/*
 * Copyright 2002-2014 Haiku, Inc. All rights reserved.
 * Distributed under the terms of the MIT License.
 *
 * Authors:
 *              Christopher ML Zumwalt May (zummy@users.sf.net)
 */


/*      A MediaKit producer node which mixes sound from the GameKit
        and sends them to the audio mixer
*/


#include "GameProducer.h"

#include <string.h>
#include <stdio.h>

#include <Buffer.h>
#include <BufferGroup.h>
#include <ByteOrder.h>
#include <List.h>
#include <MediaDefs.h>
#include <TimeSource.h>

#include "GameSoundBuffer.h"
#include "GameSoundDevice.h"
#include "GSUtility.h"


struct _gs_play  {
        gs_id           sound;
        bool*           hook;

        _gs_play*       next;
        _gs_play*       previous;
};


GameProducer::GameProducer(GameSoundBuffer* object,
        const gs_audio_format* format)
        :
        BMediaNode("GameProducer.h"),
        BBufferProducer(B_MEDIA_RAW_AUDIO),
        BMediaEventLooper(),
        fBufferGroup(NULL),
        fLatency(0),
        fInternalLatency(0),
        fOutputEnabled(true)
{
        // initialize our preferred format object
        fPreferredFormat.type = B_MEDIA_RAW_AUDIO;
        fPreferredFormat.u.raw_audio.format = format->format;
        fPreferredFormat.u.raw_audio.channel_count = format->channel_count;
        fPreferredFormat.u.raw_audio.frame_rate = format->frame_rate; // Hertz
        fPreferredFormat.u.raw_audio.byte_order = format->byte_order;
//      fPreferredFormat.u.raw_audio.channel_mask
//              = B_CHANNEL_LEFT | B_CHANNEL_RIGHT;
//      fPreferredFormat.u.raw_audio.valid_bits = 32;
//      fPreferredFormat.u.raw_audio.matrix_mask = B_MATRIX_AMBISONIC_WXYZ;

        // we'll use the consumer's preferred buffer size, if any
        fPreferredFormat.u.raw_audio.buffer_size
                = media_raw_audio_format::wildcard.buffer_size;

        // we're not connected yet
        fOutput.destination = media_destination::null;
        fOutput.format = fPreferredFormat;

        fFrameSize = get_sample_size(format->format) * format->channel_count;
        fObject = object;
}


GameProducer::~GameProducer()
{
        // Stop the BMediaEventLooper thread
        Quit();
}


// BMediaNode methods
BMediaAddOn*
GameProducer::AddOn(int32* internal_id) const
{
        return NULL;
}


// BBufferProducer methods
status_t
GameProducer::GetNextOutput(int32* cookie, media_output* _output)
{
        // we currently support only one output
        if (0 != *cookie)
                return B_BAD_INDEX;

        *_output = fOutput;
        *cookie += 1;
        return B_OK;
}


status_t
GameProducer::DisposeOutputCookie(int32 cookie)
{
        // do nothing because our cookie is only an integer
        return B_OK;
}


void
GameProducer::EnableOutput(const media_source& what, bool enabled,
        int32* _deprecated_)
{
        // If I had more than one output, I'd have to walk my list of output records
        // to see which one matched the given source, and then enable/disable that
        // one.  But this node only has one output,  so I just make sure the given
        // source matches, then set the enable state accordingly.
        if (what == fOutput.source)
        {
                fOutputEnabled = enabled;
        }
}


status_t
GameProducer::FormatSuggestionRequested(media_type type, int32 /*quality*/,
        media_format* format)
{
        // insure that we received a format
        if (!format)
                return B_BAD_VALUE;

        // returning our preferred format
        *format = fPreferredFormat;

        // our format is supported
        if (type == B_MEDIA_UNKNOWN_TYPE)
                return B_OK;

        // we only support raw audo
        return (type != B_MEDIA_RAW_AUDIO) ? B_MEDIA_BAD_FORMAT : B_OK;
}


status_t
GameProducer::FormatProposal(const media_source& output, media_format* format)
{
        // doest the proposed output match our output?
        if (output != fOutput.source)
                return B_MEDIA_BAD_SOURCE;

        // return our preferred format
        *format = fPreferredFormat;

        // we will reject the proposal if the format is not audio
        media_type requestedType = format->type;
        if ((requestedType != B_MEDIA_UNKNOWN_TYPE)
                && (requestedType != B_MEDIA_RAW_AUDIO)) {
                return B_MEDIA_BAD_FORMAT;
        }

        return B_OK;            // raw audio or wildcard type, either is okay by us
}


status_t
GameProducer::PrepareToConnect(const media_source& what,
        const media_destination& where, media_format* format,
        media_source* _source, char* out_name)
{
        // The format has been processed by the consumer at this point. We need
        // to insure the format is still acceptable and any wild care are filled in.

        // trying to connect something that isn't our source?
        if (what != fOutput.source)
                return B_MEDIA_BAD_SOURCE;

        // are we already connected?
        if (fOutput.destination != media_destination::null)
                return B_MEDIA_ALREADY_CONNECTED;

        // the format may not yet be fully specialized (the consumer might have
        // passed back some wildcards).  Finish specializing it now, and return an
        // error if we don't support the requested format.
        if (format->type != B_MEDIA_RAW_AUDIO)
                return B_MEDIA_BAD_FORMAT;

        if (format->u.raw_audio.format != fPreferredFormat.u.raw_audio.format)
                return B_MEDIA_BAD_FORMAT;

        // check the buffer size, which may still be wildcarded
        if (format->u.raw_audio.buffer_size
                == media_raw_audio_format::wildcard.buffer_size) {
                format->u.raw_audio.buffer_size = 4096;
                        // pick something comfortable to suggest
        }

        // Now reserve the connection, and return information about it
        fOutput.destination = where;
        fOutput.format = *format;
        *_source = fOutput.source;
        strlcpy(out_name, fOutput.name, B_MEDIA_NAME_LENGTH);
        return B_OK;
}


void
GameProducer::Connect(status_t error, const media_source& source,
        const media_destination& destination, const media_format& format,
        char* ioName)
{
        // If something earlier failed, Connect() might still be called, but with a
        // non-zero error code.  When that happens we simply unreserve the
        // connection and do nothing else.
        if (error) {
                fOutput.destination = media_destination::null;
                fOutput.format = fPreferredFormat;
                return;
        }

        // Okay, the connection has been confirmed.  Record the destination and
        // format that we agreed on, and report our connection name again.
        fOutput.destination = destination;
        fOutput.format = format;
        strlcpy(ioName, fOutput.name, B_MEDIA_NAME_LENGTH);

        // Now that we're connected, we can determine our downstream latency.
        // Do so, then make sure we get our events early enough.
        media_node_id id;
        FindLatencyFor(fOutput.destination, &fLatency, &id);

        if (!fBufferGroup)
                fBufferSize = fOutput.format.u.raw_audio.buffer_size;
                        // Have to set it before latency calculating

        // Use a dry run to see how long it takes me to fill a buffer of data

        // The first step to setup the buffer
        bigtime_t start, produceLatency;
        int32 frames = int32(fBufferSize / fFrameSize);
        float* data = new float[frames * 2];

        // Second, fill the buffer
        start = ::system_time();
        for (int32 i = 0; i < frames; i++) {
                data[i * 2] = 0.8 * float(i / frames);
                data[i * 2 + 1] = 0.8 * float(i / frames);
        }
        produceLatency = ::system_time();

        // Third, calculate the latency
        fInternalLatency = produceLatency - start;
        SetEventLatency(fLatency + fInternalLatency);

        // Finaily, clean up
        delete [] data;

        // reset our buffer duration, etc. to avoid later calculations
        bigtime_t duration = bigtime_t(1000000) * frames
                / bigtime_t(fOutput.format.u.raw_audio.frame_rate);
        SetBufferDuration(duration);

        // Set up the buffer group for our connection, as long as nobody handed us a
        // buffer group (via SetBufferGroup()) prior to this.
        if (!fBufferGroup) {
                int32 count = int32(fLatency / BufferDuration() + 2);
                fBufferGroup = new BBufferGroup(fBufferSize, count);
        }
}


void
GameProducer::Disconnect(const media_source& what,
        const media_destination& where)
{
        // Make sure that our connection is the one being disconnected
        if ((where == fOutput.destination) && (what == fOutput.source)) {
                fOutput.destination = media_destination::null;
                fOutput.format = fPreferredFormat;
                delete fBufferGroup;
                fBufferGroup = NULL;
        }
}


status_t
GameProducer::FormatChangeRequested(const media_source& source,
        const media_destination& destination, media_format* io_format,
        int32* _deprecated_)
{
        // we don't support any other formats, so we just reject any format changes.
        return B_ERROR;
}


status_t
GameProducer::SetBufferGroup(const media_source& forSource,
        BBufferGroup* newGroup)
{
        // verify that we didn't get bogus arguments before we proceed
        if (forSource != fOutput.source)
                return B_MEDIA_BAD_SOURCE;

        // Are we being passed the buffer group we're already using?
        if (newGroup == fBufferGroup)
                return B_OK;

        // Ahh, someone wants us to use a different buffer group.  At this point we
        // delete the one we are using and use the specified one instead. If the
        // specified group is NULL, we need to recreate one ourselves, and use
        // *that*. Note that if we're caching a BBuffer that we requested earlier,
        // we have to Recycle() that buffer *before* deleting the buffer group,
        // otherwise we'll deadlock waiting for that buffer to be recycled!
        delete fBufferGroup;            // waits for all buffers to recycle
        if (newGroup != NULL) {
                // we were given a valid group; just use that one from now on
                fBufferGroup = newGroup;

                // get buffer length from the first buffer
                BBuffer* buffers[1];
                if (newGroup->GetBufferList(1, buffers) != B_OK)
                        return B_BAD_VALUE;
                fBufferSize = buffers[0]->SizeAvailable();
        } else {
                // we were passed a NULL group pointer; that means we construct
                // our own buffer group to use from now on
                fBufferSize = fOutput.format.u.raw_audio.buffer_size;
                int32 count = int32(fLatency / BufferDuration() + 2);
                fBufferGroup = new BBufferGroup(fBufferSize, count);
        }

        return B_OK;
}


status_t
GameProducer::GetLatency(bigtime_t* _latency)
{
        // report our *total* latency:  internal plus downstream plus scheduling
        *_latency = EventLatency() + SchedulingLatency();
        return B_OK;
}


void
GameProducer::LateNoticeReceived(const media_source& what, bigtime_t howMuch,
        bigtime_t performanceDuration)
{
        // If we're late, we need to catch up.  Respond in a manner appropriate to
        // our current run mode.
        if (what == fOutput.source) {
                if (RunMode() == B_RECORDING) {
                        // A hardware capture node can't adjust; it simply emits buffers at
                        // appropriate points.  We (partially) simulate this by not
                        // adjusting our behavior upon receiving late notices -- after all,
                        // the hardware can't choose to capture "sooner"...
                } else if (RunMode() == B_INCREASE_LATENCY) {
                        // We're late, and our run mode dictates that we try to produce
                        // buffers earlier in order to catch up. This argues that the
                        // downstream nodes are not properly reporting their latency, but
                        // there's not much we can do about that at the moment, so we try
                        // to start producing buffers earlier to compensate.
                        fInternalLatency += howMuch;
                        SetEventLatency(fLatency + fInternalLatency);
                } else {
                        // The other run modes dictate various strategies for sacrificing
                        // data quality in the interests of timely data delivery. The way we
                        // do this is to skip a buffer, which catches us up in time by one
                        // buffer duration.
                        size_t nSamples = fBufferSize / fFrameSize;
                        fFramesSent += nSamples;
                }
        }
}


void
GameProducer::LatencyChanged(const media_source& source,
        const media_destination& destination, bigtime_t new_latency, uint32 flags)
{
        // something downstream changed latency, so we need to start producing
        // buffers earlier (or later) than we were previously.  Make sure that the
        // connection that changed is ours, and adjust to the new downstream
        // latency if so.
        if ((source == fOutput.source) && (destination == fOutput.destination)) {
                fLatency = new_latency;
                SetEventLatency(fLatency + fInternalLatency);
        }
}


status_t
GameProducer::SetPlayRate(int32 numerator, int32 denominator)
{
        // Play rates are weird.  We don't support them
        return B_ERROR;
}


status_t
GameProducer::HandleMessage(int32 message, const void* data, size_t size)
{
        // We currently do not handle private messages
        return B_ERROR;
}


void
GameProducer::AdditionalBufferRequested(const media_source& source,
        media_buffer_id prev_buffer, bigtime_t prev_time,
        const media_seek_tag* prev_tag)
{
        // we don't support offline mode (yet...)
        return;
}


// BMediaEventLooper methods
void
GameProducer::NodeRegistered()
{
        // set up as much information about our output as we can
        fOutput.source.port = ControlPort();
        fOutput.source.id = 0;
        fOutput.node = Node();
        strlcpy(fOutput.name, "GameProducer Output", B_MEDIA_NAME_LENGTH);

        // Start the BMediaEventLooper thread
        SetPriority(B_REAL_TIME_PRIORITY);
        Run();
}


void
GameProducer::SetRunMode(run_mode mode)
{
        // We don't support offline run mode, so broadcast an error if we're set to
        // B_OFFLINE.  Unfortunately, we can't actually reject the mode change...
        if (B_OFFLINE == mode) {
                ReportError(B_NODE_FAILED_SET_RUN_MODE);
        }
}


void
GameProducer::HandleEvent(const media_timed_event* event, bigtime_t lateness,
        bool realTimeEvent)
{
//      FPRINTF(stderr, "ToneProducer::HandleEvent\n");
        switch (event->type)
        {
        case BTimedEventQueue::B_START:
                // don't do anything if we're already running
                if (RunState() != B_STARTED) {
                        // Going to start sending buffers so setup the needed bookkeeping
                        fFramesSent = 0;
                        fStartTime = event->event_time;
                        media_timed_event firstBufferEvent(fStartTime,
                                BTimedEventQueue::B_HANDLE_BUFFER);

                        // Alternatively, we could call HandleEvent() directly with this
                        // event, to avoid a trip through the event queue like this:
                        //              this->HandleEvent(&firstBufferEvent, 0, false);
                        EventQueue()->AddEvent(firstBufferEvent);
                }
                break;

        case BTimedEventQueue::B_STOP:
                // When we handle a stop, we must ensure that downstream consumers don't
                // get any more buffers from us.  This means we have to flush any
                // pending buffer-producing events from the queue.
                EventQueue()->FlushEvents(0, BTimedEventQueue::B_ALWAYS, true,
                        BTimedEventQueue::B_HANDLE_BUFFER);
                break;

        case BTimedEventQueue::B_HANDLE_BUFFER:
                {
                        // Ensure we're both started and connected before delivering buffer
                        if ((RunState() == BMediaEventLooper::B_STARTED)
                                && (fOutput.destination != media_destination::null)) {
                                // Get the next buffer of data
                                BBuffer* buffer = FillNextBuffer(event->event_time);
                                if (buffer) {
                                        // Send the buffer downstream if output is enabled
                                        status_t err = B_ERROR;
                                        if (fOutputEnabled) {
                                                err = SendBuffer(buffer, fOutput.source,
                                                        fOutput.destination);
                                        }
                                        if (err) {
                                                // we need to recycle the buffer ourselves if output is
                                                // disabled or if the call to SendBuffer() fails
                                                buffer->Recycle();
                                        }
                                }

                                // track how much media we've delivered so far
                                size_t nFrames = fBufferSize / fFrameSize;
                                fFramesSent += nFrames;

                                // The buffer is on its way; now schedule the next one to go
                                bigtime_t nextEvent = fStartTime + bigtime_t(double(fFramesSent)
                                        / double(fOutput.format.u.raw_audio.frame_rate)
                                        * 1000000.0);
                                media_timed_event nextBufferEvent(nextEvent,
                                        BTimedEventQueue::B_HANDLE_BUFFER);
                                EventQueue()->AddEvent(nextBufferEvent);
                        }
                }
                break;

        default:
                break;
        }
}


BBuffer*
GameProducer::FillNextBuffer(bigtime_t event_time)
{
        // get a buffer from our buffer group
        BBuffer* buf = fBufferGroup->RequestBuffer(fBufferSize, BufferDuration());

        // if we fail to get a buffer (for example, if the request times out), we
        // skip this buffer and go on to the next, to avoid locking up the control
        // thread.
        if (!buf)
                return NULL;

        // we need to discribe the buffer
        int64 frames = int64(fBufferSize / fFrameSize);
        memset(buf->Data(), 0, fBufferSize);

        // now fill the buffer with data, continuing where the last buffer left off
        fObject->Play(buf->Data(), frames);

        // fill in the buffer header
        media_header* hdr = buf->Header();
        hdr->type = B_MEDIA_RAW_AUDIO;
        hdr->size_used = fBufferSize;
        hdr->time_source = TimeSource()->ID();

        bigtime_t stamp;
        if (RunMode() == B_RECORDING) {
                // In B_RECORDING mode, we stamp with the capture time.  We're not
                // really a hardware capture node, but we simulate it by using the
                // (precalculated) time at which this buffer "should" have been created.
                stamp = event_time;
        } else {
                // okay, we're in one of the "live" performance run modes.  in these
                // modes, we stamp the buffer with the time at which the buffer should
                // be rendered to the output, not with the capture time. fStartTime is
                // the cached value of the first buffer's performance time; we calculate
                // this buffer's performance time as an offset from that time, based on
                // the amount of media we've created so far.
                // Recalculating every buffer like this avoids accumulation of error.
                stamp = fStartTime + bigtime_t(double(fFramesSent)
                        / double(fOutput.format.u.raw_audio.frame_rate) * 1000000.0);
        }
        hdr->start_time = stamp;

        return buf;
}