mirror of https://github.com/mpv-player/mpv
266 lines
6.9 KiB
C
266 lines
6.9 KiB
C
/*
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* PCM audio output driver
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include "libavutil/common.h"
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#include "mpbswap.h"
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#include "subopt-helper.h"
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#include "libaf/af_format.h"
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#include "libaf/reorder_ch.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "mp_msg.h"
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#include "help_mp.h"
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static const ao_info_t info =
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{
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"RAW PCM/WAVE file writer audio output",
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"pcm",
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"Atmosfear",
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""
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};
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LIBAO_EXTERN(pcm)
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extern int vo_pts;
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static char *ao_outputfilename = NULL;
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static int ao_pcm_waveheader = 1;
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static int fast = 0;
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#define WAV_ID_RIFF 0x46464952 /* "RIFF" */
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#define WAV_ID_WAVE 0x45564157 /* "WAVE" */
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#define WAV_ID_FMT 0x20746d66 /* "fmt " */
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#define WAV_ID_DATA 0x61746164 /* "data" */
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#define WAV_ID_PCM 0x0001
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#define WAV_ID_FLOAT_PCM 0x0003
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struct WaveHeader
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{
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uint32_t riff;
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uint32_t file_length;
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uint32_t wave;
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uint32_t fmt;
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uint32_t fmt_length;
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uint16_t fmt_tag;
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uint16_t channels;
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uint32_t sample_rate;
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uint32_t bytes_per_second;
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uint16_t block_align;
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uint16_t bits;
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uint32_t data;
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uint32_t data_length;
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};
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/* init with default values */
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static struct WaveHeader wavhdr;
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static uint64_t data_length;
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static FILE *fp = NULL;
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// to set/get/query special features/parameters
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static int control(int cmd,void *arg){
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return -1;
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}
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(int rate,int channels,int format,int flags){
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int bits;
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opt_t subopts[] = {
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{"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
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{"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
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{"fast", OPT_ARG_BOOL, &fast, NULL},
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{NULL}
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};
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// set defaults
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ao_pcm_waveheader = 1;
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if (subopt_parse(ao_subdevice, subopts) != 0) {
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return 0;
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}
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if (!ao_outputfilename){
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ao_outputfilename =
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strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
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}
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bits=8;
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switch(format){
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case AF_FORMAT_S32_BE:
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format=AF_FORMAT_S32_LE;
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case AF_FORMAT_S32_LE:
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bits=32;
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break;
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case AF_FORMAT_FLOAT_BE:
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format=AF_FORMAT_FLOAT_LE;
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case AF_FORMAT_FLOAT_LE:
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bits=32;
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break;
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case AF_FORMAT_S8:
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format=AF_FORMAT_U8;
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case AF_FORMAT_U8:
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break;
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case AF_FORMAT_AC3:
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bits=16;
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break;
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default:
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format=AF_FORMAT_S16_LE;
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bits=16;
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break;
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}
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ao_data.outburst = 65536;
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ao_data.buffersize= 2*65536;
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ao_data.channels=channels;
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ao_data.samplerate=rate;
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ao_data.format=format;
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ao_data.bps=channels*rate*(bits/8);
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wavhdr.riff = le2me_32(WAV_ID_RIFF);
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wavhdr.wave = le2me_32(WAV_ID_WAVE);
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wavhdr.fmt = le2me_32(WAV_ID_FMT);
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wavhdr.fmt_length = le2me_32(16);
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wavhdr.fmt_tag = le2me_16(format == AF_FORMAT_FLOAT_LE ? WAV_ID_FLOAT_PCM : WAV_ID_PCM);
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wavhdr.channels = le2me_16(ao_data.channels);
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wavhdr.sample_rate = le2me_32(ao_data.samplerate);
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wavhdr.bytes_per_second = le2me_32(ao_data.bps);
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wavhdr.bits = le2me_16(bits);
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wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8));
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wavhdr.data = le2me_32(WAV_ID_DATA);
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wavhdr.data_length=le2me_32(0x7ffff000);
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wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
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mp_tmsg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
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(ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
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(channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
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mp_tmsg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
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fp = fopen(ao_outputfilename, "wb");
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if(fp) {
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if(ao_pcm_waveheader){ /* Reserve space for wave header */
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fwrite(&wavhdr,sizeof(wavhdr),1,fp);
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}
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return 1;
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}
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mp_tmsg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
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ao_outputfilename);
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return 0;
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}
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// close audio device
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static void uninit(int immed){
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if(ao_pcm_waveheader){ /* Rewrite wave header */
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if (fseek(fp, 0, SEEK_SET) != 0)
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mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
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else if (data_length > 0x7ffff000)
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mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
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else {
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wavhdr.file_length = data_length + sizeof(wavhdr) - 8;
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wavhdr.file_length = le2me_32(wavhdr.file_length);
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wavhdr.data_length = le2me_32(data_length);
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fwrite(&wavhdr,sizeof(wavhdr),1,fp);
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}
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}
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fclose(fp);
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if (ao_outputfilename)
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free(ao_outputfilename);
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ao_outputfilename = NULL;
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset(void){
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause(void)
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{
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// for now, just call reset();
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reset();
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}
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// resume playing, after audio_pause()
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static void audio_resume(void)
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{
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}
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// return: how many bytes can be played without blocking
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static int get_space(void){
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if(vo_pts)
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return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
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return ao_data.outburst;
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}
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// plays 'len' bytes of 'data'
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// it should round it down to outburst*n
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// return: number of bytes played
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static int play(void* data,int len,int flags){
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// let libaf to do the conversion...
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#if 0
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//#ifdef WORDS_BIGENDIAN
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if (ao_data.format == AFMT_S16_LE) {
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unsigned short *buffer = (unsigned short *) data;
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register int i;
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for(i = 0; i < len/2; ++i) {
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buffer[i] = le2me_16(buffer[i]);
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}
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}
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#endif
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if (ao_data.channels == 6 || ao_data.channels == 5) {
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int frame_size = le2me_16(wavhdr.bits) / 8;
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len -= len % (frame_size * ao_data.channels);
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reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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ao_data.channels,
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len / frame_size, frame_size);
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}
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//printf("PCM: Writing chunk!\n");
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fwrite(data,len,1,fp);
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if(ao_pcm_waveheader)
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data_length += len;
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return len;
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}
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// return: delay in seconds between first and last sample in buffer
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static float get_delay(void){
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return 0.0;
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}
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