| 1 | /* |
| 2 | * ALSA input and output |
| 3 | * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) |
| 4 | * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) |
| 5 | * |
| 6 | * This file is part of FFmpeg. |
| 7 | * |
| 8 | * FFmpeg is free software; you can redistribute it and/or |
| 9 | * modify it under the terms of the GNU Lesser General Public |
| 10 | * License as published by the Free Software Foundation; either |
| 11 | * version 2.1 of the License, or (at your option) any later version. |
| 12 | * |
| 13 | * FFmpeg is distributed in the hope that it will be useful, |
| 14 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 15 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 16 | * Lesser General Public License for more details. |
| 17 | * |
| 18 | * You should have received a copy of the GNU Lesser General Public |
| 19 | * License along with FFmpeg; if not, write to the Free Software |
| 20 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| 21 | */ |
| 22 | |
| 23 | /** |
| 24 | * @file |
| 25 | * ALSA input and output: definitions and structures |
| 26 | * @author Luca Abeni ( lucabe72 email it ) |
| 27 | * @author Benoit Fouet ( benoit fouet free fr ) |
| 28 | */ |
| 29 | |
| 30 | #ifndef AVDEVICE_ALSA_AUDIO_H |
| 31 | #define AVDEVICE_ALSA_AUDIO_H |
| 32 | |
| 33 | #include <alsa/asoundlib.h> |
| 34 | #include "config.h" |
| 35 | #include "libavutil/log.h" |
| 36 | #include "timefilter.h" |
| 37 | #include "avdevice.h" |
| 38 | |
| 39 | /* XXX: we make the assumption that the soundcard accepts this format */ |
| 40 | /* XXX: find better solution with "preinit" method, needed also in |
| 41 | other formats */ |
| 42 | #define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE) |
| 43 | |
| 44 | typedef void (*ff_reorder_func)(const void *, void *, int); |
| 45 | |
| 46 | #define ALSA_BUFFER_SIZE_MAX 65536 |
| 47 | |
| 48 | typedef struct AlsaData { |
| 49 | AVClass *class; |
| 50 | snd_pcm_t *h; |
| 51 | int frame_size; ///< bytes per sample * channels |
| 52 | int period_size; ///< preferred size for reads and writes, in frames |
| 53 | int sample_rate; ///< sample rate set by user |
| 54 | int channels; ///< number of channels set by user |
| 55 | int last_period; |
| 56 | TimeFilter *timefilter; |
| 57 | void (*reorder_func)(const void *, void *, int); |
| 58 | void *reorder_buf; |
| 59 | int reorder_buf_size; ///< in frames |
| 60 | int64_t timestamp; ///< current timestamp, without latency applied. |
| 61 | } AlsaData; |
| 62 | |
| 63 | /** |
| 64 | * Open an ALSA PCM. |
| 65 | * |
| 66 | * @param s media file handle |
| 67 | * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK |
| 68 | * @param sample_rate in: requested sample rate; |
| 69 | * out: actually selected sample rate |
| 70 | * @param channels number of channels |
| 71 | * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE; |
| 72 | * out: actually selected AVCodecID, changed only if |
| 73 | * AV_CODEC_ID_NONE was requested |
| 74 | * |
| 75 | * @return 0 if OK, AVERROR_xxx on error |
| 76 | */ |
| 77 | int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode, |
| 78 | unsigned int *sample_rate, |
| 79 | int channels, enum AVCodecID *codec_id); |
| 80 | |
| 81 | /** |
| 82 | * Close the ALSA PCM. |
| 83 | * |
| 84 | * @param s1 media file handle |
| 85 | * |
| 86 | * @return 0 |
| 87 | */ |
| 88 | int ff_alsa_close(AVFormatContext *s1); |
| 89 | |
| 90 | /** |
| 91 | * Try to recover from ALSA buffer underrun. |
| 92 | * |
| 93 | * @param s1 media file handle |
| 94 | * @param err error code reported by the previous ALSA call |
| 95 | * |
| 96 | * @return 0 if OK, AVERROR_xxx on error |
| 97 | */ |
| 98 | int ff_alsa_xrun_recover(AVFormatContext *s1, int err); |
| 99 | |
| 100 | int ff_alsa_extend_reorder_buf(AlsaData *s, int size); |
| 101 | |
| 102 | int ff_alsa_get_device_list(AVDeviceInfoList *device_list, snd_pcm_stream_t stream_type); |
| 103 | |
| 104 | #endif /* AVDEVICE_ALSA_AUDIO_H */ |