Mercurial > mplayer.hg
view stream/audio_in.c @ 33307:552f1f7731c8
Set GUI initialization flag earlier.
So far, the flag is set after the call of guiInit(), but there are calls
to GUI functions (like reading the GUI config) before that. As MPlayer
(when exiting) only calls GUI's cleanup function guiDone() if the flag
is set, it neglects calling guiDone() on any error between the first GUI
function call and execution of guiInit(). Now the flag is set before the
first GUI function call.
As a result, GUI's own exit function (also used for fatal GUI errors, and
thus also for errors occurring before guiInit() properly finishes) doesn't
have to handle cleanup any longer. The code can be moved to guiDone().
author | ib |
---|---|
date | Sat, 07 May 2011 15:30:14 +0000 |
parents | b39155e98ac3 |
children |
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/* * 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 <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "audio_in.h" #include "mp_msg.h" #include "help_mp.h" #include <string.h> #include <errno.h> // sanitizes ai structure before calling other functions int audio_in_init(audio_in_t *ai, int type) { ai->type = type; ai->setup = 0; ai->channels = -1; ai->samplerate = -1; ai->blocksize = -1; ai->bytes_per_sample = -1; ai->samplesize = -1; switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: ai->alsa.handle = NULL; ai->alsa.log = NULL; ai->alsa.device = strdup("default"); return 0; #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: ai->oss.audio_fd = -1; ai->oss.device = strdup("/dev/dsp"); return 0; #endif default: return -1; } } int audio_in_setup(audio_in_t *ai) { switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: if (ai_alsa_init(ai) < 0) return -1; ai->setup = 1; return 0; #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: if (ai_oss_init(ai) < 0) return -1; ai->setup = 1; return 0; #endif default: return -1; } } int audio_in_set_samplerate(audio_in_t *ai, int rate) { switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: ai->req_samplerate = rate; if (!ai->setup) return 0; if (ai_alsa_setup(ai) < 0) return -1; return ai->samplerate; #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: ai->req_samplerate = rate; if (!ai->setup) return 0; if (ai_oss_set_samplerate(ai) < 0) return -1; return ai->samplerate; #endif default: return -1; } } int audio_in_set_channels(audio_in_t *ai, int channels) { switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: ai->req_channels = channels; if (!ai->setup) return 0; if (ai_alsa_setup(ai) < 0) return -1; return ai->channels; #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: ai->req_channels = channels; if (!ai->setup) return 0; if (ai_oss_set_channels(ai) < 0) return -1; return ai->channels; #endif default: return -1; } } int audio_in_set_device(audio_in_t *ai, char *device) { #ifdef CONFIG_ALSA int i; #endif if (ai->setup) return -1; switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: free(ai->alsa.device); ai->alsa.device = strdup(device); /* mplayer cannot handle colons in arguments */ for (i = 0; i < (int)strlen(ai->alsa.device); i++) { if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; } return 0; #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: free(ai->oss.device); ai->oss.device = strdup(device); return 0; #endif default: return -1; } } int audio_in_uninit(audio_in_t *ai) { if (ai->setup) { switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: if (ai->alsa.log) snd_output_close(ai->alsa.log); if (ai->alsa.handle) { snd_pcm_close(ai->alsa.handle); } ai->setup = 0; return 0; #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: close(ai->oss.audio_fd); ai->setup = 0; return 0; #endif } } return -1; } int audio_in_start_capture(audio_in_t *ai) { switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: return snd_pcm_start(ai->alsa.handle); #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: return 0; #endif default: return -1; } } int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) { int ret; switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); if (ret != ai->alsa.chunk_size) { if (ret < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret)); if (ret == -EPIPE) { if (ai_alsa_xrun(ai) == 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut); } else { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover); } } } else { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); } return -1; } return ret; #endif #ifdef CONFIG_OSS_AUDIO case AUDIO_IN_OSS: ret = read(ai->oss.audio_fd, buffer, ai->blocksize); if (ret != ai->blocksize) { if (ret < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno)); } else { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); } return -1; } return ret; #endif default: return -1; } }