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https://github.com/xenia-project/FFmpeg.git
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alsa: Coalesce source files after outdev removal
This commit is contained in:
Diego Biurrun
parent
6ce13070bd
commit
d46cd24986
@ -10,7 +10,7 @@ OBJS = alldevices.o \
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OBJS-$(HAVE_LIBC_MSVCRT) += file_open.o
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# input devices
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OBJS-$(CONFIG_ALSA_INDEV) += alsa_dec.o alsa.o
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OBJS-$(CONFIG_ALSA_INDEV) += alsa.o
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OBJS-$(CONFIG_AVFOUNDATION_INDEV) += avfoundation_dec.o
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OBJS-$(CONFIG_BKTR_INDEV) += bktr.o
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OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o
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@ -1,5 +1,5 @@
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/*
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* ALSA input and output
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* ALSA input
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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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@ -22,18 +22,39 @@
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/**
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* @file
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* ALSA input and output: common code
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* ALSA input
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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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* @author Nicolas George ( nicolas george normalesup org )
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*/
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#include <alsa/asoundlib.h>
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#include "libavformat/avformat.h"
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "alsa.h"
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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/* XXX: we make the assumption that the soundcard accepts this format */
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/* XXX: find better solution with "preinit" method, needed also in
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other formats */
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#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
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#define ALSA_BUFFER_SIZE_MAX 32768
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typedef struct AlsaData {
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AVClass *class;
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snd_pcm_t *h;
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int frame_size; ///< preferred size for reads and writes
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int period_size; ///< bytes per sample * channels
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int sample_rate; ///< sample rate set by user
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int channels; ///< number of channels set by user
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void (*reorder_func)(const void *, void *, int);
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void *reorder_buf;
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int reorder_buf_size; ///< in frames
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} AlsaData;
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static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
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{
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@ -183,9 +204,23 @@ static av_cold int find_reorder_func(AlsaData *s, int codec_id, uint64_t layout,
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return s->reorder_func ? 0 : AVERROR(ENOSYS);
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}
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av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
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unsigned int *sample_rate,
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int channels, enum AVCodecID *codec_id)
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/**
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* Open an ALSA PCM.
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*
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* @param s media file handle
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* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
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* @param sample_rate in: requested sample rate;
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* out: actually selected sample rate
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* @param channels number of channels
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* @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
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* out: actually selected AVCodecID, changed only if
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* AV_CODEC_ID_NONE was requested
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*
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* @return 0 if OK, AVERROR_xxx on error
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*/
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static av_cold int alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
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unsigned int *sample_rate,
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int channels, enum AVCodecID *codec_id)
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{
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AlsaData *s = ctx->priv_data;
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const char *audio_device;
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@ -315,7 +350,14 @@ fail1:
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return AVERROR(EIO);
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}
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av_cold int ff_alsa_close(AVFormatContext *s1)
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/**
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* Close the ALSA PCM.
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*
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* @param s1 media file handle
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*
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* @return 0
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*/
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static av_cold int alsa_close(AVFormatContext *s1)
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{
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AlsaData *s = s1->priv_data;
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@ -324,7 +366,15 @@ av_cold int ff_alsa_close(AVFormatContext *s1)
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return 0;
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}
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int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
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/**
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* Try to recover from ALSA buffer underrun.
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*
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* @param s1 media file handle
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* @param err error code reported by the previous ALSA call
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*
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* @return 0 if OK, AVERROR_xxx on error
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*/
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static int alsa_xrun_recover(AVFormatContext *s1, int err)
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{
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AlsaData *s = s1->priv_data;
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snd_pcm_t *handle = s->h;
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@ -344,3 +394,125 @@ int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
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}
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return err;
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}
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static av_cold int audio_read_header(AVFormatContext *s1)
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{
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AlsaData *s = s1->priv_data;
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AVStream *st;
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int ret;
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enum AVCodecID codec_id;
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snd_pcm_sw_params_t *sw_params;
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st = avformat_new_stream(s1, NULL);
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if (!st) {
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av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
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return AVERROR(ENOMEM);
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}
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codec_id = s1->audio_codec_id;
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ret = alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
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&codec_id);
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if (ret < 0) {
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return AVERROR(EIO);
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}
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if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
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av_log(s1, AV_LOG_WARNING,
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"capture with some ALSA plugins, especially dsnoop, "
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"may hang.\n");
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ret = snd_pcm_sw_params_malloc(&sw_params);
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if (ret < 0) {
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av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
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snd_strerror(ret));
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goto fail;
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}
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snd_pcm_sw_params_current(s->h, sw_params);
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snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
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ret = snd_pcm_sw_params(s->h, sw_params);
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snd_pcm_sw_params_free(sw_params);
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if (ret < 0) {
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av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
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snd_strerror(ret));
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goto fail;
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}
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/* take real parameters */
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = codec_id;
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st->codecpar->sample_rate = s->sample_rate;
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st->codecpar->channels = s->channels;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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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_read_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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AlsaData *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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int res;
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snd_htimestamp_t timestamp;
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snd_pcm_uframes_t ts_delay;
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if (av_new_packet(pkt, s->period_size) < 0) {
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return AVERROR(EIO);
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}
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while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
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if (res == -EAGAIN) {
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av_packet_unref(pkt);
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return AVERROR(EAGAIN);
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}
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if (alsa_xrun_recover(s1, res) < 0) {
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av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
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snd_strerror(res));
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av_packet_unref(pkt);
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return AVERROR(EIO);
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}
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}
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snd_pcm_htimestamp(s->h, &ts_delay, ×tamp);
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ts_delay += res;
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pkt->pts = timestamp.tv_sec * 1000000LL
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+ (timestamp.tv_nsec * st->codecpar->sample_rate
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- (int64_t)ts_delay * 1000000000LL + st->codecpar->sample_rate * 500LL)
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/ (st->codecpar->sample_rate * 1000LL);
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pkt->size = res * s->frame_size;
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return 0;
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}
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static const AVOption options[] = {
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{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass alsa_demuxer_class = {
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.class_name = "ALSA demuxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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AVInputFormat ff_alsa_demuxer = {
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.name = "alsa",
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.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
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.priv_data_size = sizeof(AlsaData),
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.read_header = audio_read_header,
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.read_packet = audio_read_packet,
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.read_close = alsa_close,
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.flags = AVFMT_NOFILE,
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.priv_class = &alsa_demuxer_class,
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};
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@ -1,94 +0,0 @@
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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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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: definitions and structures
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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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#ifndef AVDEVICE_ALSA_H
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#define AVDEVICE_ALSA_H
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#include <alsa/asoundlib.h>
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#include "config.h"
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#include "libavformat/avformat.h"
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#include "libavutil/log.h"
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/* XXX: we make the assumption that the soundcard accepts this format */
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/* XXX: find better solution with "preinit" method, needed also in
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other formats */
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#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
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#define ALSA_BUFFER_SIZE_MAX 32768
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typedef struct AlsaData {
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AVClass *class;
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snd_pcm_t *h;
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int frame_size; ///< preferred size for reads and writes
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int period_size; ///< bytes per sample * channels
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int sample_rate; ///< sample rate set by user
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int channels; ///< number of channels set by user
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void (*reorder_func)(const void *, void *, int);
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void *reorder_buf;
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int reorder_buf_size; ///< in frames
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} AlsaData;
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/**
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* Open an ALSA PCM.
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*
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* @param s media file handle
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* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
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* @param sample_rate in: requested sample rate;
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* out: actually selected sample rate
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* @param channels number of channels
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* @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
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* out: actually selected AVCodecID, changed only if
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* AV_CODEC_ID_NONE was requested
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*
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* @return 0 if OK, AVERROR_xxx on error
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*/
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int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
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unsigned int *sample_rate,
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int channels, enum AVCodecID *codec_id);
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/**
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* Close the ALSA PCM.
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*
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* @param s1 media file handle
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*
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* @return 0
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*/
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int ff_alsa_close(AVFormatContext *s1);
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/**
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* Try to recover from ALSA buffer underrun.
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*
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* @param s1 media file handle
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* @param err error code reported by the previous ALSA call
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*
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* @return 0 if OK, AVERROR_xxx on error
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*/
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int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
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#endif /* AVDEVICE_ALSA_H */
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@ -1,178 +0,0 @@
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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 Libav.
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*
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* Libav is free software; you can redistribute it and/or
|
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* modify it under the terms of the GNU Lesser General Public
|
||||
* 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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* Libav 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
|
||||
* 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
|
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*/
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|
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/**
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* @file
|
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* ALSA input and output: input
|
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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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* @author Nicolas George ( nicolas george normalesup org )
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*
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* This avdevice decoder allows to capture audio from an ALSA (Advanced
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* Linux 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 capture period is set to the lower value available for the device,
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* which gives a low latency suitable for real-time capture.
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*
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* The PTS are an Unix time in microsecond.
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*
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* Due to a bug in the ALSA library
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* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
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* decoder does not work with certain ALSA plugins, especially the dsnoop
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* plugin.
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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/opt.h"
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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#include "alsa.h"
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static av_cold int audio_read_header(AVFormatContext *s1)
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{
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AlsaData *s = s1->priv_data;
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AVStream *st;
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int ret;
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enum AVCodecID codec_id;
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snd_pcm_sw_params_t *sw_params;
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st = avformat_new_stream(s1, NULL);
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if (!st) {
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av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
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return AVERROR(ENOMEM);
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}
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codec_id = s1->audio_codec_id;
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ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
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&codec_id);
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if (ret < 0) {
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return AVERROR(EIO);
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}
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if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
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av_log(s1, AV_LOG_WARNING,
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"capture with some ALSA plugins, especially dsnoop, "
|
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"may hang.\n");
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ret = snd_pcm_sw_params_malloc(&sw_params);
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if (ret < 0) {
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av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
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snd_strerror(ret));
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goto fail;
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}
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snd_pcm_sw_params_current(s->h, sw_params);
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snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
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ret = snd_pcm_sw_params(s->h, sw_params);
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snd_pcm_sw_params_free(sw_params);
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if (ret < 0) {
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av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
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snd_strerror(ret));
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goto fail;
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}
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/* take real parameters */
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = codec_id;
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st->codecpar->sample_rate = s->sample_rate;
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st->codecpar->channels = s->channels;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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fail:
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||||
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_packet_unref(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_packet_unref(pkt);
|
||||
|
||||
return AVERROR(EIO);
|
||||
}
|
||||
}
|
||||
|
||||
snd_pcm_htimestamp(s->h, &ts_delay, ×tamp);
|
||||
ts_delay += res;
|
||||
pkt->pts = timestamp.tv_sec * 1000000LL
|
||||
+ (timestamp.tv_nsec * st->codecpar->sample_rate
|
||||
- (int64_t)ts_delay * 1000000000LL + st->codecpar->sample_rate * 500LL)
|
||||
/ (st->codecpar->sample_rate * 1000LL);
|
||||
|
||||
pkt->size = res * s->frame_size;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static const AVOption options[] = {
|
||||
{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
|
||||
{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
|
||||
{ NULL },
|
||||
};
|
||||
|
||||
static const AVClass alsa_demuxer_class = {
|
||||
.class_name = "ALSA demuxer",
|
||||
.item_name = av_default_item_name,
|
||||
.option = options,
|
||||
.version = LIBAVUTIL_VERSION_INT,
|
||||
};
|
||||
|
||||
AVInputFormat ff_alsa_demuxer = {
|
||||
.name = "alsa",
|
||||
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
|
||||
.priv_data_size = sizeof(AlsaData),
|
||||
.read_header = audio_read_header,
|
||||
.read_packet = audio_read_packet,
|
||||
.read_close = ff_alsa_close,
|
||||
.flags = AVFMT_NOFILE,
|
||||
.priv_class = &alsa_demuxer_class,
|
||||
};
|
||||
Loading…
Reference in New Issue
Block a user