mirror of https://github.com/FFmpeg/FFmpeg.git
Diego Biurrun
7 years ago
parent
6ce13070bd
commit
d46cd24986
4 changed files with 182 additions and 282 deletions
@ -1,94 +0,0 @@ |
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/*
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* ALSA input and output |
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) |
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) |
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* |
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* This file is part of Libav. |
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* |
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* Libav 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 2.1 of the License, or (at your option) any later version. |
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* |
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* Libav 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 Libav; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/**
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* @file |
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* ALSA input and output: definitions and structures |
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* @author Luca Abeni ( lucabe72 email it ) |
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* @author Benoit Fouet ( benoit fouet free fr ) |
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*/ |
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#ifndef AVDEVICE_ALSA_H |
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#define AVDEVICE_ALSA_H |
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#include <alsa/asoundlib.h> |
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#include "config.h" |
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#include "libavformat/avformat.h" |
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#include "libavutil/log.h" |
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/* XXX: we make the assumption that the soundcard accepts this format */ |
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/* XXX: find better solution with "preinit" method, needed also in
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other formats */ |
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#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE) |
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#define ALSA_BUFFER_SIZE_MAX 32768 |
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typedef struct AlsaData { |
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AVClass *class; |
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snd_pcm_t *h; |
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int frame_size; ///< preferred size for reads and writes
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int period_size; ///< bytes per sample * channels
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int sample_rate; ///< sample rate set by user
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int channels; ///< number of channels set by user
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void (*reorder_func)(const void *, void *, int); |
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void *reorder_buf; |
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int reorder_buf_size; ///< in frames
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} AlsaData; |
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/**
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* Open an ALSA PCM. |
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* |
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* @param s media file handle |
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* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK |
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* @param sample_rate in: requested sample rate; |
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* out: actually selected sample rate |
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* @param channels number of channels |
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* @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE; |
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* out: actually selected AVCodecID, changed only if |
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* AV_CODEC_ID_NONE was requested |
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* |
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* @return 0 if OK, AVERROR_xxx on error |
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*/ |
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int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode, |
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unsigned int *sample_rate, |
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int channels, enum AVCodecID *codec_id); |
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/**
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* Close the ALSA PCM. |
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* |
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* @param s1 media file handle |
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* |
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* @return 0 |
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*/ |
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int ff_alsa_close(AVFormatContext *s1); |
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/**
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* Try to recover from ALSA buffer underrun. |
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* |
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* @param s1 media file handle |
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* @param err error code reported by the previous ALSA call |
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* |
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* @return 0 if OK, AVERROR_xxx on error |
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*/ |
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int ff_alsa_xrun_recover(AVFormatContext *s1, int err); |
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#endif /* AVDEVICE_ALSA_H */ |
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@ -1,178 +0,0 @@ |
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/*
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* ALSA input and output |
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) |
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) |
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* |
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* This file is part of Libav. |
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* |
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* Libav 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 2.1 of the License, or (at your option) any later version. |
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* |
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* Libav 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 Libav; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/**
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* @file |
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* ALSA input and output: input |
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* @author Luca Abeni ( lucabe72 email it ) |
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* @author Benoit Fouet ( benoit fouet free fr ) |
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* @author Nicolas George ( nicolas george normalesup org ) |
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* |
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* This avdevice decoder allows to capture audio from an ALSA (Advanced |
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* Linux Sound Architecture) device. |
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* |
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* The filename parameter is the name of an ALSA PCM device capable of |
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* capture, for example "default" or "plughw:1"; see the ALSA documentation |
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* for naming conventions. The empty string is equivalent to "default". |
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* |
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* The capture period is set to the lower value available for the device, |
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* which gives a low latency suitable for real-time capture. |
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* |
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* The PTS are an Unix time in microsecond. |
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* |
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* Due to a bug in the ALSA library |
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* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
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* decoder does not work with certain ALSA plugins, especially the dsnoop |
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* plugin. |
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*/ |
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#include <alsa/asoundlib.h> |
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#include "libavutil/internal.h" |
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#include "libavutil/opt.h" |
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#include "libavformat/avformat.h" |
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#include "libavformat/internal.h" |
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#include "alsa.h" |
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static av_cold int audio_read_header(AVFormatContext *s1) |
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{ |
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AlsaData *s = s1->priv_data; |
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AVStream *st; |
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int ret; |
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enum AVCodecID codec_id; |
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snd_pcm_sw_params_t *sw_params; |
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st = avformat_new_stream(s1, NULL); |
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if (!st) { |
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av_log(s1, AV_LOG_ERROR, "Cannot add stream\n"); |
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return AVERROR(ENOMEM); |
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} |
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codec_id = s1->audio_codec_id; |
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ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels, |
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&codec_id); |
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if (ret < 0) { |
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return AVERROR(EIO); |
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} |
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if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW) |
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av_log(s1, AV_LOG_WARNING, |
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"capture with some ALSA plugins, especially dsnoop, " |
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"may hang.\n"); |
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ret = snd_pcm_sw_params_malloc(&sw_params); |
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if (ret < 0) { |
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av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n", |
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snd_strerror(ret)); |
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goto fail; |
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} |
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snd_pcm_sw_params_current(s->h, sw_params); |
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snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE); |
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ret = snd_pcm_sw_params(s->h, sw_params); |
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snd_pcm_sw_params_free(sw_params); |
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if (ret < 0) { |
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av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n", |
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snd_strerror(ret)); |
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goto fail; |
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} |
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/* take real parameters */ |
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; |
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st->codecpar->codec_id = codec_id; |
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st->codecpar->sample_rate = s->sample_rate; |
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st->codecpar->channels = s->channels; |
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ |
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return 0; |
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fail: |
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snd_pcm_close(s->h); |
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return AVERROR(EIO); |
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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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AlsaData *s = s1->priv_data; |
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AVStream *st = s1->streams[0]; |
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int res; |
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snd_htimestamp_t timestamp; |
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snd_pcm_uframes_t ts_delay; |
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if (av_new_packet(pkt, s->period_size) < 0) { |
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return AVERROR(EIO); |
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} |
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while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) { |
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if (res == -EAGAIN) { |
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av_packet_unref(pkt); |
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return AVERROR(EAGAIN); |
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} |
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if (ff_alsa_xrun_recover(s1, res) < 0) { |
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av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", |
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snd_strerror(res)); |
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av_packet_unref(pkt); |
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return AVERROR(EIO); |
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} |
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} |
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snd_pcm_htimestamp(s->h, &ts_delay, ×tamp); |
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ts_delay += res; |
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pkt->pts = timestamp.tv_sec * 1000000LL |
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+ (timestamp.tv_nsec * st->codecpar->sample_rate |
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- (int64_t)ts_delay * 1000000000LL + st->codecpar->sample_rate * 500LL) |
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/ (st->codecpar->sample_rate * 1000LL); |
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pkt->size = res * s->frame_size; |
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return 0; |
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} |
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static const AVOption options[] = { |
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{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, |
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{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, |
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{ NULL }, |
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}; |
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static const AVClass alsa_demuxer_class = { |
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.class_name = "ALSA demuxer", |
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.item_name = av_default_item_name, |
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.option = options, |
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.version = LIBAVUTIL_VERSION_INT, |
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}; |
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AVInputFormat ff_alsa_demuxer = { |
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.name = "alsa", |
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.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"), |
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.priv_data_size = sizeof(AlsaData), |
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.read_header = audio_read_header, |
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.read_packet = audio_read_packet, |
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.read_close = ff_alsa_close, |
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.flags = AVFMT_NOFILE, |
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.priv_class = &alsa_demuxer_class, |
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}; |
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