Originally committed as revision 1163 to svn://svn.ffmpeg.org/ffmpeg/trunkpull/126/head
François Revol
23 years ago
committed by
Michael Niedermayer
parent
95e2ce4aa0
commit
dfdfa47cd1
6 changed files with 346 additions and 2 deletions
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/*
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* BeOS audio play interface |
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* Copyright (c) 2000, 2001 Fabrice Bellard. |
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* |
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* This library is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2 of the License, or (at your option) any later version. |
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* |
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* This library is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with this library; if not, write to the Free Software |
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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*/ |
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#include <signal.h> |
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#include <stdlib.h> |
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#include <stdio.h> |
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#include <string.h> |
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#include <unistd.h> |
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#include <sys/time.h> |
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#include <Application.h> |
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#include <SoundPlayer.h> |
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extern "C" { |
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#include "avformat.h" |
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} |
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//const char *audio_device = "/dev/dsp";
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const char *audio_device = "beosaudio:"; |
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#define AUDIO_BLOCK_SIZE 4096 |
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typedef struct { |
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int fd; |
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int sample_rate; |
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int channels; |
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int frame_size; /* in bytes ! */ |
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CodecID codec_id; |
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int flip_left : 1; |
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UINT8 buffer[AUDIO_BLOCK_SIZE]; |
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int buffer_ptr; |
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int pipefd; /* the other end of the pipe */ |
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BSoundPlayer *player; |
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int has_quit; /* signal callbacks not to wait */ |
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} AudioData; |
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static int own_BApp_created = 0; |
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/* create the BApplication and Run() it */ |
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static int32 bapp_thread(void *arg) |
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{ |
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new BApplication("application/x-vnd.ffmpeg"); |
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own_BApp_created = 1; |
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be_app->Run(); |
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/* kill the process group */ |
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kill(0, SIGINT); |
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return B_OK; |
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} |
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/* called back by BSoundPlayer */ |
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static void audioplay_callback(void *cookie, void *buffer, size_t bufferSize, const media_raw_audio_format &format) |
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{ |
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AudioData *s; |
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size_t amount; |
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unsigned char *buf = (unsigned char *)buffer; |
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s = (AudioData *)cookie; |
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if (s->has_quit) |
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return; |
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while (bufferSize > 0) { |
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if ((amount = read(s->pipefd, buf, bufferSize)) < B_OK) { |
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puts("EPIPE"); |
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snooze(100000); |
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s->player->SetHasData(false); |
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return; |
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} |
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if (amount == 0) { |
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snooze(100000); |
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s->player->SetHasData(false); |
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return; |
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} |
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buf += amount; |
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bufferSize -= amount; |
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} |
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} |
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static int audio_open(AudioData *s, int is_output) |
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{ |
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int p[2]; |
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int ret; |
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media_raw_audio_format format; |
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if (!is_output) |
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return -EIO; /* not for now */ |
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ret = pipe(p); |
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if (ret < 0) |
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return -EIO; |
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s->fd = p[is_output?1:0]; |
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s->pipefd = p[is_output?0:1]; |
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if (s->fd < 0) { |
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perror(is_output?"audio out":"audio in"); |
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return -EIO; |
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} |
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/* non blocking mode */ |
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// fcntl(s->fd, F_SETFL, O_NONBLOCK);
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// fcntl(s->pipefd, F_SETFL, O_NONBLOCK);
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s->frame_size = AUDIO_BLOCK_SIZE; |
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format = media_raw_audio_format::wildcard; |
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format.format = media_raw_audio_format::B_AUDIO_SHORT; |
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format.byte_order = B_HOST_IS_LENDIAN ? B_MEDIA_LITTLE_ENDIAN : B_MEDIA_BIG_ENDIAN; |
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format.channel_count = s->channels; |
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format.buffer_size = s->frame_size; |
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format.frame_rate = s->sample_rate; |
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s->player = new BSoundPlayer(&format, "ffmpeg output", audioplay_callback); |
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if (s->player->InitCheck() != B_OK) { |
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delete s->player; |
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s->player = NULL; |
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close(s->fd); |
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close(s->pipefd); |
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return -EIO; |
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} |
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s->player->SetCookie(s); |
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s->player->SetVolume(1.0); |
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s->player->Start(); |
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s->player->SetHasData(true); |
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return 0; |
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} |
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static int audio_close(AudioData *s) |
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{ |
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s->has_quit = 1; |
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if (s->player) { |
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s->player->Stop(); |
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} |
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if (s->player) |
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delete s->player; |
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close(s->pipefd); |
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close(s->fd); |
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return 0; |
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} |
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/* sound output support */ |
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static int audio_write_header(AVFormatContext *s1) |
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{ |
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AudioData *s = (AudioData *)s1->priv_data; |
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AVStream *st; |
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int ret; |
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st = s1->streams[0]; |
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s->sample_rate = st->codec.sample_rate; |
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s->channels = st->codec.channels; |
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ret = audio_open(s, 1); |
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if (ret < 0) |
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return -EIO; |
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return 0; |
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} |
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static int audio_write_packet(AVFormatContext *s1, int stream_index, |
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UINT8 *buf, int size, int force_pts) |
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{ |
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AudioData *s = (AudioData *)s1->priv_data; |
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int len, ret; |
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while (size > 0) { |
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len = AUDIO_BLOCK_SIZE - s->buffer_ptr; |
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if (len > size) |
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len = size; |
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memcpy(s->buffer + s->buffer_ptr, buf, len); |
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s->buffer_ptr += len; |
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if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) { |
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for(;;) { |
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snooze(1000); |
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ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE); |
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if (ret != 0) |
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break; |
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if (ret < 0 && (errno != EAGAIN && errno != EINTR)) |
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return -EIO; |
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} |
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s->buffer_ptr = 0; |
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} |
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buf += len; |
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size -= len; |
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} |
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return 0; |
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} |
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static int audio_write_trailer(AVFormatContext *s1) |
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{ |
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AudioData *s = (AudioData *)s1->priv_data; |
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audio_close(s); |
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return 0; |
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} |
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/* grab support */ |
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static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) |
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{ |
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AudioData *s = (AudioData *)s1->priv_data; |
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AVStream *st; |
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int ret; |
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if (!ap || ap->sample_rate <= 0 || ap->channels <= 0) |
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return -1; |
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st = av_new_stream(s1, 0); |
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if (!st) { |
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return -ENOMEM; |
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} |
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s->sample_rate = ap->sample_rate; |
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s->channels = ap->channels; |
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ret = audio_open(s, 0); |
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if (ret < 0) { |
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av_free(st); |
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return -EIO; |
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} else { |
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/* take real parameters */ |
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st->codec.codec_type = CODEC_TYPE_AUDIO; |
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st->codec.codec_id = s->codec_id; |
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st->codec.sample_rate = s->sample_rate; |
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st->codec.channels = s->channels; |
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return 0; |
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} |
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} |
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) |
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{ |
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AudioData *s = (AudioData *)s1->priv_data; |
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int ret; |
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if (av_new_packet(pkt, s->frame_size) < 0) |
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return -EIO; |
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for(;;) { |
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ret = read(s->fd, pkt->data, pkt->size); |
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if (ret > 0) |
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break; |
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if (ret == -1 && (errno == EAGAIN || errno == EINTR)) { |
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av_free_packet(pkt); |
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pkt->size = 0; |
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return 0; |
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} |
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if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) { |
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av_free_packet(pkt); |
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return -EIO; |
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} |
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} |
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pkt->size = ret; |
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if (s->flip_left && s->channels == 2) { |
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int i; |
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short *p = (short *) pkt->data; |
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for (i = 0; i < ret; i += 4) { |
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*p = ~*p; |
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p += 2; |
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} |
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} |
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return 0; |
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} |
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static int audio_read_close(AVFormatContext *s1) |
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{ |
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AudioData *s = (AudioData *)s1->priv_data; |
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audio_close(s); |
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return 0; |
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} |
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AVInputFormat audio_in_format = { |
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"audio_device", |
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"audio grab and output", |
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sizeof(AudioData), |
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NULL, |
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audio_read_header, |
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audio_read_packet, |
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audio_read_close, |
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NULL, |
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AVFMT_NOFILE, |
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}; |
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AVOutputFormat audio_out_format = { |
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"audio_device", |
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"audio grab and output", |
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"", |
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"", |
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sizeof(AudioData), |
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#ifdef WORDS_BIGENDIAN |
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CODEC_ID_PCM_S16BE, |
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#else |
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CODEC_ID_PCM_S16LE, |
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#endif |
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CODEC_ID_NONE, |
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audio_write_header, |
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audio_write_packet, |
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audio_write_trailer, |
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AVFMT_NOFILE, |
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}; |
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extern "C" { |
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int audio_init(void) |
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{ |
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/* needed by libmedia */ |
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if (be_app == NULL) { |
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resume_thread(spawn_thread(bapp_thread, "ffmpeg BApplication", B_NORMAL_PRIORITY, NULL)); |
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while (!own_BApp_created) |
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snooze(50000); |
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} |
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av_register_input_format(&audio_in_format); |
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av_register_output_format(&audio_out_format); |
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return 0; |
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} |
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} // "C"
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