Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 32958:80087dd00035
Fix some debug messages.
Avoid debug code if debugging isn't asked for and don't use debug messages
in such a way with control structures that there may be compiler warnings.
author | ib |
---|---|
date | Tue, 08 Mar 2011 09:57:04 +0000 |
parents | 8fa2f43cb760 |
children | 8cfe525f0ec0 |
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/* * PCM audio output driver * * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #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 "libvo/video_out.h" /* only for vo_pts */ #include "audio_out.h" #include "audio_out_internal.h" #include "mp_msg.h" #include "help_mp.h" #ifdef __MINGW32__ // for GetFileType to detect pipes #include <windows.h> #endif static const ao_info_t info = { "RAW PCM/WAVE file writer audio output", "pcm", "Atmosfear", "" }; LIBAO_EXTERN(pcm) 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 #define WAV_ID_FORMAT_EXTENSIBLE 0xfffe /* init with default values */ static uint64_t data_length; static FILE *fp = NULL; static void fput16le(uint16_t val, FILE *fp) { uint8_t bytes[2] = {val, val >> 8}; fwrite(bytes, 1, 2, fp); } static void fput32le(uint32_t val, FILE *fp) { uint8_t bytes[4] = {val, val >> 8, val >> 16, val >> 24}; fwrite(bytes, 1, 4, fp); } static void write_wave_header(FILE *fp, uint64_t data_length) { int use_waveex = (ao_data.channels >= 5 && ao_data.channels <= 8); uint16_t fmt = (ao_data.format == AF_FORMAT_FLOAT_LE) ? WAV_ID_FLOAT_PCM : WAV_ID_PCM; uint32_t fmt_chunk_size = use_waveex ? 40 : 16; int bits = af_fmt2bits(ao_data.format); // Master RIFF chunk fput32le(WAV_ID_RIFF, fp); // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size + data chunk hdr (8) + data length fput32le(12 + fmt_chunk_size + 8 + data_length, fp); fput32le(WAV_ID_WAVE, fp); // Format chunk fput32le(WAV_ID_FMT, fp); fput32le(fmt_chunk_size, fp); fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp); fput16le(ao_data.channels, fp); fput32le(ao_data.samplerate, fp); fput32le(ao_data.bps, fp); fput16le(ao_data.channels * (bits / 8), fp); fput16le(bits, fp); if (use_waveex) { // Extension chunk fput16le(22, fp); fput16le(bits, fp); switch (ao_data.channels) { case 5: fput32le(0x0607, fp); // L R C Lb Rb break; case 6: fput32le(0x060f, fp); // L R C Lb Rb LFE break; case 7: fput32le(0x0727, fp); // L R C Cb Ls Rs LFE break; case 8: fput32le(0x063f, fp); // L R C Lb Rb Ls Rs LFE break; } // 2 bytes format + 14 bytes guid fput32le(fmt, fp); fput32le(0x00100000, fp); fput32le(0xAA000080, fp); fput32le(0x719B3800, fp); } // Data chunk fput32le(WAV_ID_DATA, fp); fput32le(data_length, fp); } // 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){ const 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"); } if (ao_pcm_waveheader) { // WAV files must have one of the following formats switch(format){ case AF_FORMAT_U8: case AF_FORMAT_S16_LE: case AF_FORMAT_S24_LE: case AF_FORMAT_S32_LE: case AF_FORMAT_FLOAT_LE: case AF_FORMAT_AC3_BE: case AF_FORMAT_AC3_LE: break; default: format = AF_FORMAT_S16_LE; 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*(af_fmt2bits(format)/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 */ write_wave_header(fp, 0x7ffff000); } 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){ /* Rewrite wave header */ int broken_seek = 0; #ifdef __MINGW32__ // Windows, in its usual idiocy "emulates" seeks on pipes so it always looks // like they work. So we have to detect them brute-force. broken_seek = GetFileType((HANDLE)_get_osfhandle(_fileno(fp))) != FILE_TYPE_DISK; #endif if (broken_seek || fseek(fp, 0, SEEK_SET) != 0) mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n"); else { if (data_length > 0xfffff000) { mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n"); data_length = 0xfffff000; } write_wave_header(fp, data_length); } } fclose(fp); 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){ if (ao_data.channels == 5 || ao_data.channels == 6 || ao_data.channels == 8) { int frame_size = af_fmt2bits(ao_data.format) / 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) data_length += len; return len; } // return: delay in seconds between first and last sample in buffer static float get_delay(void){ return 0.0; }