/* $Id$ */

/***
  This file is part of gst-pulse.

  gst-pulse 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) any later version.

  gst-pulse 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 gst-pulse; if not, write to the Free Software
  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
  USA.
***/

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

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

#include <gst/base/gstbasesrc.h>
#include <gst/gsttaglist.h>

#include "pulsesrc.h"
#include "pulseutil.h"
#include "pulsemixerctrl.h"

GST_DEBUG_CATEGORY_EXTERN(pulse_debug);
#define GST_CAT_DEFAULT pulse_debug

enum {
    PROP_SERVER = 1,
    PROP_DEVICE
};

static GstAudioSrcClass *parent_class = NULL;

GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS(GstPulseSrc, gst_pulsesrc)

static void gst_pulsesrc_destroy_stream(GstPulseSrc *pulsesrc);
static void gst_pulsesrc_destroy_context(GstPulseSrc *pulsesrc);

static void gst_pulsesrc_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
static void gst_pulsesrc_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
static void gst_pulsesrc_finalize(GObject *object);
static void gst_pulsesrc_dispose(GObject *object);

static gboolean gst_pulsesrc_open(GstAudioSrc *asrc);
static gboolean gst_pulsesrc_close(GstAudioSrc *asrc);

static gboolean gst_pulsesrc_prepare(GstAudioSrc *asrc, GstRingBufferSpec *spec);
static gboolean gst_pulsesrc_unprepare(GstAudioSrc *asrc);

static guint gst_pulsesrc_read(GstAudioSrc *asrc, gpointer data, guint length);
static guint gst_pulsesrc_delay(GstAudioSrc *asrc);

static GstStateChangeReturn gst_pulsesrc_change_state(GstElement *element, GstStateChange transition);

#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ENDIANNESS   "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ENDIANNESS   "BIG_ENDIAN, LITTLE_ENDIAN"
#endif

static gboolean gst_pulsesrc_interface_supported(GstImplementsInterface* iface, GType interface_type) {
    GstPulseSrc *this = GST_PULSESRC(iface);

    if (interface_type == GST_TYPE_MIXER && this->mixer)
        return TRUE;

    return FALSE;
}

static void gst_pulsesrc_implements_interface_init(GstImplementsInterfaceClass* klass) {
    klass->supported = gst_pulsesrc_interface_supported;
}

static void gst_pulsesrc_init_interfaces(GType type) {
    static const GInterfaceInfo implements_iface_info = {
        (GInterfaceInitFunc) gst_pulsesrc_implements_interface_init,
        NULL,
        NULL,
    };
    static const GInterfaceInfo mixer_iface_info = {
        (GInterfaceInitFunc) gst_pulsesrc_mixer_interface_init,
        NULL,
        NULL,
    };

    g_type_add_interface_static(type, GST_TYPE_IMPLEMENTS_INTERFACE, &implements_iface_info);
    g_type_add_interface_static(type, GST_TYPE_MIXER, &mixer_iface_info);
}

static void gst_pulsesrc_base_init(gpointer g_class) {

    static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE(
            "src",
            GST_PAD_SRC,
            GST_PAD_ALWAYS,
            GST_STATIC_CAPS(
                    "audio/x-raw-int, "
                    "endianness = (int) { " ENDIANNESS " }, "
                    "signed = (boolean) TRUE, "
                    "width = (int) 16, "
                    "depth = (int) 16, "
                    "rate = (int) [ 1, MAX ], "
                    "channels = (int) [ 1, 16 ];"

                    "audio/x-raw-int, "
                    "endianness = (int) { " ENDIANNESS " }, "
                    "signed = (boolean) TRUE, "
                    "width = (int) 32, "
                    "depth = (int) 32, "
                    "rate = (int) [ 1, MAX ], "
                    "channels = (int) [ 1, 16 ];"

                    "audio/x-raw-float, "
                    "endianness = (int) { " ENDIANNESS " }, "
                    "width = (int) 32, "
                    "rate = (int) [ 1, MAX ], "
                    "channels = (int) [ 1, 16 ];"

                    "audio/x-raw-int, "
                    "signed = (boolean) FALSE, "
                    "width = (int) 8, "
                    "depth = (int) 8, "
                    "rate = (int) [ 1, MAX ], "
                    "channels = (int) [ 1, 16 ];"

                    "audio/x-alaw, "
                    "rate = (int) [ 1, MAX], "
                    "channels = (int) [ 1, 16 ];"

                    "audio/x-mulaw, "
                    "rate = (int) [ 1, MAX], "
                    "channels = (int) [ 1, 16 ]"
            )
    );

    static const GstElementDetails details =
        GST_ELEMENT_DETAILS(
                "PulseAudio Audio Source",
                "Source/Audio",
                "Captures audio from a PulseAudio server",
                "Lennart Poettering");

    GstElementClass *element_class = GST_ELEMENT_CLASS(g_class);

    gst_element_class_set_details(element_class, &details);
    gst_element_class_add_pad_template(element_class, gst_static_pad_template_get(&pad_template));
}

static void gst_pulsesrc_class_init(
        gpointer g_class,
        gpointer class_data) {

    GObjectClass *gobject_class = G_OBJECT_CLASS(g_class);
    GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS(g_class);
    GstElementClass *gstelement_class = GST_ELEMENT_CLASS(g_class);
    parent_class = g_type_class_peek_parent(g_class);

    gstelement_class->change_state = GST_DEBUG_FUNCPTR(gst_pulsesrc_change_state);

    gobject_class->dispose = GST_DEBUG_FUNCPTR(gst_pulsesrc_dispose);
    gobject_class->finalize = GST_DEBUG_FUNCPTR(gst_pulsesrc_finalize);
    gobject_class->set_property = GST_DEBUG_FUNCPTR(gst_pulsesrc_set_property);
    gobject_class->get_property = GST_DEBUG_FUNCPTR(gst_pulsesrc_get_property);

    gstaudiosrc_class->open = GST_DEBUG_FUNCPTR(gst_pulsesrc_open);
    gstaudiosrc_class->close = GST_DEBUG_FUNCPTR(gst_pulsesrc_close);
    gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR(gst_pulsesrc_prepare);
    gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR(gst_pulsesrc_unprepare);
    gstaudiosrc_class->read = GST_DEBUG_FUNCPTR(gst_pulsesrc_read);
    gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR(gst_pulsesrc_delay);

    /* Overwrite GObject fields */
    g_object_class_install_property(
            gobject_class,
            PROP_SERVER,
            g_param_spec_string("server", "Server", "The PulseAudio server to connect to", NULL, G_PARAM_READWRITE));
    g_object_class_install_property(
            gobject_class,
            PROP_DEVICE,
            g_param_spec_string("device", "Source", "The PulseAudio source device to connect to", NULL, G_PARAM_READWRITE));
}

static void gst_pulsesrc_init(
        GTypeInstance * instance,
        gpointer g_class) {

    GstPulseSrc *pulsesrc = GST_PULSESRC(instance);
    int e;

    pulsesrc->server = pulsesrc->device = NULL;

    pulsesrc->context = NULL;
    pulsesrc->stream = NULL;

    pulsesrc->read_buffer = NULL;
    pulsesrc->read_buffer_length = 0;

    pulsesrc->mainloop = pa_threaded_mainloop_new();
    g_assert(pulsesrc->mainloop);

    e = pa_threaded_mainloop_start(pulsesrc->mainloop);
    g_assert(e == 0);

    pulsesrc->mixer = NULL;
}

static void gst_pulsesrc_destroy_stream(GstPulseSrc* pulsesrc) {
    if (pulsesrc->stream) {
        pa_stream_disconnect(pulsesrc->stream);
        pa_stream_unref(pulsesrc->stream);
        pulsesrc->stream = NULL;
    }
}

static void gst_pulsesrc_destroy_context(GstPulseSrc* pulsesrc) {

    gst_pulsesrc_destroy_stream(pulsesrc);

    if (pulsesrc->context) {
        pa_context_disconnect(pulsesrc->context);
        pa_context_unref(pulsesrc->context);
        pulsesrc->context = NULL;
    }
}

static void gst_pulsesrc_finalize(GObject * object) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(object);

    pa_threaded_mainloop_stop(pulsesrc->mainloop);

    gst_pulsesrc_destroy_context(pulsesrc);

    g_free(pulsesrc->server);
    g_free(pulsesrc->device);

    pa_threaded_mainloop_free(pulsesrc->mainloop);

    if (pulsesrc->mixer)
        gst_pulsemixer_ctrl_free(pulsesrc->mixer);

    G_OBJECT_CLASS(parent_class)->finalize(object);
}

static void gst_pulsesrc_dispose(GObject * object) {
    G_OBJECT_CLASS(parent_class)->dispose(object);
}

static void gst_pulsesrc_set_property(
        GObject * object,
        guint prop_id,
        const GValue * value,
        GParamSpec * pspec) {

    GstPulseSrc *pulsesrc = GST_PULSESRC(object);

    switch (prop_id) {
        case PROP_SERVER:
            g_free(pulsesrc->server);
            pulsesrc->server = g_value_dup_string(value);
            break;

        case PROP_DEVICE:
            g_free(pulsesrc->device);
            pulsesrc->device = g_value_dup_string(value);
            break;

        default:
            G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec);
            break;
    }
}

static void gst_pulsesrc_get_property(
        GObject * object,
        guint prop_id,
        GValue * value,
        GParamSpec * pspec) {

    GstPulseSrc *pulsesrc = GST_PULSESRC(object);

    switch(prop_id) {
        case PROP_SERVER:
            g_value_set_string(value, pulsesrc->server);
            break;

        case PROP_DEVICE:
            g_value_set_string(value, pulsesrc->device);
            break;

        default:
            G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec);
            break;
    }
}

static void gst_pulsesrc_context_state_cb(pa_context *c, void *userdata) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(userdata);

    switch (pa_context_get_state(c)) {
        case PA_CONTEXT_READY:
        case PA_CONTEXT_TERMINATED:
        case PA_CONTEXT_FAILED:
            pa_threaded_mainloop_signal(pulsesrc->mainloop, 0);
            break;

        case PA_CONTEXT_UNCONNECTED:
        case PA_CONTEXT_CONNECTING:
        case PA_CONTEXT_AUTHORIZING:
        case PA_CONTEXT_SETTING_NAME:
            break;
    }
}

static void gst_pulsesrc_stream_state_cb(pa_stream *s, void * userdata) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(userdata);

    switch (pa_stream_get_state(s)) {

        case PA_STREAM_READY:
        case PA_STREAM_FAILED:
        case PA_STREAM_TERMINATED:
            pa_threaded_mainloop_signal(pulsesrc->mainloop, 0);
            break;

        case PA_STREAM_UNCONNECTED:
        case PA_STREAM_CREATING:
            break;
    }
}

static void gst_pulsesrc_stream_request_cb(pa_stream *s, size_t length, void *userdata) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(userdata);

    pa_threaded_mainloop_signal(pulsesrc->mainloop, 0);
}

static gboolean gst_pulsesrc_open(GstAudioSrc *asrc) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(asrc);
    gchar *name = gst_pulse_client_name();

    pa_threaded_mainloop_lock(pulsesrc->mainloop);

    if (!(pulsesrc->context = pa_context_new(pa_threaded_mainloop_get_api(pulsesrc->mainloop), name))) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to create context"), (NULL));
        goto unlock_and_fail;
    }

    pa_context_set_state_callback(pulsesrc->context, gst_pulsesrc_context_state_cb, pulsesrc);

    if (pa_context_connect(pulsesrc->context, pulsesrc->server, 0, NULL) < 0) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
        goto unlock_and_fail;
    }

    /* Wait until the context is ready */
    pa_threaded_mainloop_wait(pulsesrc->mainloop);

    if (pa_context_get_state(pulsesrc->context) != PA_CONTEXT_READY) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
        goto unlock_and_fail;
    }

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);

    g_free(name);
    return TRUE;

unlock_and_fail:

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);

    g_free(name);
    return FALSE;
}

static gboolean gst_pulsesrc_close(GstAudioSrc *asrc) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(asrc);

    pa_threaded_mainloop_lock(pulsesrc->mainloop);
    gst_pulsesrc_destroy_context(pulsesrc);
    pa_threaded_mainloop_unlock(pulsesrc->mainloop);

    return TRUE;
}
static gboolean gst_pulsesrc_prepare(GstAudioSrc *asrc, GstRingBufferSpec *spec) {
    pa_buffer_attr buf_attr;
    pa_channel_map channel_map;

    GstPulseSrc *pulsesrc = GST_PULSESRC(asrc);

    if (!gst_pulse_fill_sample_spec(spec, &pulsesrc->sample_spec)) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, SETTINGS, ("Invalid sample specification."), (NULL));
        goto unlock_and_fail;
    }

    pa_threaded_mainloop_lock(pulsesrc->mainloop);

    if (!pulsesrc->context || pa_context_get_state(pulsesrc->context) != PA_CONTEXT_READY) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Bad context state: %s", pulsesrc->context ? pa_strerror(pa_context_errno(pulsesrc->context)) : NULL), (NULL));
        goto unlock_and_fail;
    }

    if (!(pulsesrc->stream = pa_stream_new(
                  pulsesrc->context,
                  "Record Stream",
                  &pulsesrc->sample_spec,
                  gst_pulse_gst_to_channel_map(&channel_map, spec)))) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to create stream: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
        goto unlock_and_fail;
    }

    pa_stream_set_state_callback(pulsesrc->stream, gst_pulsesrc_stream_state_cb, pulsesrc);
    pa_stream_set_read_callback(pulsesrc->stream, gst_pulsesrc_stream_request_cb, pulsesrc);

    memset(&buf_attr, 0, sizeof(buf_attr));
    buf_attr.maxlength = spec->segtotal*spec->segsize*2;
    buf_attr.fragsize = spec->segsize;

    if (pa_stream_connect_record(pulsesrc->stream, pulsesrc->device, &buf_attr, PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE|PA_STREAM_NOT_MONOTONOUS) < 0) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect stream: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
        goto unlock_and_fail;
    }

    /* Wait until the stream is ready */
    pa_threaded_mainloop_wait(pulsesrc->mainloop);

    if (pa_stream_get_state(pulsesrc->stream) != PA_STREAM_READY) {
        GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("Failed to connect stream: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
        goto unlock_and_fail;
    }

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);

    spec->bytes_per_sample = pa_frame_size(&pulsesrc->sample_spec);
    memset(spec->silence_sample, 0, spec->bytes_per_sample);

    return TRUE;

unlock_and_fail:

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);
    return FALSE;
}

static gboolean gst_pulsesrc_unprepare(GstAudioSrc * asrc) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(asrc);

    pa_threaded_mainloop_lock(pulsesrc->mainloop);
    gst_pulsesrc_destroy_stream(pulsesrc);

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);

    pulsesrc->read_buffer = NULL;
    pulsesrc->read_buffer_length = 0;

    return TRUE;
}

#define CHECK_DEAD_GOTO(pulsesrc, label) \
if (!(pulsesrc)->context || pa_context_get_state((pulsesrc)->context) != PA_CONTEXT_READY || \
    !(pulsesrc)->stream || pa_stream_get_state((pulsesrc)->stream) != PA_STREAM_READY) { \
    GST_ELEMENT_ERROR((pulsesrc), RESOURCE, FAILED, ("Disconnected: %s", (pulsesrc)->context ? pa_strerror(pa_context_errno((pulsesrc)->context)) : NULL), (NULL)); \
    goto label; \
}

static guint gst_pulsesrc_read(GstAudioSrc *asrc, gpointer data, guint length) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(asrc);
    size_t sum = 0;

    pa_threaded_mainloop_lock(pulsesrc->mainloop);

    CHECK_DEAD_GOTO(pulsesrc, unlock_and_fail);

    while (length > 0) {
        size_t l;

        if (!pulsesrc->read_buffer) {

            for (;;) {
                if (pa_stream_peek(pulsesrc->stream, &pulsesrc->read_buffer, &pulsesrc->read_buffer_length) < 0) {
                    GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("pa_stream_peek() failed: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
                    goto unlock_and_fail;
                }

                if (pulsesrc->read_buffer)
                    break;

                pa_threaded_mainloop_wait(pulsesrc->mainloop);

                CHECK_DEAD_GOTO(pulsesrc, unlock_and_fail);
            }
        }

        g_assert(pulsesrc->read_buffer && pulsesrc->read_buffer_length);

        l = pulsesrc->read_buffer_length > length ? length : pulsesrc->read_buffer_length;

        memcpy(data, pulsesrc->read_buffer, l);

        pulsesrc->read_buffer = (const guint8*) pulsesrc->read_buffer + l;
        pulsesrc->read_buffer_length -= l;

        data = (guint8*) data + l;
        length -= l;

        sum += l;

        if (pulsesrc->read_buffer_length <= 0) {

            if (pa_stream_drop(pulsesrc->stream) < 0) {
                GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("pa_stream_drop() failed: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
                goto unlock_and_fail;
            }

            pulsesrc->read_buffer = NULL;
            pulsesrc->read_buffer_length = 0;
        }
    }

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);

    return sum;

unlock_and_fail:
    pa_threaded_mainloop_unlock(pulsesrc->mainloop);
    return 0;
}

static guint gst_pulsesrc_delay(GstAudioSrc *asrc) {
    GstPulseSrc *pulsesrc = GST_PULSESRC(asrc);
    pa_usec_t t;
    int negative;

    pa_threaded_mainloop_lock(pulsesrc->mainloop);

    CHECK_DEAD_GOTO(pulsesrc, unlock_and_fail);

    if (pa_stream_get_latency(pulsesrc->stream, &t, &negative) < 0) {

        if (pa_context_errno(pulsesrc->context) != PA_ERR_NODATA) {
            GST_ELEMENT_ERROR(pulsesrc, RESOURCE, FAILED, ("pa_stream_get_latency() failed: %s", pa_strerror(pa_context_errno(pulsesrc->context))), (NULL));
            goto unlock_and_fail;
        }

        GST_WARNING("Not data while querying latency");
        t = 0;
    } else if (negative)
        t = 0;

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);

    return (guint) ((t * pulsesrc->sample_spec.rate) / 1000000LL);

unlock_and_fail:

    pa_threaded_mainloop_unlock(pulsesrc->mainloop);
    return 0;
}

static GstStateChangeReturn gst_pulsesrc_change_state(GstElement *element, GstStateChange transition) {
    GstPulseSrc *this = GST_PULSESRC(element);

    switch (transition) {
        case GST_STATE_CHANGE_NULL_TO_READY:

            if (!this->mixer)
                this->mixer = gst_pulsemixer_ctrl_new(this->server, this->device, GST_PULSEMIXER_SOURCE);

            break;

        case GST_STATE_CHANGE_READY_TO_NULL:

            if (this->mixer) {
                gst_pulsemixer_ctrl_free(this->mixer);
                this->mixer = NULL;
            }

            break;

        default:
            ;
    }

    if (GST_ELEMENT_CLASS(parent_class)->change_state)
        return GST_ELEMENT_CLASS(parent_class)->change_state(element, transition);

    return GST_STATE_CHANGE_SUCCESS;
}

GType gst_pulsesrc_get_type(void) {
    static GType pulsesrc_type = 0;

    if (!pulsesrc_type) {

        static const GTypeInfo pulsesrc_info = {
            sizeof(GstPulseSrcClass),
            gst_pulsesrc_base_init,
            NULL,
            gst_pulsesrc_class_init,
            NULL,
            NULL,
            sizeof(GstPulseSrc),
            0,
            gst_pulsesrc_init,
        };

        pulsesrc_type = g_type_register_static(
                GST_TYPE_AUDIO_SRC,
                "GstPulseSrc",
                &pulsesrc_info,
                0);

        gst_pulsesrc_init_interfaces(pulsesrc_type);
    }

    return pulsesrc_type;
}