Add ALSA support in libavdevice.

Patch by Nicolas George: name surname normalesup org
Original thread: [FFmpeg-devel] [PATCH] ALSA for libavdevice
Date: 12/09/2008 07:17 PM

Originally committed as revision 16800 to svn://svn.ffmpeg.org/ffmpeg/trunk

7 files changed, 564 insertions, 0 deletions

diff --git a/configure b/configure
index 15f6370e20..43881cd12f 100755
--- a/configure
+++ b/configure
@@ -838,6 +838,7 @@ ARCH_EXT_LIST='
 HAVE_LIST="
     $ARCH_EXT_LIST
     $THREADS_LIST
+    alsa_asoundlib_h
     altivec_h
     arpa_inet_h
     bswap
@@ -1069,6 +1070,10 @@ vdpau_deps="vdpau_vdpau_h vdpau_vdpau_x11_h"
 
 # demuxers / muxers
 ac3_demuxer_deps="ac3_parser"
+alsa_demuxer_deps="alsa_asoundlib_h snd_pcm_htimestamp"
+alsa_demuxer_extralibs="-lasound"
+alsa_muxer_deps="alsa_asoundlib_h"
+alsa_muxer_extralibs="-lasound"
 audio_beos_demuxer_deps="audio_beos"
 audio_beos_demuxer_extralibs="-lmedia -lbe"
 audio_beos_muxer_deps="audio_beos"
@@ -2044,6 +2049,9 @@ check_header dev/ic/bt8xx.h
 check_header sys/soundcard.h
 check_header soundcard.h
 
+check_header alsa/asoundlib.h &&
+check_lib2 alsa/asoundlib.h snd_pcm_htimestamp -lasound
+
 # deal with the X11 frame grabber
 enabled x11grab                         &&
 check_header X11/Xlib.h                 &&
diff --git a/libavdevice/Makefile b/libavdevice/Makefile
index 31337766ea..8bd706b03f 100644
--- a/libavdevice/Makefile
+++ b/libavdevice/Makefile
@@ -8,6 +8,8 @@ HEADERS = avdevice.h
 OBJS    = alldevices.o
 
 # input/output devices
+OBJS-$(CONFIG_ALSA_DEMUXER)              += alsa-audio-common.o alsa-audio-dec.o
+OBJS-$(CONFIG_ALSA_MUXER)                += alsa-audio-common.o alsa-audio-enc.o
 OBJS-$(CONFIG_BKTR_DEMUXER)              += bktr.o
 OBJS-$(CONFIG_DV1394_DEMUXER)            += dv1394.o
 OBJS-$(CONFIG_OSS_DEMUXER)               += oss_audio.o
diff --git a/libavdevice/alldevices.c b/libavdevice/alldevices.c
index 342e26ec10..38ce6f1eca 100644
--- a/libavdevice/alldevices.c
+++ b/libavdevice/alldevices.c
@@ -44,6 +44,7 @@ void avdevice_register_all(void)
     initialized = 1;
 
     /* devices */
+    REGISTER_MUXDEMUX (ALSA, alsa);
     REGISTER_MUXDEMUX (AUDIO_BEOS, audio_beos);
     REGISTER_DEMUXER  (BKTR, bktr);
     REGISTER_DEMUXER  (DV1394, dv1394);
diff --git a/libavdevice/alsa-audio-common.c b/libavdevice/alsa-audio-common.c
new file mode 100644
index 0000000000..1cb2640998
--- /dev/null
+++ b/libavdevice/alsa-audio-common.c
@@ -0,0 +1,186 @@
+/*
+ * ALSA input and output
+ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
+ * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file alsa-audio-common.c
+ * ALSA input and output: common code
+ * @author Luca Abeni ( lucabe72 email it )
+ * @author Benoit Fouet ( benoit fouet free fr )
+ * @author Nicolas George ( nicolas george normalesup org )
+ */
+
+#include "libavformat/avformat.h"
+#include <alsa/asoundlib.h>
+
+#include "alsa-audio.h"
+
+static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
+{
+    switch(codec_id) {
+        case CODEC_ID_PCM_S16LE: return SND_PCM_FORMAT_S16_LE;
+        case CODEC_ID_PCM_S16BE: return SND_PCM_FORMAT_S16_BE;
+        case CODEC_ID_PCM_S8:    return SND_PCM_FORMAT_S8;
+        default:                 return SND_PCM_FORMAT_UNKNOWN;
+    }
+}
+
+av_cold int ff_alsa_open(AVFormatContext *ctx, int mode,
+                         unsigned int *sample_rate,
+                         int channels, int *codec_id)
+{
+    AlsaData *s = ctx->priv_data;
+    const char *audio_device;
+    int res, flags = 0;
+    snd_pcm_format_t format;
+    snd_pcm_t *h;
+    snd_pcm_hw_params_t *hw_params;
+    snd_pcm_uframes_t buffer_size, period_size;
+
+    if (ctx->filename[0] == 0) audio_device = "default";
+    else                       audio_device = ctx->filename;
+
+    if (*codec_id == CODEC_ID_NONE)
+        *codec_id = DEFAULT_CODEC_ID;
+    format = codec_id_to_pcm_format(*codec_id);
+    if (format == SND_PCM_FORMAT_UNKNOWN) {
+        av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
+        return AVERROR(ENOSYS);
+    }
+    s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;
+
+    if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
+        flags = O_NONBLOCK;
+    }
+    res = snd_pcm_open(&h, audio_device, mode, flags);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
+               audio_device, snd_strerror(res));
+        return AVERROR_IO;
+    }
+
+    res = snd_pcm_hw_params_malloc(&hw_params);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n",
+               snd_strerror(res));
+        goto fail1;
+    }
+
+    res = snd_pcm_hw_params_any(h, hw_params);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n",
+               snd_strerror(res));
+        goto fail;
+    }
+
+    res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n",
+               snd_strerror(res));
+        goto fail;
+    }
+
+    res = snd_pcm_hw_params_set_format(h, hw_params, format);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n",
+               *codec_id, format, snd_strerror(res));
+        goto fail;
+    }
+
+    res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n",
+               snd_strerror(res));
+        goto fail;
+    }
+
+    res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
+               channels, snd_strerror(res));
+        goto fail;
+    }
+
+    snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
+    /* TODO: maybe use ctx->max_picture_buffer somehow */
+    res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n",
+               snd_strerror(res));
+        goto fail;
+    }
+
+    snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
+    res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n",
+               snd_strerror(res));
+        goto fail;
+    }
+    s->period_size = period_size;
+
+    res = snd_pcm_hw_params(h, hw_params);
+    if (res < 0) {
+        av_log(ctx, AV_LOG_ERROR, "cannot set parameters (%s)\n",
+               snd_strerror(res));
+        goto fail;
+    }
+
+    snd_pcm_hw_params_free(hw_params);
+
+    s->h = h;
+    return 0;
+
+fail:
+    snd_pcm_hw_params_free(hw_params);
+fail1:
+    snd_pcm_close(h);
+    return AVERROR_IO;
+}
+
+av_cold int ff_alsa_close(AVFormatContext *s1)
+{
+    AlsaData *s = s1->priv_data;
+
+    snd_pcm_close(s->h);
+    return 0;
+}
+
+int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
+{
+    AlsaData *s = s1->priv_data;
+    snd_pcm_t *handle = s->h;
+
+    av_log(s1, AV_LOG_WARNING, "ALSA buffer xrun.\n");
+    if (err == -EPIPE) {
+        err = snd_pcm_prepare(handle);
+        if (err < 0) {
+            av_log(s1, AV_LOG_ERROR, "cannot recover from underrun (snd_pcm_prepare failed: %s)\n", snd_strerror(err));
+
+            return AVERROR_IO;
+        }
+    } else if (err == -ESTRPIPE) {
+        av_log(s1, AV_LOG_ERROR, "-ESTRPIPE... Unsupported!\n");
+
+        return -1;
+    }
+    return err;
+}
diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c
new file mode 100644
index 0000000000..1be0a6c7eb
--- /dev/null
+++ b/libavdevice/alsa-audio-dec.c
@@ -0,0 +1,175 @@
+/*
+ * ALSA input and output
+ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
+ * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file alsa-audio-dec.c
+ * ALSA input and output: input
+ * @author Luca Abeni ( lucabe72 email it )
+ * @author Benoit Fouet ( benoit fouet free fr )
+ * @author Nicolas George ( nicolas george normalesup org )
+ *
+ * This avdevice decoder allows to capture audio from an ALSA (Advanced
+ * Linux Sound Architecture) device.
+ *
+ * The filename parameter is the name of an ALSA PCM device capable of
+ * capture, for example "default" or "plughw:1"; see the ALSA documentation
+ * for naming conventions. The empty string is equivalent to "default".
+ *
+ * The capture period is set to the lower value available for the device,
+ * which gives a low latency suitable for real-time capture.
+ *
+ * The PTS are an Unix time in microsecond.
+ *
+ * Due to a bug in the ALSA library
+ * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
+ * decoder does not work with certain ALSA plugins, especially the dsnoop
+ * plugin.
+ */
+
+#include "libavformat/avformat.h"
+#include <alsa/asoundlib.h>
+
+#include "alsa-audio.h"
+
+av_cold static int audio_read_header(AVFormatContext *s1,
+                                     AVFormatParameters *ap)
+{
+    AlsaData *s = s1->priv_data;
+    AVStream *st;
+    int ret;
+    unsigned int sample_rate;
+    int codec_id;
+    snd_pcm_sw_params_t *sw_params;
+
+    if (ap->sample_rate <= 0) {
+        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
+
+        return AVERROR(EIO);
+    }
+
+    if (ap->channels <= 0) {
+        av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
+
+        return AVERROR(EIO);
+    }
+
+    st = av_new_stream(s1, 0);
+    if (!st) {
+        av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
+
+        return AVERROR(ENOMEM);
+    }
+    sample_rate = ap->sample_rate;
+    codec_id    = ap->audio_codec_id;
+
+    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
+        &codec_id);
+    if (ret < 0) {
+        return AVERROR(EIO);
+    }
+
+    if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
+        av_log(s1, AV_LOG_WARNING,
+               "capture with some ALSA plugins, especially dsnoop, "
+               "may hang.\n");
+
+    ret = snd_pcm_sw_params_malloc(&sw_params);
+    if (ret < 0) {
+        av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
+               snd_strerror(ret));
+        goto fail;
+    }
+
+    snd_pcm_sw_params_current(s->h, sw_params);
+    snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
+
+    ret = snd_pcm_sw_params(s->h, sw_params);
+    snd_pcm_sw_params_free(sw_params);
+    if (ret < 0) {
+        av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
+               snd_strerror(ret));
+        goto fail;
+    }
+
+    /* take real parameters */
+    st->codec->codec_type  = CODEC_TYPE_AUDIO;
+    st->codec->codec_id    = codec_id;
+    st->codec->sample_rate = sample_rate;
+    st->codec->channels    = ap->channels;
+    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
+
+    return 0;
+
+fail:
+    snd_pcm_close(s->h);
+    return AVERROR(EIO);
+}
+
+static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+    AlsaData *s  = s1->priv_data;
+    AVStream *st = s1->streams[0];
+    int res;
+    snd_htimestamp_t timestamp;
+    snd_pcm_uframes_t ts_delay;
+
+    if (av_new_packet(pkt, s->period_size) < 0) {
+        return AVERROR(EIO);
+    }
+
+    while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
+        if (res == -EAGAIN) {
+            av_free_packet(pkt);
+
+            return AVERROR(EAGAIN);
+        }
+        if (ff_alsa_xrun_recover(s1, res) < 0) {
+            av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
+                   snd_strerror(res));
+            av_free_packet(pkt);
+
+            return AVERROR(EIO);
+        }
+    }
+
+    snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
+    ts_delay += res;
+    pkt->pts = timestamp.tv_sec * 1000000LL
+               + (timestamp.tv_nsec * st->codec->sample_rate
+                  - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
+               / (st->codec->sample_rate * 1000LL);
+
+    pkt->size = res * s->frame_size;
+
+    return 0;
+}
+
+AVInputFormat alsa_demuxer = {
+    "alsa",
+    NULL_IF_CONFIG_SMALL("ALSA audio input"),
+    sizeof(AlsaData),
+    NULL,
+    audio_read_header,
+    audio_read_packet,
+    ff_alsa_close,
+    .flags = AVFMT_NOFILE,
+};
diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c
new file mode 100644
index 0000000000..375ad63765
--- /dev/null
+++ b/libavdevice/alsa-audio-enc.c
@@ -0,0 +1,108 @@
+/*
+ * ALSA input and output
+ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
+ * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file alsa-audio-enc.c
+ * ALSA input and output: output
+ * @author Luca Abeni ( lucabe72 email it )
+ * @author Benoit Fouet ( benoit fouet free fr )
+ *
+ * This avdevice encoder allows to play audio to an ALSA (Advanced Linux
+ * Sound Architecture) device.
+ *
+ * The filename parameter is the name of an ALSA PCM device capable of
+ * capture, for example "default" or "plughw:1"; see the ALSA documentation
+ * for naming conventions. The empty string is equivalent to "default".
+ *
+ * The playback period is set to the lower value available for the device,
+ * which gives a low latency suitable for real-time playback.
+ */
+
+#include "libavformat/avformat.h"
+#include <alsa/asoundlib.h>
+
+#include "alsa-audio.h"
+
+av_cold static int audio_write_header(AVFormatContext *s1)
+{
+    AlsaData *s = s1->priv_data;
+    AVStream *st;
+    unsigned int sample_rate;
+    int codec_id;
+    int res;
+
+    st = s1->streams[0];
+    sample_rate = st->codec->sample_rate;
+    codec_id    = st->codec->codec_id;
+    res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
+        st->codec->channels, &codec_id);
+    if (sample_rate != st->codec->sample_rate) {
+        av_log(s1, AV_LOG_ERROR,
+               "sample rate %d not available, nearest is %d\n",
+               st->codec->sample_rate, sample_rate);
+        goto fail;
+    }
+
+    return res;
+
+fail:
+    snd_pcm_close(s->h);
+    return AVERROR(EIO);
+}
+
+static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+    AlsaData *s = s1->priv_data;
+    int res;
+    int size     = pkt->size;
+    uint8_t *buf = pkt->data;
+
+    while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) {
+        if (res == -EAGAIN) {
+
+            return AVERROR(EAGAIN);
+        }
+
+        if (ff_alsa_xrun_recover(s1, res) < 0) {
+            av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
+                   snd_strerror(res));
+
+            return AVERROR(EIO);
+        }
+    }
+
+    return 0;
+}
+
+AVOutputFormat alsa_muxer = {
+    "alsa",
+    NULL_IF_CONFIG_SMALL("ALSA audio output"),
+    "",
+    "",
+    sizeof(AlsaData),
+    DEFAULT_CODEC_ID,
+    CODEC_ID_NONE,
+    audio_write_header,
+    audio_write_packet,
+    ff_alsa_close,
+    .flags = AVFMT_NOFILE,
+};
diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h
new file mode 100644
index 0000000000..9547f790af
--- /dev/null
+++ b/libavdevice/alsa-audio.h
@@ -0,0 +1,84 @@
+/*
+ * ALSA input and output
+ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
+ * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file alsa-audio.h
+ * ALSA input and output: definitions and structures
+ * @author Luca Abeni ( lucabe72 email it )
+ * @author Benoit Fouet ( benoit fouet free fr )
+ */
+
+#ifndef AVDEVICE_ALSA_AUDIO_H
+#define AVDEVICE_ALSA_AUDIO_H
+
+/* XXX: we make the assumption that the soundcard accepts this format */
+/* XXX: find better solution with "preinit" method, needed also in
+        other formats */
+#ifdef WORDS_BIGENDIAN
+#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16BE
+#else
+#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE
+#endif
+
+typedef struct {
+    snd_pcm_t *h;
+    int frame_size;  ///< preferred size for reads and writes
+    int period_size; ///< bytes per sample * channels
+} AlsaData;
+
+/**
+ * Opens an ALSA PCM.
+ *
+ * @param s media file handle
+ * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
+ * @param sample_rate in: requested sample rate;
+ *                    out: actually selected sample rate
+ * @param channels number of channels
+ * @param codec_id in: requested CodecID or CODEC_ID_NONE;
+ *                 out: actually selected CodecID, changed only if
+ *                 CODEC_ID_NONE was requested
+ *
+ * @return 0 if OK, AVERROR_xxx on error
+ */
+int ff_alsa_open(AVFormatContext *s, int mode, unsigned int *sample_rate,
+                 int channels, int *codec_id);
+
+/**
+ * Closes the ALSA PCM.
+ *
+ * @param s1 media file handle
+ *
+ * @return 0
+ */
+int ff_alsa_close(AVFormatContext *s1);
+
+/**
+ * Tries to recover from ALSA buffer underrun.
+ *
+ * @param s1 media file handle
+ * @param err error code reported by the previous ALSA call
+ *
+ * @return 0 if OK, AVERROR_xxx on error
+ */
+int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
+
+#endif /* AVDEVICE_ALSA_AUDIO_H */