[FFmpeg-devel] [PATCH] ALSA for libavdevice

Nicolas George nicolas.george
Fri Dec 12 18:38:47 CET 2008


Hi. Thanks for your review.


Le nonidi 19 frimaire, an CCXVII, Michael Niedermayer a ?crit?:
> > + */ 
> trailing whitespace

Fixed.

> > +static snd_pcm_format_t alsa_codec_id(int codec_id)
> ff2alsa_codec_id() or a similar name that makes it clear that its not the
> inverse

Renamed to codec_id_to_pcm_format.

> > +    switch(codec_id) {
> > +        case CODEC_ID_PCM_S16LE:
> > +            return SND_PCM_FORMAT_S16_LE;
> putting each case only on a single line and vertically aligned should be
> more readable

It seems to be.


> > +        av_log(NULL, AV_LOG_ERROR, "sample format %x is not supported\n", s->codec_id);
> null is a poor context and should be avoided where possible, there may
> be some cases where its too messy to avoid but here ctx can be used

Fixed.

> > +    av_log(s1, AV_LOG_WARNING, "XRUN!!!\n");
> This message can be improved i think

"ALSA buffer underrun.\n"

> > +static int audio_read_close(AVFormatContext *s1)
> > +{
> > +    AlsaData *s = s1->priv_data;
> > +
> > +    ff_alsa_close(s);
> > +    return 0;
> > +}
> silly wraper function

Fixed.

> > +    s->sample_rate = ap->sample_rate;
> > +    s->channels = ap->channels;
> > +    s->codec_id = ap->audio_codec_id;
> redundant, they are placed in st->codec already

Fixed, at the cost of a few more parameters to ff_alsa_open.

> > +    pkt->pts = av_gettime();   /* FIXME: We might need something better... */
> yes, this really needs to be improved, this is unacceptable unless alsa
> completely lacks functionality to do better

Changed to use snd_pcm_htimestamp.

> > +static int audio_write_trailer(AVFormatContext *s1)
> > +{
> > +    AlsaData *s = s1->priv_data;
> > +
> > +    ff_alsa_close(s);
> > +    return 0;
> > +}
> silly wraper function

Ditto.

> > +extern int ff_alsa_open(AVFormatContext *s, int mode);
> > +extern int ff_alsa_close(AlsaData *s);
> > +extern int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
> the extern is unneeded and they lack doxy commets explaining what they do

Done.

Regarding Luca's remarks: I added Benoit Fouet in the header. As for the
format detection, I do not think it would be worth the additional code.

Regards,

-- 
  Nicolas George
-------------- next part --------------
diff --git a/configure b/configure
index 7b0f615..24a8aa7 100755
--- a/configure
+++ b/configure
@@ -819,6 +819,7 @@ ARCH_EXT_LIST='
 HAVE_LIST="
     $ARCH_EXT_LIST
     $THREADS_LIST
+    alsa_asoundlib_h
     altivec_h
     arpa_inet_h
     bswap
@@ -1037,6 +1038,10 @@ mpeg4aac_decoder_deps="libfaad"
 
 # demuxers / muxers
 ac3_demuxer_deps="ac3_parser"
+alsa_demuxer_deps="alsa_asoundlib_h"
+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"
@@ -2007,6 +2012,8 @@ check_header dev/ic/bt8xx.h
 check_header sys/soundcard.h
 check_header soundcard.h
 
+check_header alsa/asoundlib.h
+
 # deal with the X11 frame grabber
 enabled x11grab                         &&
 check_header X11/Xlib.h                 &&
diff --git a/libavdevice/Makefile b/libavdevice/Makefile
index 655c033..3ab27a0 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)               += audio.o
diff --git a/libavdevice/alldevices.c b/libavdevice/alldevices.c
index b94db63..bfce1cd 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 0000000..f80275e
--- /dev/null
+++ b/libavdevice/alsa-audio-common.c
@@ -0,0 +1,188 @@
+/*
+ * 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 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;
+    }
+}
+
+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 %x 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 %d %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;
+}
+
+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 underrun.\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 0;
+        }
+    } else if (err == -ESTRPIPE) {
+        av_log(NULL, AV_LOG_ERROR, "-ESTPIPE... 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 0000000..4123508
--- /dev/null
+++ b/libavdevice/alsa-audio-dec.c
@@ -0,0 +1,157 @@
+/*
+ * 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 )
+ */
+
+#include "libavformat/avformat.h"
+#include <alsa/asoundlib.h>
+
+#include "alsa-audio.h"
+
+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 || ap->channels <= 0) {
+        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d or channels number %d\n",
+                   ap->sample_rate, ap->channels);
+
+        return AVERROR(EIO);
+    }
+
+    st = av_new_stream(s1, 0);
+    if (st == NULL) {
+        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);
+    }
+
+    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));
+        snd_pcm_close(s->h);
+        return AVERROR(EIO);
+    }
+
+    snd_pcm_sw_params_current(s->h, sw_params);
+
+    ret = snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params,
+        SND_PCM_TSTAMP_ENABLE);
+    if (ret < 0) {
+        av_log(s1, AV_LOG_ERROR, "cannot set ALSA timestamp mode (%s)\n",
+            snd_strerror(ret));
+        snd_pcm_close(s->h);
+        return AVERROR(EIO);
+    }
+
+    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));
+        snd_pcm_close(s->h);
+         return AVERROR(EIO);
+     }
+
+    /* 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_log(NULL, AV_LOG_INFO, "sample rate: %d\n", sample_rate);
+    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
+
+    return 0;
+}
+
+#include <sys/time.h>
+
+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) {
+            pkt->size = 0;
+            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 = (int64_t)timestamp.tv_sec * 1000000 + timestamp.tv_nsec / 1000;
+    pkt->pts -= (int64_t)ts_delay * 1000000 / st->codec->sample_rate;
+
+    pkt->size = res * s->frame_size;
+
+    return 0;
+}
+
+AVInputFormat alsa_demuxer = {
+    "alsa",
+    "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 0000000..ce9eb5d
--- /dev/null
+++ b/libavdevice/alsa-audio-enc.c
@@ -0,0 +1,87 @@
+/*
+ * 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 )
+ */
+
+#include "libavformat/avformat.h"
+#include <alsa/asoundlib.h>
+
+#include "alsa-audio.h"
+
+static int audio_write_header(AVFormatContext *s1)
+{
+    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);
+
+    return res;
+}
+
+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",
+    "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 0000000..9af2170
--- /dev/null
+++ b/libavdevice/alsa-audio.h
@@ -0,0 +1,81 @@
+/*
+ * 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 sample rate
+ * @param channels number of channels
+ * @param codec_id CodecID
+ *
+ * @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 */
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