GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavdevice/alsa_enc.c Lines: 0 59 0.0 %
Date: 2020-09-25 23:16:12 Branches: 0 26 0.0 %

Line Branch Exec Source
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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 FFmpeg.
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 *
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 * FFmpeg 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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 * FFmpeg 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 FFmpeg; 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: output
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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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 * This avdevice encoder can play audio to an ALSA (Advanced Linux
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 * 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 playback period is set to the lower value available for the device,
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 * which gives a low latency suitable for real-time playback.
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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/time.h"
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#include "libavformat/internal.h"
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#include "avdevice.h"
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#include "alsa.h"
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static av_cold int audio_write_header(AVFormatContext *s1)
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{
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    AlsaData *s = s1->priv_data;
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    AVStream *st = NULL;
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    unsigned int sample_rate;
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    enum AVCodecID codec_id;
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    int res;
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    if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
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        av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
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        return AVERROR(EINVAL);
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    }
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    st = s1->streams[0];
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    sample_rate = st->codecpar->sample_rate;
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    codec_id    = st->codecpar->codec_id;
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    res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
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        st->codecpar->channels, &codec_id);
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    if (sample_rate != st->codecpar->sample_rate) {
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        av_log(s1, AV_LOG_ERROR,
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               "sample rate %d not available, nearest is %d\n",
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               st->codecpar->sample_rate, sample_rate);
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        goto fail;
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    }
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    avpriv_set_pts_info(st, 64, 1, sample_rate);
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    return res;
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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_write_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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    AlsaData *s = s1->priv_data;
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    int res;
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    int size     = pkt->size;
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    uint8_t *buf = pkt->data;
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    size /= s->frame_size;
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    if (pkt->dts != AV_NOPTS_VALUE)
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        s->timestamp = pkt->dts;
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    s->timestamp += pkt->duration ? pkt->duration : size;
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    if (s->reorder_func) {
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        if (size > s->reorder_buf_size)
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            if (ff_alsa_extend_reorder_buf(s, size))
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                return AVERROR(ENOMEM);
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        s->reorder_func(buf, s->reorder_buf, size);
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        buf = s->reorder_buf;
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    }
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    while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
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        if (res == -EAGAIN) {
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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 write error: %s\n",
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                   snd_strerror(res));
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            return AVERROR(EIO);
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        }
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    }
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    return 0;
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}
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static int audio_write_frame(AVFormatContext *s1, int stream_index,
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                             AVFrame **frame, unsigned flags)
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{
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    AlsaData *s = s1->priv_data;
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    AVPacket pkt;
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    /* ff_alsa_open() should have accepted only supported formats */
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    if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
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        return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ?
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               AVERROR(EINVAL) : 0;
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    /* set only used fields */
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    pkt.data     = (*frame)->data[0];
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    pkt.size     = (*frame)->nb_samples * s->frame_size;
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    pkt.dts      = (*frame)->pkt_dts;
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    pkt.duration = (*frame)->pkt_duration;
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    return audio_write_packet(s1, &pkt);
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}
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static void
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audio_get_output_timestamp(AVFormatContext *s1, int stream,
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    int64_t *dts, int64_t *wall)
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{
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    AlsaData *s  = s1->priv_data;
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    snd_pcm_sframes_t delay = 0;
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    *wall = av_gettime();
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    snd_pcm_delay(s->h, &delay);
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    *dts = s->timestamp - delay;
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}
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static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
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{
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    return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
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}
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static const AVClass alsa_muxer_class = {
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    .class_name     = "ALSA outdev",
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    .item_name      = av_default_item_name,
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    .version        = LIBAVUTIL_VERSION_INT,
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    .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
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};
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AVOutputFormat ff_alsa_muxer = {
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    .name           = "alsa",
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    .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
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    .priv_data_size = sizeof(AlsaData),
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    .audio_codec    = DEFAULT_CODEC_ID,
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    .video_codec    = AV_CODEC_ID_NONE,
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    .write_header   = audio_write_header,
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    .write_packet   = audio_write_packet,
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    .write_trailer  = ff_alsa_close,
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    .write_uncoded_frame = audio_write_frame,
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    .get_device_list = audio_get_device_list,
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    .get_output_timestamp = audio_get_output_timestamp,
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    .flags          = AVFMT_NOFILE,
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    .priv_class     = &alsa_muxer_class,
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};