Mercurial > mplayer.hg
view stream/audio_in.c @ 31139:9a2e299dc256
Add final missing bits of CineForm HD support on Linux (via the Windows
DirectShow codec).
Required changes:
- codecs.conf entry (of course).
- Allow opening files with ¡È.col¡É in the file name, just like ¡Èvp3¡É and ¡È.fpf¡É
already was allowed. (CineForm expects to be able to do this, presumably
for some color management code.)
- In registry.c, fake a few registry keys that the codec expects the installer
to have written. Also, change a few magic numbers (0, 2) to the appropriate
constants (ERROR_SUCCESS, ERROR_FILE_NOT_FOUND) where appropriate, so the code
is easier to follow.
SMP works fine, but seemingly performs suboptimally (e.g., on my dual-core
laptop, CineForm performs better if I lie to it and tell it I have four cores).
I don't know if this is inherent in the codec, or some inefficiency in the
emulated synchronization primitives.
author | sesse |
---|---|
date | Sun, 23 May 2010 16:01:12 +0000 |
parents | ce0122361a39 |
children | b39155e98ac3 |
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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: if (ai->alsa.device) 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: if (ai->oss.device) 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; } }