Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 26110:69790f7caffa
configure: Set CONFIG_ENCODERS=yes in config.mak unconditionally
config.h already had "#define CONFIG_ENCODERS 1" unconditionally, but
the config.mak value depended on whether MEncoder was enabled.
Encoders need to be enabled as some encoder code is used by MPlayer
too. The inconsistent values broke compilation with --disable-mencoder
after libavcodec Makefile made compilation of i386/dsputilenc_mmx.o
depend on the config.mak value.
author | uau |
---|---|
date | Mon, 03 Mar 2008 03:02:36 +0000 |
parents | dfa8a510c81c |
children | 9946e4a6e457 |
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line source
#include "config.h" #include <stdio.h> #include <stdlib.h> #include <string.h> #include "libavutil/common.h" #include "mpbswap.h" #include "subopt-helper.h" #include "libaf/af_format.h" #include "libaf/reorder_ch.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; static int fast = 0; #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; 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; opt_t subopts[] = { {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL}, {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL}, {"fast", OPT_ARG_BOOL, &fast, NULL}, {NULL} }; // set defaults ao_pcm_waveheader = 1; if (subopt_parse(ao_subdevice, subopts) != 0) { return 0; } if (!ao_outputfilename){ 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; case AF_FORMAT_AC3: bits=16; 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.riff = le2me_32(WAV_ID_RIFF); wavhdr.wave = le2me_32(WAV_ID_WAVE); wavhdr.fmt = le2me_32(WAV_ID_FMT); wavhdr.fmt_length = le2me_32(16); wavhdr.fmt_tag = le2me_16(WAV_ID_PCM); 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.block_align = le2me_16(ao_data.channels * (bits / 8)); wavhdr.data = le2me_32(WAV_ID_DATA); 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(void){ } // stop playing, keep buffers (for pause) static void audio_pause(void) { // for now, just call reset(); reset(); } // resume playing, after audio_pause() static void audio_resume(void) { } // return: how many bytes can be played without blocking static int get_space(void){ if(vo_pts) return ao_data.pts < vo_pts + fast * 30000 ? 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 if (ao_data.channels == 6 || ao_data.channels == 5) { int frame_size = le2me_16(wavhdr.bits) / 8; len -= len % (frame_size * ao_data.channels); reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT, ao_data.channels, len / frame_size, frame_size); } //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(void){ return 0.0; }