Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 14333:3ac847d2018d
changed wording enkodier* to encode* to be consistant with manpage
author | kraymer |
---|---|
date | Mon, 03 Jan 2005 17:53:23 +0000 |
parents | 54e42c7eb713 |
children | de08cc60fd7e |
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#include "config.h" #include <stdio.h> #include <stdlib.h> #include <string.h> #include "bswap.h" #include "subopt-helper.h" #include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" #include "mp_msg.h" #include "help_mp.h" static 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; #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 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 = { le2me_32(WAV_ID_RIFF), /* same conventions than in sox/wav.c/wavwritehdr() */ 0, //le2me_32(0x7ffff024), le2me_32(WAV_ID_WAVE), le2me_32(WAV_ID_FMT), le2me_32(16), le2me_16(WAV_ID_PCM), le2me_16(2), le2me_32(44100), le2me_32(192000), le2me_16(4), le2me_16(16), le2me_32(WAV_ID_DATA), 0, //le2me_32(0x7ffff000) }; 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; strarg_t file; opt_t subopts[] = { {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL}, {"file", OPT_ARG_STR, &file, NULL}, {NULL} }; // set defaults ao_pcm_waveheader = 1; file.str = NULL; file.len = 0; if (subopt_parse(ao_subdevice, subopts) != 0) { return 0; } if (file.len > 0) { ao_outputfilename = malloc(file.len + 1); memcpy(ao_outputfilename, file.str, file.len); ao_outputfilename[file.len] = 0; } else ao_outputfilename = strdup((ao_pcm_waveheader)?"audiodump.wav":"audiodump.pcm"); /* bits is only equal to format if (format == 8) or (format == 16); this means that the following "if" is a kludge and should really be a switch to be correct in all cases */ bits=8; switch(format){ case AF_FORMAT_S8: format=AF_FORMAT_U8; case AF_FORMAT_U8: 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.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.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); wavhdr.file_length=wavhdr.data_length=0; } 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 && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */ wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; wavhdr.file_length = le2me_32(wavhdr.file_length); wavhdr.data_length = le2me_32(wavhdr.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(){ } // stop playing, keep buffers (for pause) static void audio_pause() { // for now, just call reset(); reset(); } // resume playing, after audio_pause() static void audio_resume() { } // return: how many bytes can be played without blocking static int get_space(){ if(vo_pts) return ao_data.pts < vo_pts ? 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 //printf("PCM: Writing chunk!\n"); fwrite(data,len,1,fp); if(ao_pcm_waveheader) wavhdr.data_length += len; return len; } // return: delay in seconds between first and last sample in buffer static float get_delay(){ return 0.0; }