/* * 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 Libav. * * Libav 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. * * Libav 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 Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * 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 #include "libavformat/avformat.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_F64LE: return SND_PCM_FORMAT_FLOAT64_LE; case CODEC_ID_PCM_F64BE: return SND_PCM_FORMAT_FLOAT64_BE; case CODEC_ID_PCM_F32LE: return SND_PCM_FORMAT_FLOAT_LE; case CODEC_ID_PCM_F32BE: return SND_PCM_FORMAT_FLOAT_BE; case CODEC_ID_PCM_S32LE: return SND_PCM_FORMAT_S32_LE; case CODEC_ID_PCM_S32BE: return SND_PCM_FORMAT_S32_BE; case CODEC_ID_PCM_U32LE: return SND_PCM_FORMAT_U32_LE; case CODEC_ID_PCM_U32BE: return SND_PCM_FORMAT_U32_BE; case CODEC_ID_PCM_S24LE: return SND_PCM_FORMAT_S24_3LE; case CODEC_ID_PCM_S24BE: return SND_PCM_FORMAT_S24_3BE; case CODEC_ID_PCM_U24LE: return SND_PCM_FORMAT_U24_3LE; case CODEC_ID_PCM_U24BE: return SND_PCM_FORMAT_U24_3BE; 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_U16LE: return SND_PCM_FORMAT_U16_LE; case CODEC_ID_PCM_U16BE: return SND_PCM_FORMAT_U16_BE; case CODEC_ID_PCM_S8: return SND_PCM_FORMAT_S8; case CODEC_ID_PCM_U8: return SND_PCM_FORMAT_U8; case CODEC_ID_PCM_MULAW: return SND_PCM_FORMAT_MU_LAW; case CODEC_ID_PCM_ALAW: return SND_PCM_FORMAT_A_LAW; default: return SND_PCM_FORMAT_UNKNOWN; } } av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode, unsigned int *sample_rate, int channels, enum CodecID *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 = SND_PCM_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(EIO); } 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(EIO); } 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(EIO); } } else if (err == -ESTRPIPE) { av_log(s1, AV_LOG_ERROR, "-ESTRPIPE... Unsupported!\n"); return -1; } return err; }