Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 3639:64ee21b3bd09
Modified the sync code once again, commented out hardware pts sync (I'll likely burn in hell before understanding how to get this bastard to sync well)
Added automagic setup of aspect ratio, will remove the "aspect-bug" (I hope). As well as please you rich 16:9 doods ;)
author | mswitch |
---|---|
date | Thu, 20 Dec 2001 20:50:35 +0000 |
parents | 981a9e5118ce |
children | de4074ab4e5f |
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#include <stdio.h> #include <stdlib.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) 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; }; static struct WaveHeader wavhdr = { WAV_ID_RIFF, 0x00000000, WAV_ID_WAVE, WAV_ID_FMT, 16, WAV_ID_PCM, 2, 44100, 192000, 4, 16, WAV_ID_DATA, 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){ if(!ao_outputfilename) { ao_outputfilename = (char *) malloc(sizeof(char) * 14); strcpy(ao_outputfilename,(ao_pcm_waveheader ? "audiodump.wav" : "audiodump.pcm")); } wavhdr.channels = channels; wavhdr.sample_rate = rate; wavhdr.bytes_per_second = rate * (format / 8) * channels; wavhdr.bits = format; printf("PCM: File: %s (%s) 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"); printf("PCM: Info - to write WAVE files use -waveheader (default), for RAW PCM -nowaveheader.\n"); fp = fopen(ao_outputfilename, "wb"); ao_data.outburst = 4096; 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); 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(){ 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){ //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; }