Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 6398:13c35228abe8
updated to english version
author | jonas |
---|---|
date | Wed, 12 Jun 2002 16:31:32 +0000 |
parents | 2eec40929570 |
children | 827f08ddf044 |
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#include "config.h" #include <stdio.h> #include <stdlib.h> #include <string.h> #include "bswap.h" #include "afmt.h" #include "audio_out.h" #include "audio_out_internal.h" static ao_info_t info = { "RAW PCM/WAVE file writer audio output", "pcm", "Atmosfear", "" }; LIBAO_EXTERN(pcm) extern int vo_pts; char *ao_outputfilename = NULL; 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 { unsigned long riff; unsigned long file_length; unsigned long wave; unsigned long fmt; unsigned long fmt_length; short fmt_tag; short channels; unsigned long sample_rate; unsigned long bytes_per_second; short block_align; short bits; unsigned long data; unsigned long data_length; }; /* init with default values */ static struct WaveHeader wavhdr = { le2me_32(WAV_ID_RIFF), le2me_32(0x00000000), 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), le2me_32(0x00000000) }; static FILE *fp = NULL; // to set/get/query special features/parameters static int control(int cmd,int 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; if(!ao_outputfilename) { ao_outputfilename = (char *) malloc(sizeof(char) * 14); strcpy(ao_outputfilename, (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 */ if (format == AFMT_S16_BE) { bits = 16; } else { bits = format; } wavhdr.channels = le2me_16(channels); wavhdr.sample_rate = le2me_32(rate); wavhdr.bytes_per_second = rate * (bits / 8) * channels; wavhdr.bytes_per_second = le2me_32(wavhdr.bytes_per_second); wavhdr.bits = le2me_16(bits); printf("PCM: File: %s (%s)\n" "PCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename, (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); printf("PCM: Info: fastest dumping is achieved with -vo null " "-hardframedrop.\n" "PCM: Info: to write WAVE files use -waveheader (default); " "for RAW PCM -nowaveheader.\n"); fp = fopen(ao_outputfilename, "wb"); ao_data.outburst = 65536; if(fp) { if(ao_pcm_waveheader) /* Reserve space for wave header */ fseek(fp, sizeof(wavhdr), SEEK_SET); return 1; } printf("PCM: Failed to open %s for writing!\n", ao_outputfilename); return 0; } // close audio device static void uninit(){ if(ao_pcm_waveheader){ /* Write wave header */ wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; wavhdr.file_length = le2me_32(wavhdr.file_length); fseek(fp, 0, SEEK_SET); fwrite(&wavhdr,sizeof(wavhdr),1,fp); } fclose(fp); free(ao_outputfilename); } // 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){ #ifdef WORDS_BIGENDIAN register int i; unsigned short *buffer = (unsigned short *) data; if (wavhdr.bits == le2me_16(16)) { for(i = 0; i < len/2; ++i) { buffer[i] = le2me_16(buffer[i]); } } /* FIXME: take care of cases with more than 8 bits here? */ #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; }