ASSERT(p->bqPlayerVolume);
+	SLresult rc = (*p->bqPlayerVolume)->GetMaxVolumeLevel(p->bqPlayerVolume, &level);
+
+	if (SL_RESULT_SUCCESS != rc)
+		return 0;
+
+	return level;
+}
+
+BOOL android_SetOutputVolume(OPENSL_STREAM* p, int level)
+{
+	SLresult rc = (*p->bqPlayerVolume)->SetVolumeLevel(p->bqPlayerVolume, level);
+
+	if (SL_RESULT_SUCCESS != rc)
+		return FALSE;
+
+	return TRUE;
+}

third_party/FreeRDP/channels/rdpsnd/client/opensles/opensl_io.h

@@ -0,0 +1,110 @@
+/*
+opensl_io.c:
+Android OpenSL input/output module header
+Copyright (c) 2012, Victor Lazzarini
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are met:
+    * Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+    * Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+    * Neither the name of the <organization> nor the
+      names of its contributors may be used to endorse or promote products
+      derived from this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+DISCLAIMED. IN NO EVENT SHALL <COPYRIGHT HOLDER> BE LIABLE FOR ANY
+DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef FREERDP_CHANNEL_RDPSND_CLIENT_OPENSL_IO_H
+#define FREERDP_CHANNEL_RDPSND_CLIENT_OPENSL_IO_H
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <stdlib.h>
+#include <winpr/synch.h>
+
+#include <freerdp/api.h>
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+	typedef struct
+	{
+		// engine interfaces
+		SLObjectItf engineObject;
+		SLEngineItf engineEngine;
+
+		// output mix interfaces
+		SLObjectItf outputMixObject;
+
+		// buffer queue player interfaces
+		SLObjectItf bqPlayerObject;
+		SLPlayItf bqPlayerPlay;
+		SLVolumeItf bqPlayerVolume;
+		SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue;
+		SLEffectSendItf bqPlayerEffectSend;
+
+		unsigned int outchannels;
+		unsigned int sr;
+
+		unsigned int queuesize;
+		wQueue* queue;
+	} OPENSL_STREAM;
+
+	/*
+	Open the audio device with a given sampling rate (sr), output channels and IO buffer size
+	in frames. Returns a handle to the OpenSL stream
+	*/
+	FREERDP_LOCAL OPENSL_STREAM* android_OpenAudioDevice(int sr, int outchannels, int bufferframes);
+	/*
+	Close the audio device
+	*/
+	FREERDP_LOCAL void android_CloseAudioDevice(OPENSL_STREAM* p);
+	/*
+	Write a buffer to the OpenSL stream *p, of size samples. Returns the number of samples written.
+	*/
+	FREERDP_LOCAL int android_AudioOut(OPENSL_STREAM* p, const short* buffer, int size);
+	/*
+	 * Set the volume input level.
+	 */
+	FREERDP_LOCAL void android_SetInputVolume(OPENSL_STREAM* p, int level);
+	/*
+	 * Get the current output mute setting.
+	 */
+	FREERDP_LOCAL int android_GetOutputMute(OPENSL_STREAM* p);
+	/*
+	 * Change the current output mute setting.
+	 */
+	FREERDP_LOCAL BOOL android_SetOutputMute(OPENSL_STREAM* p, BOOL mute);
+	/*
+	 * Get the current output volume level.
+	 */
+	FREERDP_LOCAL int android_GetOutputVolume(OPENSL_STREAM* p);
+	/*
+	 * Get the maximum output volume level.
+	 */
+	FREERDP_LOCAL int android_GetOutputVolumeMax(OPENSL_STREAM* p);
+
+	/*
+	 * Set the volume output level.
+	 */
+	FREERDP_LOCAL BOOL android_SetOutputVolume(OPENSL_STREAM* p, int level);
+#ifdef __cplusplus
+};
+#endif
+
+#endif /* FREERDP_CHANNEL_RDPSND_CLIENT_OPENSL_IO_H */

third_party/FreeRDP/channels/rdpsnd/client/opensles/rdpsnd_opensles.c

@@ -0,0 +1,373 @@
+/**
+ * FreeRDP: A Remote Desktop Protocol Implementation
+ * Audio Output Virtual Channel
+ *
+ * Copyright 2013 Armin Novak <armin.novak@gmail.com>
+ * Copyright 2015 Thincast Technologies GmbH
+ * Copyright 2015 DI (FH) Martin Haimberger <martin.haimberger@thincast.com>
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <freerdp/config.h>
+
+#include <winpr/assert.h>
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <stdbool.h>
+
+#include <winpr/crt.h>
+#include <winpr/cmdline.h>
+#include <winpr/sysinfo.h>
+#include <winpr/collections.h>
+
+#include <freerdp/types.h>
+#include <freerdp/channels/log.h>
+
+#include "opensl_io.h"
+#include "rdpsnd_main.h"
+
+typedef struct
+{
+	rdpsndDevicePlugin device;
+
+	UINT32 latency;
+	int wformat;
+	int block_size;
+	char* device_name;
+
+	OPENSL_STREAM* stream;
+
+	UINT32 volume;
+
+	UINT32 rate;
+	UINT32 channels;
+	int format;
+} rdpsndopenslesPlugin;
+
+static int rdpsnd_opensles_volume_to_millibel(unsigned short level, int max)
+{
+	const int min = SL_MILLIBEL_MIN;
+	const int step = max - min;
+	const int rc = (level * step / 0xFFFF) + min;
+	DEBUG_SND("level=%hu, min=%d, max=%d, step=%d, result=%d", level, min, max, step, rc);
+	return rc;
+}
+
+static unsigned short rdpsnd_opensles_millibel_to_volume(int millibel, int max)
+{
+	const int min = SL_MILLIBEL_MIN;
+	const int range = max - min;
+	const int rc = ((millibel - min) * 0xFFFF + range / 2 + 1) / range;
+	DEBUG_SND("millibel=%d, min=%d, max=%d, range=%d, result=%d", millibel, min, max, range, rc);
+	return rc;
+}
+
+static bool rdpsnd_opensles_check_handle(const rdpsndopenslesPlugin* hdl)
+{
+	bool rc = true;
+
+	if (!hdl)
+		rc = false;
+	else
+	{
+		if (!hdl->stream)
+			rc = false;
+	}
+
+	return rc;
+}
+
+static BOOL rdpsnd_opensles_set_volume(rdpsndDevicePlugin* device, UINT32 volume);
+
+static int rdpsnd_opensles_set_params(rdpsndopenslesPlugin* opensles)
+{
+	DEBUG_SND("opensles=%p", (void*)opensles);
+
+	if (!rdpsnd_opensles_check_handle(opensles))
+		return 0;
+
+	if (opensles->stream)
+		android_CloseAudioDevice(opensles->stream);
+
+	opensles->stream = android_OpenAudioDevice(opensles->rate, opensles->channels, 20);
+	return 0;
+}
+
+static BOOL rdpsnd_opensles_set_format(rdpsndDevicePlugin* device, const AUDIO_FORMAT* format,
+                                       UINT32 latency)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	rdpsnd_opensles_check_handle(opensles);
+	DEBUG_SND("opensles=%p format=%p, latency=%" PRIu32, (void*)opensles, (void*)format, latency);
+
+	if (format)
+	{
+		DEBUG_SND("format=%" PRIu16 ", cbsize=%" PRIu16 ", samples=%" PRIu32 ", bits=%" PRIu16
+		          ", channels=%" PRIu16 ", align=%" PRIu16 "",
+		          format->wFormatTag, format->cbSize, format->nSamplesPerSec,
+		          format->wBitsPerSample, format->nChannels, format->nBlockAlign);
+		opensles->rate = format->nSamplesPerSec;
+		opensles->channels = format->nChannels;
+		opensles->format = format->wFormatTag;
+		opensles->wformat = format->wFormatTag;
+		opensles->block_size = format->nBlockAlign;
+	}
+
+	opensles->latency = latency;
+	return (rdpsnd_opensles_set_params(opensles) == 0);
+}
+
+static BOOL rdpsnd_opensles_open(rdpsndDevicePlugin* device, const AUDIO_FORMAT* format,
+                                 UINT32 latency)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	DEBUG_SND("opensles=%p format=%p, latency=%" PRIu32 ", rate=%" PRIu32 "", (void*)opensles,
+	          (void*)format, latency, opensles->rate);
+
+	if (rdpsnd_opensles_check_handle(opensles))
+		return TRUE;
+
+	opensles->stream = android_OpenAudioDevice(opensles->rate, opensles->channels, 20);
+	WINPR_ASSERT(opensles->stream);
+
+	if (!opensles->stream)
+		WLog_ERR(TAG, "android_OpenAudioDevice failed");
+	else
+		rdpsnd_opensles_set_volume(device, opensles->volume);
+
+	return rdpsnd_opensles_set_format(device, format, latency);
+}
+
+static void rdpsnd_opensles_close(rdpsndDevicePlugin* device)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	DEBUG_SND("opensles=%p", (void*)opensles);
+
+	if (!rdpsnd_opensles_check_handle(opensles))
+		return;
+
+	android_CloseAudioDevice(opensles->stream);
+	opensles->stream = nullptr;
+}
+
+static void rdpsnd_opensles_free(rdpsndDevicePlugin* device)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	DEBUG_SND("opensles=%p", (void*)opensles);
+	WINPR_ASSERT(opensles);
+	WINPR_ASSERT(opensles->device_name);
+	free(opensles->device_name);
+	free(opensles);
+}
+
+static BOOL rdpsnd_opensles_format_supported(rdpsndDevicePlugin* device, const AUDIO_FORMAT* format)
+{
+	DEBUG_SND("format=%" PRIu16 ", cbsize=%" PRIu16 ", samples=%" PRIu32 ", bits=%" PRIu16
+	          ", channels=%" PRIu16 ", align=%" PRIu16 "",
+	          format->wFormatTag, format->cbSize, format->nSamplesPerSec, format->wBitsPerSample,
+	          format->nChannels, format->nBlockAlign);
+	WINPR_ASSERT(device);
+	WINPR_ASSERT(format);
+
+	switch (format->wFormatTag)
+	{
+		case WAVE_FORMAT_PCM:
+			if (format->cbSize == 0 && format->nSamplesPerSec <= 48000 &&
+			    (format->wBitsPerSample == 8 || format->wBitsPerSample == 16) &&
+			    (format->nChannels == 1 || format->nChannels == 2))
+			{
+				return TRUE;
+			}
+
+			break;
+
+		default:
+			break;
+	}
+
+	return FALSE;
+}
+
+static UINT32 rdpsnd_opensles_get_volume(rdpsndDevicePlugin* device)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	DEBUG_SND("opensles=%p", (void*)opensles);
+	WINPR_ASSERT(opensles);
+
+	if (opensles->stream)
+	{
+		const int max = android_GetOutputVolumeMax(opensles->stream);
+		const int rc = android_GetOutputVolume(opensles->stream);
+
+		if (android_GetOutputMute(opensles->stream))
+			opensles->volume = 0;
+		else
+		{
+			const unsigned short vol = rdpsnd_opensles_millibel_to_volume(rc, max);
+			opensles->volume = (vol << 16) | (vol & 0xFFFF);
+		}
+	}
+
+	return opensles->volume;
+}
+
+static BOOL rdpsnd_opensles_set_volume(rdpsndDevicePlugin* device, UINT32 value)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	DEBUG_SND("opensles=%p, value=%" PRIu32 "", (void*)opensles, value);
+	WINPR_ASSERT(opensles);
+	opensles->volume = value;
+
+	if (opensles->stream)
+	{
+		if (0 == opensles->volume)
+			return android_SetOutputMute(opensles->stream, true);
+		else
+		{
+			const int max = android_GetOutputVolumeMax(opensles->stream);
+			const int vol = rdpsnd_opensles_volume_to_millibel(value & 0xFFFF, max);
+
+			if (!android_SetOutputMute(opensles->stream, false))
+				return FALSE;
+
+			if (!android_SetOutputVolume(opensles->stream, vol))
+				return FALSE;
+		}
+	}
+
+	return TRUE;
+}
+
+static UINT rdpsnd_opensles_play(rdpsndDevicePlugin* device, const BYTE* data, size_t size)
+{
+	union
+	{
+		const BYTE* b;
+		const short* s;
+	} src;
+	int ret;
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	DEBUG_SND("opensles=%p, data=%p, size=%d", (void*)opensles, (void*)data, size);
+
+	if (!rdpsnd_opensles_check_handle(opensles))
+		return 0;
+
+	src.b = data;
+	DEBUG_SND("size=%d, src=%p", size, (void*)src.b);
+	WINPR_ASSERT(0 == size % 2);
+	WINPR_ASSERT(size > 0);
+	WINPR_ASSERT(src.b);
+	ret = android_AudioOut(opensles->stream, src.s, size / 2);
+
+	if (ret < 0)
+		WLog_ERR(TAG, "android_AudioOut failed (%d)", ret);
+
+	return 10; /* TODO: Get real latencry in [ms] */
+}
+
+static void rdpsnd_opensles_start(rdpsndDevicePlugin* device)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	rdpsnd_opensles_check_handle(opensles);
+	DEBUG_SND("opensles=%p", (void*)opensles);
+}
+
+static int rdpsnd_opensles_parse_addin_args(rdpsndDevicePlugin* device, const ADDIN_ARGV* args)
+{
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)device;
+	COMMAND_LINE_ARGUMENT_A rdpsnd_opensles_args[] = {
+		{ "dev", COMMAND_LINE_VALUE_REQUIRED, "<device>", nullptr, nullptr, -1, nullptr, "device" },
+		{ nullptr, 0, nullptr, nullptr, nullptr, -1, nullptr, nullptr }
+	};
+
+	WINPR_ASSERT(opensles);
+	WINPR_ASSERT(args);
+	DEBUG_SND("opensles=%p, args=%p", (void*)opensles, (void*)args);
+	const DWORD flags =
+	    COMMAND_LINE_SIGIL_NONE | COMMAND_LINE_SEPARATOR_COLON | COMMAND_LINE_IGN_UNKNOWN_KEYWORD;
+	const int status = CommandLineParseArgumentsA(args->argc, args->argv, rdpsnd_opensles_args,
+	                                              flags, opensles, nullptr, nullptr);
+
+	if (status < 0)
+		return status;
+
+	const COMMAND_LINE_ARGUMENT_A* arg = rdpsnd_opensles_args;
+
+	do
+	{
+		if (!(arg->Flags & COMMAND_LINE_VALUE_PRESENT))
+			continue;
+
+		CommandLineSwitchStart(arg) CommandLineSwitchCase(arg, "dev")
+		{
+			opensles->device_name = _strdup(arg->Value);
+
+			if (!opensles->device_name)
+				return ERROR_OUTOFMEMORY;
+		}
+		CommandLineSwitchEnd(arg)
+	} while ((arg = CommandLineFindNextArgumentA(arg)) != nullptr);
+
+	return status;
+}
+
+/**
+ * Function description
+ *
+ * @return 0 on success, otherwise a Win32 error code
+ */
+FREERDP_ENTRY_POINT(UINT VCAPITYPE opensles_freerdp_rdpsnd_client_subsystem_entry(
+    PFREERDP_RDPSND_DEVICE_ENTRY_POINTS pEntryPoints))
+{
+	UINT error = ERROR_INTERNAL_ERROR;
+	DEBUG_SND("pEntryPoints=%p", (void*)pEntryPoints);
+	rdpsndopenslesPlugin* opensles = (rdpsndopenslesPlugin*)calloc(1, sizeof(rdpsndopenslesPlugin));
+
+	if (!opensles)
+		return CHANNEL_RC_NO_MEMORY;
+
+	opensles->device.Open = rdpsnd_opensles_open;
+	opensles->device.FormatSupported = rdpsnd_opensles_format_supported;
+	opensles->device.GetVolume = rdpsnd_opensles_get_volume;
+	opensles->device.SetVolume = rdpsnd_opensles_set_volume;
+	opensles->device.Start = rdpsnd_opensles_start;
+	opensles->device.Play = rdpsnd_opensles_play;
+	opensles->device.Close = rdpsnd_opensles_close;
+	opensles->device.Free = rdpsnd_opensles_free;
+	const ADDIN_ARGV* args = pEntryPoints->args;
+	rdpsnd_opensles_parse_addin_args((rdpsndDevicePlugin*)opensles, args);
+
+	if (!opensles->device_name)
+	{
+		opensles->device_name = _strdup("default");
+
+		if (!opensles->device_name)
+		{
+			error = CHANNEL_RC_NO_MEMORY;
+			goto outstrdup;
+		}
+	}
+
+	opensles->rate = 44100;
+	opensles->channels = 2;
+	opensles->format = WAVE_FORMAT_ADPCM;
+	pEntryPoints->pRegisterRdpsndDevice(pEntryPoints->rdpsnd, (rdpsndDevicePlugin*)opensles);
+	DEBUG_SND("success");
+	return CHANNEL_RC_OK;
+outstrdup:
+	free(opensles);
+	return error;
+}