view libao2/ao_pcm.c @ 29288:4a1c217a844b

In all demux_info_add calls change "name" to "title". Currently "name" and "title" are both used at random, this makes it consistent. "title" was chosen because it is less ambiguous and also the get_meta_title slave mode command uses that (there is no get_meta_name command).

/*
 * 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 "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>
#endif

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

LIBAO_EXTERN(pcm)

extern int vo_pts;

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

struct WaveHeader
{
    uint32_t riff;
    uint32_t file_length;
    uint32_t wave;
    uint32_t fmt;
    uint32_t fmt_length;
    uint16_t fmt_tag;
    uint16_t channels;
    uint32_t sample_rate;
    uint32_t bytes_per_second;
    uint16_t block_align;
    uint16_t bits;
    uint32_t data;
    uint32_t data_length;
};

/* init with default values */
static struct WaveHeader wavhdr;
static uint64_t data_length;

static FILE *fp = NULL;

// 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){
    int bits;
    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");
    }

    bits=8;
    switch(format){
    case AF_FORMAT_S32_BE:
        format=AF_FORMAT_S32_LE;
    case AF_FORMAT_S32_LE:
        bits=32;
        break;
    case AF_FORMAT_FLOAT_BE:
        format=AF_FORMAT_FLOAT_LE;
    case AF_FORMAT_FLOAT_LE:
        bits=32;
        break;
    case AF_FORMAT_S8:
        format=AF_FORMAT_U8;
    case AF_FORMAT_U8:
        break;
    case AF_FORMAT_AC3:
        bits=16;
        break;
    default:
        format=AF_FORMAT_S16_LE;
        bits=16;
        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*(bits/8);

    wavhdr.riff = le2me_32(WAV_ID_RIFF);
    wavhdr.wave = le2me_32(WAV_ID_WAVE);
    wavhdr.fmt = le2me_32(WAV_ID_FMT);
    wavhdr.fmt_length = le2me_32(16);
    wavhdr.fmt_tag = le2me_16(format == AF_FORMAT_FLOAT_LE ? WAV_ID_FLOAT_PCM : WAV_ID_PCM);
    wavhdr.channels = le2me_16(ao_data.channels);
    wavhdr.sample_rate = le2me_32(ao_data.samplerate);
    wavhdr.bytes_per_second = le2me_32(ao_data.bps);
    wavhdr.bits = le2me_16(bits);
    wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8));

    wavhdr.data = le2me_32(WAV_ID_DATA);
    wavhdr.data_length=le2me_32(0x7ffff000);
    wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 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 */
            fwrite(&wavhdr,sizeof(wavhdr),1,fp);
        }
        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 > 0x7ffff000)
            mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
        else {
            wavhdr.file_length = data_length + sizeof(wavhdr) - 8;
            wavhdr.file_length = le2me_32(wavhdr.file_length);
            wavhdr.data_length = le2me_32(data_length);
            fwrite(&wavhdr,sizeof(wavhdr),1,fp);
        }
    }
    fclose(fp);
    if (ao_outputfilename)
        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){

// let libaf to do the conversion...
#if 0
//#ifdef WORDS_BIGENDIAN
    if (ao_data.format == AFMT_S16_LE) {
      unsigned short *buffer = (unsigned short *) data;
      register int i;
      for(i = 0; i < len/2; ++i) {
        buffer[i] = le2me_16(buffer[i]);
      }
    }
#endif

    if (ao_data.channels == 6 || ao_data.channels == 5) {
        int frame_size = le2me_16(wavhdr.bits) / 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;
}