/*  This file is part of the KDE project
    Copyright (C) 2006 Matthias Kretz <kretz@kde.org>
    Copyright (C) 2009 Martin Sandsmark <sandsmark@samfundet.no>

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 of the License, or (at your option) version 3, or any
    later version accepted by the membership of KDE e.V. (or its
    successor approved by the membership of KDE e.V.), Nokia Corporation
    (or its successors, if any) and the KDE Free Qt Foundation, which shall
    act as a proxy defined in Section 6 of version 3 of the license.

    This library 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
    Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public
    License along with this library.  If not, see <http://www.gnu.org/licenses/>.

*/

#include "audiodataoutput.h"
#include "gsthelper.h"
#include "medianode.h"
#include "medianodeevent.h"
#include <QtCore/QVector>
#include <QtCore/QMap>
#include <phonon/audiooutput.h>

#include <gst/gstghostpad.h>
#include <gst/gstutils.h>

QT_BEGIN_HEADER
QT_BEGIN_NAMESPACE

namespace Phonon
{
namespace Gstreamer
{
AudioDataOutput::AudioDataOutput(Backend *backend, QObject *parent)
    : QObject(parent),
    MediaNode(backend, AudioSink)
{
    static int count = 0;
    m_name = "AudioDataOutput" + QString::number(count++);

    m_queue = gst_bin_new(NULL);
    gst_object_ref(GST_OBJECT(m_queue));
    gst_object_sink(GST_OBJECT(m_queue));
    GstElement* sink = gst_element_factory_make("fakesink", NULL);
    GstElement* queue = gst_element_factory_make("queue", NULL);
    GstElement* convert = gst_element_factory_make("audioconvert", NULL);

    g_signal_connect(sink, "handoff", G_CALLBACK(processBuffer), this);
    g_object_set(G_OBJECT(sink), "signal-handoffs", true, (const char*)NULL);

    //G_BYTE_ORDER is the host machine's endianess
    GstCaps *caps = gst_caps_new_simple("audio/x-raw-int",
                                        "endianess", G_TYPE_INT, G_BYTE_ORDER,
                                        "depth", G_TYPE_INT, 16,
                                        "channels", G_TYPE_INT, 2,
                                        NULL);

    gst_bin_add_many(GST_BIN(m_queue), sink, convert, queue, NULL);
    gst_element_link(queue, convert);
    gst_element_link_filtered(convert, sink, caps);

    GstPad *inputpad = gst_element_get_static_pad(queue, "sink");
    gst_element_add_pad(m_queue, gst_ghost_pad_new("sink", inputpad));
    gst_object_unref(inputpad);

    g_object_set(G_OBJECT(sink), "sync", true, (const char*)NULL);

    m_isValid = true;
}

AudioDataOutput::~AudioDataOutput()
{
    gst_element_set_state(m_queue, GST_STATE_NULL);
    gst_object_unref(m_queue);
}

int AudioDataOutput::dataSize() const
{
    return m_dataSize;
}

int AudioDataOutput::sampleRate() const
{
    return 44100;
}

void AudioDataOutput::setDataSize(int size)
{
    m_dataSize = size;
}

typedef QMap<Phonon::AudioDataOutput::Channel, QVector<float> > FloatMap;
typedef QMap<Phonon::AudioDataOutput::Channel, QVector<qint16> > IntMap;

inline void AudioDataOutput::convertAndEmit(const QVector<qint16> &leftBuffer, const QVector<qint16> &rightBuffer)
{
    //TODO: Floats
    IntMap map;
    map.insert(Phonon::AudioDataOutput::LeftChannel, leftBuffer);
    map.insert(Phonon::AudioDataOutput::RightChannel, rightBuffer);
    emit dataReady(map);
}

void AudioDataOutput::processBuffer(GstElement*, GstBuffer* buffer, GstPad*, gpointer gThat)
{
    // TODO emit endOfMedia
    AudioDataOutput *that = reinterpret_cast<AudioDataOutput*>(gThat);

    // determine the number of channels
    GstStructure* structure = gst_caps_get_structure (GST_BUFFER_CAPS(buffer), 0);
    gst_structure_get_int (structure, "channels", &that->m_channels);

    if (that->m_channels > 2 || that->m_channels < 0) {
        qWarning() << Q_FUNC_INFO << ": Number of channels not supported: " << that->m_channels;
        return;
    }

    gint16 *data = reinterpret_cast<gint16*>(GST_BUFFER_DATA(buffer));
    guint size = GST_BUFFER_SIZE(buffer) / sizeof(gint16);

    that->m_pendingData.reserve(that->m_pendingData.size() + size);

    for (uint i=0; i<size; i++) {
        // 8 bit? interleaved? yay for lacking documentation!
        that->m_pendingData.append(data[i]);
    }

    while (that->m_pendingData.size() > that->m_dataSize * that->m_channels) {
        if (that->m_channels == 1) {
            QVector<qint16> intBuffer(that->m_dataSize);
            memcpy(intBuffer.data(), that->m_pendingData.constData(), that->m_dataSize * sizeof(qint16));

            that->convertAndEmit(intBuffer, intBuffer);
            int newSize = that->m_pendingData.size() - that->m_dataSize;
            memmove(that->m_pendingData.data(), that->m_pendingData.constData() + that->m_dataSize, newSize * sizeof(qint16));
            that->m_pendingData.resize(newSize);
        } else {
            QVector<qint16> left(that->m_dataSize), right(that->m_dataSize);
            for (int i=0; i<that->m_dataSize; i++) {
                left[i] = that->m_pendingData[i*2];
                right[i] = that->m_pendingData[i*2+1];
            }
            that->m_pendingData.resize(that->m_pendingData.size() - that->m_dataSize*2);
            that->convertAndEmit(left, right);
        }
    }
}

}} //namespace Phonon::Gstreamer

QT_END_NAMESPACE
QT_END_HEADER

#include "moc_audiodataoutput.cpp"
// vim: sw=4 ts=4

