Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 27969:7ddd69cf214f
Lock/unlock surface only once even when drawing many slices.
Patch originally by Jim Hauxwell [james dattrax co.uk]
author | reimar |
---|---|
date | Sun, 23 Nov 2008 18:42:29 +0000 |
parents | c75d2f3cf4eb |
children | e45b08f2f5d3 |
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#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 #define WAV_ID_FLOAT_PCM 0x0003 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=8; switch(format){ case AF_FORMAT_S32_BE: format=AF_FORMAT_S32_LE; case AF_FORMAT_S32_LE: bits=32; break; case AF_FORMAT_FLOAT_BE: format=AF_FORMAT_FLOAT_LE; case AF_FORMAT_FLOAT_LE: bits=32; break; 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(format == AF_FORMAT_FLOAT_LE ? WAV_ID_FLOAT_PCM : 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; }