annotate libao2/ao_kai.c @ 31139:9a2e299dc256

Add final missing bits of CineForm HD support on Linux (via the Windows DirectShow codec). Required changes: - codecs.conf entry (of course). - Allow opening files with ¡È.col¡É in the file name, just like ¡Èvp3¡É and ¡È.fpf¡É already was allowed. (CineForm expects to be able to do this, presumably for some color management code.) - In registry.c, fake a few registry keys that the codec expects the installer to have written. Also, change a few magic numbers (0, 2) to the appropriate constants (ERROR_SUCCESS, ERROR_FILE_NOT_FOUND) where appropriate, so the code is easier to follow. SMP works fine, but seemingly performs suboptimally (e.g., on my dual-core laptop, CineForm performs better if I lie to it and tell it I have four cores). I don't know if this is inherent in the codec, or some inefficiency in the emulated synchronization primitives.

/*
 * OS/2 KAI audio output driver
 *
 * Copyright (c) 2010 by KO Myung-Hun ([e-mail])
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#define INCL_DOS
#define INCL_DOSERRORS
#include <os2.h>

#include <stdio.h>
#include <stdlib.h>
#include <sys/time.h>
#include <float.h>

#include <kai.h>

#include "config.h"
#include "libaf/af_format.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
#include "libvo/fastmemcpy.h"
#include "subopt-helper.h"
#include "libavutil/avutil.h"
#include "libavutil/fifo.h"

static const ao_info_t info = {
    "KAI audio output",
    "kai",
    "KO Myung-Hun <[e-mail]>",
    ""
};

LIBAO_EXTERN(kai)

#define OUTBURST_SAMPLES        512
#define DEFAULT_SAMPLES         (OUTBURST_SAMPLES << 2)

#define CHUNK_SIZE  ao_data.outburst

static AVFifoBuffer *m_audioBuf;

static int m_nBufSize = 0;

static volatile int m_fQuit = FALSE;

static KAISPEC m_kaiSpec;

static HKAI m_hkai;

static int write_buffer(unsigned char *data, int len)
{
    int nFree = av_fifo_space(m_audioBuf);

    len = FFMIN(len, nFree);

    return av_fifo_generic_write(m_audioBuf, data, len, NULL);
}

static int read_buffer(unsigned char *data, int len)
{
    int nBuffered = av_fifo_size(m_audioBuf);

    len = FFMIN(len, nBuffered);

    av_fifo_generic_read(m_audioBuf, data, len, NULL);
    return len;
}

// end ring buffer stuff

static ULONG APIENTRY kai_audio_callback(PVOID pCBData, PVOID pBuffer,
                                         ULONG ulSize)
{
    int nReadLen;

    nReadLen = read_buffer(pBuffer, ulSize);
    if (nReadLen < ulSize && !m_fQuit) {
        memset((uint8_t *)pBuffer + nReadLen, m_kaiSpec.bSilence, ulSize - nReadLen);
        nReadLen = ulSize;
    }

    return nReadLen;
}

// to set/get/query special features/parameters
static int control(int cmd, void *arg)
{
    switch (cmd) {
    case AOCONTROL_GET_VOLUME:
        {
        ao_control_vol_t *vol = arg;

        vol->left = vol->right = kaiGetVolume(m_hkai, MCI_STATUS_AUDIO_ALL);

        return CONTROL_OK;
        }

    case AOCONTROL_SET_VOLUME:
        {
        int mid;
        ao_control_vol_t *vol = arg;

        mid = (vol->left + vol->right) / 2;
        kaiSetVolume(m_hkai, MCI_SET_AUDIO_ALL, mid);

        return CONTROL_OK;
        }
    }

    return CONTROL_UNKNOWN;
}

static void print_help(void)
{
    mp_msg(MSGT_AO, MSGL_FATAL,
           "\n-ao kai commandline help:\n"
           "Example: mplayer -ao kai:noshare\n"
           "    open audio in exclusive mode\n"
           "\nOptions:\n"
           "    uniaud\n"
           "        Use UNIAUD audio driver\n"
           "    dart\n"
           "        Use DART audio driver\n"
           "    (no)share\n"
           "        Open audio in shareable or exclusive mode\n"
           "    bufsize=<size>\n"
           "        Set buffer size to <size> in samples(default: 2048)\n");
}

// open & set up audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags)
{
    int     fUseUniaud = 0;
    int     fUseDart = 0;
    int     fShare = 1;
    ULONG   kaiMode;
    KAICAPS kc;
    int     nSamples = DEFAULT_SAMPLES;
    int     nBytesPerSample;
    KAISPEC ksWanted;

    const opt_t subopts[] = {
        {"uniaud",  OPT_ARG_BOOL, &fUseUniaud, NULL},
        {"dart",    OPT_ARG_BOOL, &fUseDart,   NULL},
        {"share",   OPT_ARG_BOOL, &fShare,     NULL},
        {"bufsize", OPT_ARG_INT,  &nSamples,   int_non_neg},
        {NULL}
    };

    const char *audioDriver[] = {"DART", "UNIAUD",};

    if (subopt_parse(ao_subdevice, subopts) != 0) {
        print_help();
        return 0;
    }

    if (fUseUniaud && fUseDart)
        mp_msg(MSGT_VO, MSGL_WARN,"KAI: Multiple mode specified!!!\n");

    if (fUseUniaud)
        kaiMode = KAIM_UNIAUD;
    else if (fUseDart)
        kaiMode = KAIM_DART;
    else
        kaiMode = KAIM_AUTO;

    if (kaiInit(kaiMode)) {
        mp_msg(MSGT_VO, MSGL_ERR, "KAI: Init failed!!!\n");
        return 0;
    }

    kaiCaps(&kc);
    mp_msg(MSGT_AO, MSGL_V, "KAI: selected audio driver = %s\n",
           audioDriver[kc.ulMode - 1]);
    mp_msg(MSGT_AO, MSGL_V, "KAI: PDD name = %s, maximum channels = %lu\n",
           kc.szPDDName, kc.ulMaxChannels);

    if (!nSamples)
        nSamples = DEFAULT_SAMPLES;

    mp_msg(MSGT_AO, MSGL_V, "KAI: open in %s mode, buffer size = %d sample(s)\n",
           fShare ? "shareable" : "exclusive", nSamples);

    switch (format) {
    case AF_FORMAT_S16_LE:
    case AF_FORMAT_S8:
        break;

    default:
        format = AF_FORMAT_S16_LE;
        mp_msg(MSGT_AO, MSGL_V, "KAI: format %s not supported defaulting to Signed 16-bit Little-Endian\n",
               af_fmt2str_short(format));
        break;
    }

    nBytesPerSample = (af_fmt2bits(format) >> 3) * channels;

    ksWanted.usDeviceIndex      = 0;
    ksWanted.ulType             = KAIT_PLAY;
    ksWanted.ulBitsPerSample    = af_fmt2bits(format);
    ksWanted.ulSamplingRate     = rate;
    ksWanted.ulDataFormat       = MCI_WAVE_FORMAT_PCM;
    ksWanted.ulChannels         = channels;
    ksWanted.ulNumBuffers       = 2;
    ksWanted.ulBufferSize       = nBytesPerSample * nSamples;
    ksWanted.fShareable         = fShare;
    ksWanted.pfnCallBack        = kai_audio_callback;
    ksWanted.pCallBackData      = NULL;

    if (kaiOpen(&ksWanted, &m_kaiSpec, &m_hkai)) {
        mp_msg(MSGT_VO, MSGL_ERR, "KAI: Open failed!!!\n");
        return 0;
    }

    mp_msg(MSGT_AO, MSGL_V, "KAI: obtained buffer count = %lu, size = %lu bytes\n",
           m_kaiSpec.ulNumBuffers, m_kaiSpec.ulBufferSize);

    m_fQuit = FALSE;

    ao_data.channels    = channels;
    ao_data.samplerate  = rate;
    ao_data.format      = format;
    ao_data.bps         = nBytesPerSample * rate;
    ao_data.outburst    = nBytesPerSample * OUTBURST_SAMPLES;
    ao_data.buffersize  = m_kaiSpec.ulBufferSize;

    m_nBufSize = (m_kaiSpec.ulBufferSize * m_kaiSpec.ulNumBuffers) << 2;

    // multiple of CHUNK_SIZE
    m_nBufSize = (m_nBufSize / CHUNK_SIZE) * CHUNK_SIZE;

    // and one more chunk plus round up
    m_nBufSize += 2 * CHUNK_SIZE;

    mp_msg(MSGT_AO, MSGL_V, "KAI: internal audio buffer size = %d bytes\n",
           m_nBufSize);

    m_audioBuf = av_fifo_alloc(m_nBufSize);

    kaiPlay(m_hkai);

    // might cause PM DLLs to be loaded which incorrectly enable SIG_FPE,
    // which AAC decoding might trigger.
    // so, mask off all floating-point exceptions.
    _control87(MCW_EM, MCW_EM);

    return 1;
}

// close audio device
static void uninit(int immed)
{
    m_fQuit = TRUE;

    if (!immed)
        while (kaiStatus(m_hkai) & KAIS_PLAYING)
            DosSleep(1);

    kaiClose(m_hkai);

    kaiDone();

    av_fifo_free(m_audioBuf);
}

// stop playing and empty buffers (for seeking/pause)
static void reset(void)
{
    kaiPause(m_hkai);

    // Reset ring-buffer state
    av_fifo_reset(m_audioBuf);

    kaiResume(m_hkai);
}

// stop playing, keep buffers (for pause)
static void audio_pause(void)
{
    kaiPause(m_hkai);
}

// resume playing, after audio_pause()
static void audio_resume(void)
{
    kaiResume(m_hkai);
}

// return: how many bytes can be played without blocking
static int get_space(void)
{
    return av_fifo_space(m_audioBuf);
}

// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void *data, int len, int flags)
{

    if (!(flags & AOPLAY_FINAL_CHUNK))
        len = (len / ao_data.outburst) * ao_data.outburst;

    return write_buffer(data, len);
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(void)
{
    int nBuffered = av_fifo_size(m_audioBuf); // could be less

    return (float)nBuffered / (float)ao_data.bps;
}