Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 7951:32ae0a9d06aa
Updated the DXR3 section to reflect command changes, prebuf replaced
noprebuf and sync is a brand new command.
Also indicates that prebuffering no longer is the default operation.
Note regarding divx playback how you should set the lavc fps to 29.97
for proper playback on the em8300.
I saw something on mplayer-dev-eng about everyone being allowed to
commit doc-updates now. I hope this is valid, otherwise let me know.
author | mswitch |
---|---|
date | Tue, 29 Oct 2002 01:41:16 +0000 |
parents | 65037d309de3 |
children | af1cf8a7e9d2 |
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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 */ bits=8; switch(format){ case AFMT_S8: format=AFMT_U8; case AFMT_U8: break; default: #ifdef WORDS_BIGENDIAN format=AFMT_S16_BE; #else format=AFMT_S16_LE; #endif 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); 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 -vc null -vo null\n" "PCM: Info: to write WAVE files use -waveheader (default); " "for RAW PCM -nowaveheader.\n"); fp = fopen(ao_outputfilename, "wb"); if(fp) { if(ao_pcm_waveheader) /* Reserve space for wave header */ fwrite(&wavhdr,sizeof(wavhdr),1,fp); 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 && 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); 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){ // 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; }