Libav 0.7.1
libavdevice/alsa-audio-common.c
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00001 /*
00002  * ALSA input and output
00003  * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
00004  * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
00005  *
00006  * This file is part of Libav.
00007  *
00008  * Libav is free software; you can redistribute it and/or
00009  * modify it under the terms of the GNU Lesser General Public
00010  * License as published by the Free Software Foundation; either
00011  * version 2.1 of the License, or (at your option) any later version.
00012  *
00013  * Libav is distributed in the hope that it will be useful,
00014  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00015  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00016  * Lesser General Public License for more details.
00017  *
00018  * You should have received a copy of the GNU Lesser General Public
00019  * License along with Libav; if not, write to the Free Software
00020  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00021  */
00022 
00031 #include <alsa/asoundlib.h>
00032 #include "libavformat/avformat.h"
00033 
00034 #include "alsa-audio.h"
00035 
00036 static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
00037 {
00038     switch(codec_id) {
00039         case CODEC_ID_PCM_F64LE: return SND_PCM_FORMAT_FLOAT64_LE;
00040         case CODEC_ID_PCM_F64BE: return SND_PCM_FORMAT_FLOAT64_BE;
00041         case CODEC_ID_PCM_F32LE: return SND_PCM_FORMAT_FLOAT_LE;
00042         case CODEC_ID_PCM_F32BE: return SND_PCM_FORMAT_FLOAT_BE;
00043         case CODEC_ID_PCM_S32LE: return SND_PCM_FORMAT_S32_LE;
00044         case CODEC_ID_PCM_S32BE: return SND_PCM_FORMAT_S32_BE;
00045         case CODEC_ID_PCM_U32LE: return SND_PCM_FORMAT_U32_LE;
00046         case CODEC_ID_PCM_U32BE: return SND_PCM_FORMAT_U32_BE;
00047         case CODEC_ID_PCM_S24LE: return SND_PCM_FORMAT_S24_3LE;
00048         case CODEC_ID_PCM_S24BE: return SND_PCM_FORMAT_S24_3BE;
00049         case CODEC_ID_PCM_U24LE: return SND_PCM_FORMAT_U24_3LE;
00050         case CODEC_ID_PCM_U24BE: return SND_PCM_FORMAT_U24_3BE;
00051         case CODEC_ID_PCM_S16LE: return SND_PCM_FORMAT_S16_LE;
00052         case CODEC_ID_PCM_S16BE: return SND_PCM_FORMAT_S16_BE;
00053         case CODEC_ID_PCM_U16LE: return SND_PCM_FORMAT_U16_LE;
00054         case CODEC_ID_PCM_U16BE: return SND_PCM_FORMAT_U16_BE;
00055         case CODEC_ID_PCM_S8:    return SND_PCM_FORMAT_S8;
00056         case CODEC_ID_PCM_U8:    return SND_PCM_FORMAT_U8;
00057         case CODEC_ID_PCM_MULAW: return SND_PCM_FORMAT_MU_LAW;
00058         case CODEC_ID_PCM_ALAW:  return SND_PCM_FORMAT_A_LAW;
00059         default:                 return SND_PCM_FORMAT_UNKNOWN;
00060     }
00061 }
00062 
00063 av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
00064                          unsigned int *sample_rate,
00065                          int channels, enum CodecID *codec_id)
00066 {
00067     AlsaData *s = ctx->priv_data;
00068     const char *audio_device;
00069     int res, flags = 0;
00070     snd_pcm_format_t format;
00071     snd_pcm_t *h;
00072     snd_pcm_hw_params_t *hw_params;
00073     snd_pcm_uframes_t buffer_size, period_size;
00074 
00075     if (ctx->filename[0] == 0) audio_device = "default";
00076     else                       audio_device = ctx->filename;
00077 
00078     if (*codec_id == CODEC_ID_NONE)
00079         *codec_id = DEFAULT_CODEC_ID;
00080     format = codec_id_to_pcm_format(*codec_id);
00081     if (format == SND_PCM_FORMAT_UNKNOWN) {
00082         av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
00083         return AVERROR(ENOSYS);
00084     }
00085     s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;
00086 
00087     if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
00088         flags = SND_PCM_NONBLOCK;
00089     }
00090     res = snd_pcm_open(&h, audio_device, mode, flags);
00091     if (res < 0) {
00092         av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
00093                audio_device, snd_strerror(res));
00094         return AVERROR(EIO);
00095     }
00096 
00097     res = snd_pcm_hw_params_malloc(&hw_params);
00098     if (res < 0) {
00099         av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n",
00100                snd_strerror(res));
00101         goto fail1;
00102     }
00103 
00104     res = snd_pcm_hw_params_any(h, hw_params);
00105     if (res < 0) {
00106         av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n",
00107                snd_strerror(res));
00108         goto fail;
00109     }
00110 
00111     res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
00112     if (res < 0) {
00113         av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n",
00114                snd_strerror(res));
00115         goto fail;
00116     }
00117 
00118     res = snd_pcm_hw_params_set_format(h, hw_params, format);
00119     if (res < 0) {
00120         av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n",
00121                *codec_id, format, snd_strerror(res));
00122         goto fail;
00123     }
00124 
00125     res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0);
00126     if (res < 0) {
00127         av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n",
00128                snd_strerror(res));
00129         goto fail;
00130     }
00131 
00132     res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
00133     if (res < 0) {
00134         av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
00135                channels, snd_strerror(res));
00136         goto fail;
00137     }
00138 
00139     snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
00140     buffer_size = FFMIN(buffer_size, ALSA_BUFFER_SIZE_MAX);
00141     /* TODO: maybe use ctx->max_picture_buffer somehow */
00142     res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size);
00143     if (res < 0) {
00144         av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n",
00145                snd_strerror(res));
00146         goto fail;
00147     }
00148 
00149     snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
00150     if (!period_size)
00151         period_size = buffer_size / 4;
00152     res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL);
00153     if (res < 0) {
00154         av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n",
00155                snd_strerror(res));
00156         goto fail;
00157     }
00158     s->period_size = period_size;
00159 
00160     res = snd_pcm_hw_params(h, hw_params);
00161     if (res < 0) {
00162         av_log(ctx, AV_LOG_ERROR, "cannot set parameters (%s)\n",
00163                snd_strerror(res));
00164         goto fail;
00165     }
00166 
00167     snd_pcm_hw_params_free(hw_params);
00168 
00169     s->h = h;
00170     return 0;
00171 
00172 fail:
00173     snd_pcm_hw_params_free(hw_params);
00174 fail1:
00175     snd_pcm_close(h);
00176     return AVERROR(EIO);
00177 }
00178 
00179 av_cold int ff_alsa_close(AVFormatContext *s1)
00180 {
00181     AlsaData *s = s1->priv_data;
00182 
00183     snd_pcm_close(s->h);
00184     return 0;
00185 }
00186 
00187 int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
00188 {
00189     AlsaData *s = s1->priv_data;
00190     snd_pcm_t *handle = s->h;
00191 
00192     av_log(s1, AV_LOG_WARNING, "ALSA buffer xrun.\n");
00193     if (err == -EPIPE) {
00194         err = snd_pcm_prepare(handle);
00195         if (err < 0) {
00196             av_log(s1, AV_LOG_ERROR, "cannot recover from underrun (snd_pcm_prepare failed: %s)\n", snd_strerror(err));
00197 
00198             return AVERROR(EIO);
00199         }
00200     } else if (err == -ESTRPIPE) {
00201         av_log(s1, AV_LOG_ERROR, "-ESTRPIPE... Unsupported!\n");
00202 
00203         return -1;
00204     }
00205     return err;
00206 }