/**

 * FreeRDP: A Remote Desktop Protocol Implementation

 * FreeRDP Mac OS X Server (Audio Output)

 *

 * Copyright 2012 Marc-Andre Moreau <marcandre.moreau@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.

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <freerdp/server/rdpsnd.h>

#include "mf_info.h"

#include "mf_rdpsnd.h"

#include <winpr/sysinfo.h>

#include <freerdp/server/server-common.h>

#include <freerdp/log.h>

#define TAG SERVER_TAG("mac")

AQRecorderState recorderState;

static void mf_peer_rdpsnd_activated(RdpsndServerContext* context)

{

	OSStatus status;

	int i, j;

	BOOL formatAgreed = FALSE;

	AUDIO_FORMAT* agreedFormat = NULL;

	// we should actually loop through the list of client formats here

	// and see if we can send the client something that it supports...

	WLog_DBG(TAG, "Client supports the following %d formats: ", context->num_client_formats);

	for (i = 0; i < context->num_client_formats; i++)

	{

		/* TODO: improve the way we agree on a format */

		for (j = 0; j < context->num_server_formats; j++)

		{

			if ((context->client_formats[i].wFormatTag == context->server_formats[j].wFormatTag) &&

			    (context->client_formats[i].nChannels == context->server_formats[j].nChannels) &&

			    (context->client_formats[i].nSamplesPerSec ==

			     context->server_formats[j].nSamplesPerSec))

			{

				WLog_DBG(TAG, "agreed on format!");

				formatAgreed = TRUE;

				agreedFormat = (AUDIO_FORMAT*)&context->server_formats[j];

				break;

			}

		}

		if (formatAgreed == TRUE)

			break;

	}

	if (formatAgreed == FALSE)

	{

		WLog_DBG(TAG, "Could not agree on a audio format with the server");

		return;

	}

	context->SelectFormat(context, i);

	context->SetVolume(context, 0x7FFF, 0x7FFF);

	switch (agreedFormat->wFormatTag)

	{

		case WAVE_FORMAT_ALAW:

			recorderState.dataFormat.mFormatID = kAudioFormatDVIIntelIMA;

			break;

		case WAVE_FORMAT_PCM:

			recorderState.dataFormat.mFormatID = kAudioFormatLinearPCM;

			break;

		default:

			recorderState.dataFormat.mFormatID = kAudioFormatLinearPCM;

			break;

	}

	recorderState.dataFormat.mSampleRate = agreedFormat->nSamplesPerSec;

	recorderState.dataFormat.mFormatFlags =

	    kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;

	recorderState.dataFormat.mBytesPerPacket = 4;

	recorderState.dataFormat.mFramesPerPacket = 1;

	recorderState.dataFormat.mBytesPerFrame = 4;

	recorderState.dataFormat.mChannelsPerFrame = agreedFormat->nChannels;

	recorderState.dataFormat.mBitsPerChannel = agreedFormat->wBitsPerSample;

	recorderState.snd_context = context;

	status =

	    AudioQueueNewInput(&recorderState.dataFormat, mf_peer_rdpsnd_input_callback, &recorderState,

	                       NULL, kCFRunLoopCommonModes, 0, &recorderState.queue);

	if (status != noErr)

	{

		WLog_DBG(TAG, "Failed to create a new Audio Queue. Status code: %" PRId32 "", status);

	}

	UInt32 dataFormatSize = sizeof(recorderState.dataFormat);

	AudioQueueGetProperty(recorderState.queue, kAudioConverterCurrentInputStreamDescription,

	                      &recorderState.dataFormat, &dataFormatSize);

	mf_rdpsnd_derive_buffer_size(recorderState.queue, &recorderState.dataFormat, 0.05,

	                             &recorderState.bufferByteSize);

	for (i = 0; i < SND_NUMBUFFERS; ++i)

	{

		AudioQueueAllocateBuffer(recorderState.queue, recorderState.bufferByteSize,

		                         &recorderState.buffers[i]);

		AudioQueueEnqueueBuffer(recorderState.queue, recorderState.buffers[i], 0, NULL);

	}

	recorderState.currentPacket = 0;

	recorderState.isRunning = true;

	AudioQueueStart(recorderState.queue, NULL);

}

BOOL mf_peer_rdpsnd_init(mfPeerContext* context)

{

	context->rdpsnd = rdpsnd_server_context_new(context->vcm);

	context->rdpsnd->rdpcontext = &context->_p;

	context->rdpsnd->data = context;

	context->rdpsnd->num_server_formats =

	    server_rdpsnd_get_formats(&context->rdpsnd->server_formats);

	if (context->rdpsnd->num_server_formats > 0)

		context->rdpsnd->src_format = &context->rdpsnd->server_formats[0];

	context->rdpsnd->Activated = mf_peer_rdpsnd_activated;

	context->rdpsnd->Initialize(context->rdpsnd, TRUE);

	return TRUE;

}

BOOL mf_peer_rdpsnd_stop()

{

	recorderState.isRunning = false;

	AudioQueueStop(recorderState.queue, true);

	return TRUE;

}

void mf_peer_rdpsnd_input_callback(void* inUserData, AudioQueueRef inAQ,

                                   AudioQueueBufferRef inBuffer, const AudioTimeStamp* inStartTime,

                                   UInt32 inNumberPacketDescriptions,

                                   const AudioStreamPacketDescription* inPacketDescs)

{

	OSStatus status;

	AQRecorderState* rState;

	rState = inUserData;

	if (inNumberPacketDescriptions == 0 && rState->dataFormat.mBytesPerPacket != 0)

	{

		inNumberPacketDescriptions =

		    inBuffer->mAudioDataByteSize / rState->dataFormat.mBytesPerPacket;

	}

	if (rState->isRunning == 0)

	{

		return;

	}

	rState->snd_context->SendSamples(rState->snd_context, inBuffer->mAudioData,

	                                 inBuffer->mAudioDataByteSize / 4,

	                                 (UINT16)(GetTickCount() & 0xffff));

	status = AudioQueueEnqueueBuffer(rState->queue, inBuffer, 0, NULL);

	if (status != noErr)

	{

		WLog_DBG(TAG, "AudioQueueEnqueueBuffer() returned status = %" PRId32 "", status);

	}

}

void mf_rdpsnd_derive_buffer_size(AudioQueueRef audioQueue,

                                  AudioStreamBasicDescription* ASBDescription, Float64 seconds,

                                  UInt32* outBufferSize)

{

	static const int maxBufferSize = 0x50000;

	int maxPacketSize = ASBDescription->mBytesPerPacket;

	if (maxPacketSize == 0)

	{

		UInt32 maxVBRPacketSize = sizeof(maxPacketSize);

		AudioQueueGetProperty(audioQueue, kAudioQueueProperty_MaximumOutputPacketSize,

		                      // in Mac OS X v10.5, instead use

		                      //   kAudioConverterPropertyMaximumOutputPacketSize

		                      &maxPacketSize, &maxVBRPacketSize);

	}

	Float64 numBytesForTime = ASBDescription->mSampleRate * maxPacketSize * seconds;

	*outBufferSize = (UInt32)(numBytesForTime < maxBufferSize ? numBytesForTime : maxBufferSize);

}