/*
 * This file is part of mpv.
 *
 * mpv 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.
 *
 * mpv 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 mpv.  If not, see <http://www.gnu.org/licenses/>.
 */

/*
 * This file contains functions interacting with the CoreAudio framework
 * that are not specific to the AUHAL. These are split in a separate file for
 * the sake of readability. In the future the could be used by other AOs based
 * on CoreAudio but not the AUHAL (such as using AudioQueue services).
 */

#include "audio/out/ao_coreaudio_utils.h"
#include "osdep/timer.h"
#include "osdep/endian.h"
#include "osdep/semaphore.h"
#include "audio/format.h"

#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
#include "audio/out/ao_coreaudio_properties.h"
#include <CoreAudio/HostTime.h>
#else
#include <mach/mach_time.h>
#endif

#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
static bool ca_is_output_device(struct ao *ao, AudioDeviceID dev)
{
    size_t n_buffers;
    AudioBufferList *buffers;
    const ca_scope scope = kAudioDevicePropertyStreamConfiguration;
    OSStatus err = CA_GET_ARY_O(dev, scope, &buffers, &n_buffers);
    if (err != noErr)
        return false;
    talloc_free(buffers);
    return n_buffers > 0;
}

void ca_get_device_list(struct ao *ao, struct ao_device_list *list)
{
    AudioDeviceID *devs;
    size_t n_devs;
    OSStatus err =
        CA_GET_ARY(kAudioObjectSystemObject, kAudioHardwarePropertyDevices,
                   &devs, &n_devs);
    CHECK_CA_ERROR("Failed to get list of output devices.");
    for (int i = 0; i < n_devs; i++) {
        if (!ca_is_output_device(ao, devs[i]))
            continue;
        void *ta_ctx = talloc_new(NULL);
        char *name;
        char *desc;
        err = CA_GET_STR(devs[i], kAudioDevicePropertyDeviceUID, &name);
        if (err != noErr) {
            MP_VERBOSE(ao, "skipping device %d, which has no UID\n", i);
            talloc_free(ta_ctx);
            continue;
        }
        talloc_steal(ta_ctx, name);
        err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &desc);
        if (err != noErr)
            desc = talloc_strdup(NULL, "Unknown");
        talloc_steal(ta_ctx, desc);
        ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc});
        talloc_free(ta_ctx);
    }
    talloc_free(devs);
coreaudio_error:
    return;
}

OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device)
{
    OSStatus err = noErr;
    *device = kAudioObjectUnknown;

    if (name && name[0]) {
        CFStringRef uid = cfstr_from_cstr(name);
        AudioValueTranslation v = (AudioValueTranslation) {
            .mInputData = &uid,
            .mInputDataSize = sizeof(CFStringRef),
            .mOutputData = device,
            .mOutputDataSize = sizeof(*device),
        };
        uint32_t size = sizeof(AudioValueTranslation);
        AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
            .mSelector = kAudioHardwarePropertyDeviceForUID,
            .mScope    = kAudioObjectPropertyScopeGlobal,
            .mElement  = kAudioObjectPropertyElementMaster,
        };
        err = AudioObjectGetPropertyData(
            kAudioObjectSystemObject, &p_addr, 0, 0, &size, &v);
        CFRelease(uid);
        CHECK_CA_ERROR("unable to query for device UID");

        uint32_t is_alive = 1;
        err = CA_GET(*device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
        CHECK_CA_ERROR("could not check whether device is alive (invalid device?)");

        if (!is_alive)
            MP_WARN(ao, "device is not alive!\n");
    } else {
        // device not set by user, get the default one
        err = CA_GET(kAudioObjectSystemObject,
                     kAudioHardwarePropertyDefaultOutputDevice,
                     device);
        CHECK_CA_ERROR("could not get default audio device");
    }

    if (mp_msg_test(ao->log, MSGL_V)) {
        char *desc;
        OSStatus err2 = CA_GET_STR(*device, kAudioObjectPropertyName, &desc);
        if (err2 == noErr) {
            MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
                           desc, *device);
            talloc_free(desc);
        }
    }

coreaudio_error:
    return err;
}
#endif

bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
{
    if (code == noErr) return true;

    if (ao)
        mp_msg(ao->log, level, "%s (%s/%d)\n", message, mp_tag_str(code), (int)code);

    return false;
}

static void ca_fill_asbd_raw(AudioStreamBasicDescription *asbd, int mp_format,
                             int samplerate, int num_channels)
{
    asbd->mSampleRate       = samplerate;
    // Set "AC3" for other spdif formats too - unknown if that works.
    asbd->mFormatID         = af_fmt_is_spdif(mp_format) ?
                              kAudioFormat60958AC3 :
                              kAudioFormatLinearPCM;
    asbd->mChannelsPerFrame = num_channels;
    asbd->mBitsPerChannel   = af_fmt_to_bytes(mp_format) * 8;
    asbd->mFormatFlags      = kAudioFormatFlagIsPacked;

    int channels_per_buffer = num_channels;
    if (af_fmt_is_planar(mp_format)) {
        asbd->mFormatFlags |= kAudioFormatFlagIsNonInterleaved;
        channels_per_buffer = 1;
    }

    if (af_fmt_is_float(mp_format)) {
        asbd->mFormatFlags |= kAudioFormatFlagIsFloat;
    } else if (!af_fmt_is_unsigned(mp_format)) {
        asbd->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
    }

    if (BYTE_ORDER == BIG_ENDIAN)
        asbd->mFormatFlags |= kAudioFormatFlagIsBigEndian;

    asbd->mFramesPerPacket = 1;
    asbd->mBytesPerPacket = asbd->mBytesPerFrame =
        asbd->mFramesPerPacket * channels_per_buffer *
        (asbd->mBitsPerChannel / 8);
}

void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
{
    ca_fill_asbd_raw(asbd, ao->format, ao->samplerate, ao->channels.num);
}

bool ca_formatid_is_compressed(uint32_t formatid)
{
    switch (formatid)
    case 'IAC3':
    case 'iac3':
    case  kAudioFormat60958AC3:
    case  kAudioFormatAC3:
        return true;
    return false;
}

// This might be wrong, but for now it's sufficient for us.
static uint32_t ca_normalize_formatid(uint32_t formatID)
{
    return ca_formatid_is_compressed(formatID) ? kAudioFormat60958AC3 : formatID;
}

bool ca_asbd_equals(const AudioStreamBasicDescription *a,
                    const AudioStreamBasicDescription *b)
{
    int flags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsFloat |
                kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsBigEndian;
    bool spdif = ca_formatid_is_compressed(a->mFormatID) &&
                 ca_formatid_is_compressed(b->mFormatID);

    return (a->mFormatFlags & flags) == (b->mFormatFlags & flags) &&
           a->mBitsPerChannel == b->mBitsPerChannel &&
           ca_normalize_formatid(a->mFormatID) ==
                ca_normalize_formatid(b->mFormatID) &&
           (spdif || a->mBytesPerPacket == b->mBytesPerPacket) &&
           (spdif || a->mChannelsPerFrame == b->mChannelsPerFrame) &&
           a->mSampleRate == b->mSampleRate;
}

// Return the AF_FORMAT_* (AF_FORMAT_S16 etc.) corresponding to the asbd.
int ca_asbd_to_mp_format(const AudioStreamBasicDescription *asbd)
{
    for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
        AudioStreamBasicDescription mp_asbd = {0};
        ca_fill_asbd_raw(&mp_asbd, fmt, asbd->mSampleRate, asbd->mChannelsPerFrame);
        if (ca_asbd_equals(&mp_asbd, asbd))
            return af_fmt_is_spdif(fmt) ? AF_FORMAT_S_AC3 : fmt;
    }
    return 0;
}

void ca_print_asbd(struct ao *ao, const char *description,
                   const AudioStreamBasicDescription *asbd)
{
    uint32_t flags  = asbd->mFormatFlags;
    char *format    = mp_tag_str(asbd->mFormatID);
    int mpfmt       = ca_asbd_to_mp_format(asbd);

    MP_VERBOSE(ao,
        "%s %7.1fHz %" PRIu32 "bit %s "
        "[%" PRIu32 "][%" PRIu32 "bpp][%" PRIu32 "fbp]"
        "[%" PRIu32 "bpf][%" PRIu32 "ch] "
        "%s %s %s%s%s%s (%s)\n",
        description, asbd->mSampleRate, asbd->mBitsPerChannel, format,
        asbd->mFormatFlags, asbd->mBytesPerPacket, asbd->mFramesPerPacket,
        asbd->mBytesPerFrame, asbd->mChannelsPerFrame,
        (flags & kAudioFormatFlagIsFloat) ? "float" : "int",
        (flags & kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
        (flags & kAudioFormatFlagIsSignedInteger) ? "S" : "U",
        (flags & kAudioFormatFlagIsPacked) ? " packed" : "",
        (flags & kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
        (flags & kAudioFormatFlagIsNonInterleaved) ? " P" : "",
        mpfmt ? af_fmt_to_str(mpfmt) : "-");
}

// Return whether new is an improvement over old. Assume a higher value means
// better quality, and we always prefer the value closest to the requested one,
// which is still larger than the requested one.
// Equal values prefer the new one (so ca_asbd_is_better() checks other params).
static bool value_is_better(double req, double old, double new)
{
    if (new >= req) {
        return old < req || new <= old;
    } else {
        return old < req && new >= old;
    }
}

// Return whether new is an improvement over old (req is the requested format).
bool ca_asbd_is_better(AudioStreamBasicDescription *req,
                       AudioStreamBasicDescription *old,
                       AudioStreamBasicDescription *new)
{
    if (new->mChannelsPerFrame > MP_NUM_CHANNELS)
        return false;
    if (old->mChannelsPerFrame > MP_NUM_CHANNELS)
        return true;
    if (req->mFormatID != new->mFormatID)
        return false;
    if (req->mFormatID != old->mFormatID)
        return true;

    if (!value_is_better(req->mBitsPerChannel, old->mBitsPerChannel,
                         new->mBitsPerChannel))
        return false;

    if (!value_is_better(req->mSampleRate, old->mSampleRate, new->mSampleRate))
        return false;

    if (!value_is_better(req->mChannelsPerFrame, old->mChannelsPerFrame,
                         new->mChannelsPerFrame))
        return false;

    return true;
}

int64_t ca_frames_to_ns(struct ao *ao, uint32_t frames)
{
    return MP_TIME_S_TO_NS(frames / (double)ao->samplerate);
}

int64_t ca_get_latency(const AudioTimeStamp *ts)
{
#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
    uint64_t out = AudioConvertHostTimeToNanos(ts->mHostTime);
    uint64_t now = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());

    if (now > out)
        return 0;

    return out - now;
#else
    static mach_timebase_info_data_t timebase;
    if (timebase.denom == 0)
        mach_timebase_info(&timebase);

    uint64_t out = ts->mHostTime;
    uint64_t now = mach_absolute_time();

    if (now > out)
        return 0;

    return (out - now) * timebase.numer / timebase.denom;
#endif
}

#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
bool ca_stream_supports_compressed(struct ao *ao, AudioStreamID stream)
{
    AudioStreamRangedDescription *formats = NULL;
    size_t n_formats;

    OSStatus err =
        CA_GET_ARY(stream, kAudioStreamPropertyAvailablePhysicalFormats,
                   &formats, &n_formats);

    CHECK_CA_ERROR("Could not get number of stream formats.");

    for (int i = 0; i < n_formats; i++) {
        AudioStreamBasicDescription asbd = formats[i].mFormat;

        ca_print_asbd(ao, "- ", &asbd);

        if (ca_formatid_is_compressed(asbd.mFormatID)) {
            talloc_free(formats);
            return true;
        }
    }

    talloc_free(formats);
coreaudio_error:
    return false;
}

OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid)
{
    *pid = getpid();
    OSStatus err = CA_SET(device, kAudioDevicePropertyHogMode, pid);
    if (err != noErr)
        *pid = -1;

    return err;
}

OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid)
{
    if (*pid == getpid()) {
        *pid = -1;
        return CA_SET(device, kAudioDevicePropertyHogMode, &pid);
    }
    return noErr;
}

static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
                                 uint32_t val, bool *changed)
{
    *changed = false;

    AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
        .mSelector = kAudioDevicePropertySupportsMixing,
        .mScope    = kAudioObjectPropertyScopeGlobal,
        .mElement  = kAudioObjectPropertyElementMaster,
    };

    if (AudioObjectHasProperty(device, &p_addr)) {
        OSStatus err;
        Boolean writeable = 0;
        err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing,
                          &writeable);

        if (!CHECK_CA_WARN("can't tell if mixing property is settable")) {
            return err;
        }

        if (!writeable)
            return noErr;

        err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val);
        if (err != noErr)
            return err;

        if (!CHECK_CA_WARN("can't set mix mode")) {
            return err;
        }

        *changed = true;
    }

    return noErr;
}

OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed)
{
    return ca_change_mixing(ao, device, 0, changed);
}

OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed)
{
    if (changed) {
        bool dont_care = false;
        return ca_change_mixing(ao, device, 1, &dont_care);
    }

    return noErr;
}

int64_t ca_get_device_latency_ns(struct ao *ao, AudioDeviceID device)
{
    uint32_t latency_frames = 0;
    uint32_t latency_properties[] = {
        kAudioDevicePropertyLatency,
        kAudioDevicePropertyBufferFrameSize,
        kAudioDevicePropertySafetyOffset,
    };
    for (int n = 0; n < MP_ARRAY_SIZE(latency_properties); n++) {
        uint32_t temp;
        OSStatus err = CA_GET_O(device, latency_properties[n], &temp);
        CHECK_CA_WARN("cannot get device latency");
        if (err == noErr) {
            latency_frames += temp;
            MP_VERBOSE(ao, "Latency property %s: %d frames\n",
                       mp_tag_str(latency_properties[n]), (int)temp);
        }
    }

    double sample_rate = ao->samplerate;
    OSStatus err = CA_GET_O(device, kAudioDevicePropertyNominalSampleRate,
                            &sample_rate);
    CHECK_CA_WARN("cannot get device sample rate, falling back to AO sample rate!");
    if (err == noErr) {
        MP_VERBOSE(ao, "Device sample rate: %f\n", sample_rate);
    }

    return MP_TIME_S_TO_NS(latency_frames / sample_rate);
}

static OSStatus ca_change_format_listener(
    AudioObjectID object, uint32_t n_addresses,
    const AudioObjectPropertyAddress addresses[],
    void *data)
{
    mp_sem_t *sem = data;
    mp_sem_post(sem);
    return noErr;
}

bool ca_change_physical_format_sync(struct ao *ao, AudioStreamID stream,
                                    AudioStreamBasicDescription change_format)
{
    OSStatus err = noErr;
    bool format_set = false;

    ca_print_asbd(ao, "setting stream physical format:", &change_format);

    mp_sem_t wakeup;
    if (mp_sem_init(&wakeup, 0, 0))
        MP_HANDLE_OOM(0);

    AudioStreamBasicDescription prev_format;
    err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
    CHECK_CA_ERROR("can't get current physical format");

    ca_print_asbd(ao, "format in use before switching:", &prev_format);

    /* Install the callback. */
    AudioObjectPropertyAddress p_addr = {
        .mSelector = kAudioStreamPropertyPhysicalFormat,
        .mScope    = kAudioObjectPropertyScopeGlobal,
        .mElement  = kAudioObjectPropertyElementMaster,
    };

    err = AudioObjectAddPropertyListener(stream, &p_addr,
                                         ca_change_format_listener,
                                         &wakeup);
    CHECK_CA_ERROR("can't add property listener during format change");

    /* Change the format. */
    err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &change_format);
    CHECK_CA_WARN("error changing physical format");

    /* The AudioStreamSetProperty is not only asynchronous,
     * it is also not Atomic, in its behaviour. */
    int64_t wait_until = mp_time_ns() + MP_TIME_S_TO_NS(2);
    AudioStreamBasicDescription actual_format = {0};
    while (1) {
        err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
        if (!CHECK_CA_WARN("could not retrieve physical format"))
            break;

        format_set = ca_asbd_equals(&change_format, &actual_format);
        if (format_set)
            break;

        if (mp_sem_timedwait(&wakeup, wait_until)) {
            MP_VERBOSE(ao, "reached timeout\n");
            break;
        }
    }

    ca_print_asbd(ao, "actual format in use:", &actual_format);

    if (!format_set) {
        MP_WARN(ao, "changing physical format failed\n");
        // Some drivers just fuck up and get into a broken state. Restore the
        // old format in this case.
        err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
        CHECK_CA_WARN("error restoring physical format");
    }

    err = AudioObjectRemovePropertyListener(stream, &p_addr,
                                            ca_change_format_listener,
                                            &wakeup);
    CHECK_CA_ERROR("can't remove property listener");

coreaudio_error:
    mp_sem_destroy(&wakeup);
    return format_set;
}
#endif