annotate libao2/ao_pcm.c @ 36883:e299b6fd7eaa

Fix image size restrictions set by r35242. Luma blocks are 16x16, but chroma blocks are (two interweaved) 8x16. Since chroma is at quarter resolution it means that the image width must be multiple of 16, while height must be multiple of 32.

/*
 * PCM audio output driver
 *
 * 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.
 */

#include "config.h"

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

#include "libavutil/common.h"
#include "mpbswap.h"
#include "subopt-helper.h"
#include "libaf/af_format.h"
#include "libaf/reorder_ch.h"
#include "libvo/video_out.h" /* only for vo_pts */
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
#include "help_mp.h"

#ifdef __MINGW32__
// for GetFileType to detect pipes
#include <windows.h>
#include <io.h>
#endif

static const ao_info_t info =
{
    "RAW PCM/WAVE file writer audio output",
    "pcm",
    "Atmosfear",
    ""
};

LIBAO_EXTERN(pcm)

static char *ao_outputfilename = NULL;
static int ao_pcm_waveheader = 1;
static int fast = 0;

#define WAV_ID_RIFF 0x46464952 /* "RIFF" */
#define WAV_ID_WAVE 0x45564157 /* "WAVE" */
#define WAV_ID_FMT  0x20746d66 /* "fmt " */
#define WAV_ID_DATA 0x61746164 /* "data" */
#define WAV_ID_PCM  0x0001
#define WAV_ID_FLOAT_PCM  0x0003
#define WAV_ID_FORMAT_EXTENSIBLE 0xfffe

/* init with default values */
static uint64_t data_length;
static FILE *fp = NULL;


static void fput16le(uint16_t val, FILE *fp) {
    uint8_t bytes[2] = {val, val >> 8};
    fwrite(bytes, 1, 2, fp);
}

static void fput32le(uint32_t val, FILE *fp) {
    uint8_t bytes[4] = {val, val >> 8, val >> 16, val >> 24};
    fwrite(bytes, 1, 4, fp);
}

static void write_wave_header(FILE *fp, uint64_t data_length) {
    int use_waveex = (ao_data.channels >= 5 && ao_data.channels <= 8);
    uint16_t fmt = (ao_data.format == AF_FORMAT_FLOAT_LE) ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
    uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
    int bits = af_fmt2bits(ao_data.format);

    // Master RIFF chunk
    fput32le(WAV_ID_RIFF, fp);
    // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size + data chunk hdr (8) + data length
    fput32le(12 + fmt_chunk_size + 8 + data_length, fp);
    fput32le(WAV_ID_WAVE, fp);

    // Format chunk
    fput32le(WAV_ID_FMT, fp);
    fput32le(fmt_chunk_size, fp);
    fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
    fput16le(ao_data.channels, fp);
    fput32le(ao_data.samplerate, fp);
    fput32le(ao_data.bps, fp);
    fput16le(ao_data.channels * (bits / 8), fp);
    fput16le(bits, fp);

    if (use_waveex) {
        // Extension chunk
        fput16le(22, fp);
        fput16le(bits, fp);
        switch (ao_data.channels) {
            case 5:
                fput32le(0x0607, fp); // L R C Lb Rb
                break;
            case 6:
                fput32le(0x060f, fp); // L R C Lb Rb LFE
                break;
            case 7:
                fput32le(0x0727, fp); // L R C Cb Ls Rs LFE
                break;
            case 8:
                fput32le(0x063f, fp); // L R C Lb Rb Ls Rs LFE
                break;
        }
        // 2 bytes format + 14 bytes guid
        fput32le(fmt, fp);
        fput32le(0x00100000, fp);
        fput32le(0xAA000080, fp);
        fput32le(0x719B3800, fp);
    }

    // Data chunk
    fput32le(WAV_ID_DATA, fp);
    fput32le(data_length, fp);
}

// to set/get/query special features/parameters
static int control(int cmd,void *arg){
    return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
    const opt_t subopts[] = {
        {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
        {"file",       OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
        {"fast",       OPT_ARG_BOOL, &fast, NULL},
        {NULL}
    };
    // set defaults
    ao_pcm_waveheader = 1;

    if (subopt_parse(ao_subdevice, subopts) != 0) {
        return 0;
    }
    if (!ao_outputfilename){
        ao_outputfilename =
            strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
    }

    if (ao_pcm_waveheader)
    {
        // WAV files must have one of the following formats

        switch(format){
        case AF_FORMAT_U8:
        case AF_FORMAT_S16_LE:
        case AF_FORMAT_S24_LE:
        case AF_FORMAT_S32_LE:
        case AF_FORMAT_FLOAT_LE:
        case AF_FORMAT_AC3_BE:
        case AF_FORMAT_AC3_LE:
             break;
        default:
            format = AF_FORMAT_S16_LE;
            break;
        }
    }

    ao_data.outburst = 65536;
    ao_data.buffersize= 2*65536;
    ao_data.channels=channels;
    ao_data.samplerate=rate;
    ao_data.format=format;
    ao_data.bps=channels*rate*(af_fmt2bits(format)/8);

    mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
           (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
           (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
    mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);

    fp = fopen(ao_outputfilename, "wb");
    if(fp) {
        if(ao_pcm_waveheader){ /* Reserve space for wave header */
            write_wave_header(fp, 0x7ffff000);
        }
        return 1;
    }
    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
               ao_outputfilename);
    return 0;
}

// close audio device
static void uninit(int immed){

    if(ao_pcm_waveheader){ /* Rewrite wave header */
        int broken_seek = 0;
#ifdef __MINGW32__
        // Windows, in its usual idiocy "emulates" seeks on pipes so it always looks
        // like they work. So we have to detect them brute-force.
        broken_seek = GetFileType((HANDLE)_get_osfhandle(_fileno(fp))) != FILE_TYPE_DISK;
#endif
        if (broken_seek || fseek(fp, 0, SEEK_SET) != 0)
            mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
        else {
            if (data_length > 0xfffff000) {
                mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
                data_length = 0xfffff000;
            }
            write_wave_header(fp, data_length);
        }
    }
    fclose(fp);
    free(ao_outputfilename);
    ao_outputfilename = NULL;
}

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

}

// stop playing, keep buffers (for pause)
static void audio_pause(void)
{
    // for now, just call reset();
    reset();
}

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

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

    if(vo_pts)
        return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
    return ao_data.outburst;
}

// 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 (ao_data.channels == 5 || ao_data.channels == 6 || ao_data.channels == 8) {
        int frame_size = af_fmt2bits(ao_data.format) / 8;
        len -= len % (frame_size * ao_data.channels);
        reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
                            AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
                            ao_data.channels,
                            len / frame_size, frame_size);
    }

    //printf("PCM: Writing chunk!\n");
    fwrite(data,len,1,fp);

    if(ao_pcm_waveheader)
        data_length += len;

    return len;
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(void){

    return 0.0;
}