From d46cd2498614ae770f6c93e89f7665239b947e1c Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Wed, 27 Sep 2017 15:24:58 +0200
Subject: [PATCH] alsa: Coalesce source files after outdev removal
---
libavdevice/Makefile | 2 +-
libavdevice/alsa.c | 190 +++++++++++++++++++++++++++++++++++++++--
libavdevice/alsa.h | 94 --------------------
libavdevice/alsa_dec.c | 178 --------------------------------------
4 files changed, 182 insertions(+), 282 deletions(-)
delete mode 100644 libavdevice/alsa.h
delete mode 100644 libavdevice/alsa_dec.c
diff --git a/libavdevice/Makefile b/libavdevice/Makefile
index c7c5a319eeb..1160088c329 100644
--- a/libavdevice/Makefile
+++ b/libavdevice/Makefile
@@ -10,7 +10,7 @@ OBJS = alldevices.o \
OBJS-$(HAVE_LIBC_MSVCRT) += file_open.o
# input devices
-OBJS-$(CONFIG_ALSA_INDEV) += alsa_dec.o alsa.o
+OBJS-$(CONFIG_ALSA_INDEV) += alsa.o
OBJS-$(CONFIG_AVFOUNDATION_INDEV) += avfoundation_dec.o
OBJS-$(CONFIG_BKTR_INDEV) += bktr.o
OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o
diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c
index 81c94049cb9..276a6c84cf2 100644
--- a/libavdevice/alsa.c
+++ b/libavdevice/alsa.c
@@ -1,5 +1,5 @@
/*
- * ALSA input and output
+ * ALSA input
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
*
@@ -22,18 +22,39 @@
/**
* @file
- * ALSA input and output: common code
+ * ALSA input
* @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr )
* @author Nicolas George ( nicolas george normalesup org )
*/
#include <alsa/asoundlib.h>
-#include "libavformat/avformat.h"
+
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+
+#include "libavformat/avformat.h"
+#include "libavformat/internal.h"
-#include "alsa.h"
+/* XXX: we make the assumption that the soundcard accepts this format */
+/* XXX: find better solution with "preinit" method, needed also in
+ other formats */
+#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
+
+#define ALSA_BUFFER_SIZE_MAX 32768
+
+typedef struct AlsaData {
+ AVClass *class;
+ snd_pcm_t *h;
+ int frame_size; ///< preferred size for reads and writes
+ int period_size; ///< bytes per sample * channels
+ int sample_rate; ///< sample rate set by user
+ int channels; ///< number of channels set by user
+ void (*reorder_func)(const void *, void *, int);
+ void *reorder_buf;
+ int reorder_buf_size; ///< in frames
+} AlsaData;
static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
{
@@ -183,9 +204,23 @@ static av_cold int find_reorder_func(AlsaData *s, int codec_id, uint64_t layout,
return s->reorder_func ? 0 : AVERROR(ENOSYS);
}
-av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
- unsigned int *sample_rate,
- int channels, enum AVCodecID *codec_id)
+/**
+ * Open an ALSA PCM.
+ *
+ * @param s media file handle
+ * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
+ * @param sample_rate in: requested sample rate;
+ * out: actually selected sample rate
+ * @param channels number of channels
+ * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
+ * out: actually selected AVCodecID, changed only if
+ * AV_CODEC_ID_NONE was requested
+ *
+ * @return 0 if OK, AVERROR_xxx on error
+ */
+static av_cold int alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
+ unsigned int *sample_rate,
+ int channels, enum AVCodecID *codec_id)
{
AlsaData *s = ctx->priv_data;
const char *audio_device;
@@ -315,7 +350,14 @@ fail1:
return AVERROR(EIO);
}
-av_cold int ff_alsa_close(AVFormatContext *s1)
+/**
+ * Close the ALSA PCM.
+ *
+ * @param s1 media file handle
+ *
+ * @return 0
+ */
+static av_cold int alsa_close(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
@@ -324,7 +366,15 @@ av_cold int ff_alsa_close(AVFormatContext *s1)
return 0;
}
-int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
+/**
+ * Try 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
+ */
+static int alsa_xrun_recover(AVFormatContext *s1, int err)
{
AlsaData *s = s1->priv_data;
snd_pcm_t *handle = s->h;
@@ -344,3 +394,125 @@ int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
}
return err;
}
+
+static av_cold int audio_read_header(AVFormatContext *s1)
+{
+ AlsaData *s = s1->priv_data;
+ AVStream *st;
+ int ret;
+ enum AVCodecID codec_id;
+ snd_pcm_sw_params_t *sw_params;
+
+ st = avformat_new_stream(s1, NULL);
+ if (!st) {
+ av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
+
+ return AVERROR(ENOMEM);
+ }
+ codec_id = s1->audio_codec_id;
+
+ ret = alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
+ &codec_id);
+ if (ret < 0) {
+ return AVERROR(EIO);
+ }
+
+ if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
+ av_log(s1, AV_LOG_WARNING,
+ "capture with some ALSA plugins, especially dsnoop, "
+ "may hang.\n");
+
+ 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));
+ goto fail;
+ }
+
+ snd_pcm_sw_params_current(s->h, sw_params);
+ snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
+
+ 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));
+ goto fail;
+ }
+
+ /* take real parameters */
+ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codecpar->codec_id = codec_id;
+ st->codecpar->sample_rate = s->sample_rate;
+ st->codecpar->channels = s->channels;
+ avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
+
+ return 0;
+
+fail:
+ 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 (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 = alsa_close,
+ .flags = AVFMT_NOFILE,
+ .priv_class = &alsa_demuxer_class,
+};
diff --git a/libavdevice/alsa.h b/libavdevice/alsa.h
deleted file mode 100644
index 773cd2faf8c..00000000000
--- a/libavdevice/alsa.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * 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: definitions and structures
- * @author Luca Abeni ( lucabe72 email it )
- * @author Benoit Fouet ( benoit fouet free fr )
- */
-
-#ifndef AVDEVICE_ALSA_H
-#define AVDEVICE_ALSA_H
-
-#include <alsa/asoundlib.h>
-#include "config.h"
-#include "libavformat/avformat.h"
-#include "libavutil/log.h"
-
-/* XXX: we make the assumption that the soundcard accepts this format */
-/* XXX: find better solution with "preinit" method, needed also in
- other formats */
-#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
-
-#define ALSA_BUFFER_SIZE_MAX 32768
-
-typedef struct AlsaData {
- AVClass *class;
- snd_pcm_t *h;
- int frame_size; ///< preferred size for reads and writes
- int period_size; ///< bytes per sample * channels
- int sample_rate; ///< sample rate set by user
- int channels; ///< number of channels set by user
- void (*reorder_func)(const void *, void *, int);
- void *reorder_buf;
- int reorder_buf_size; ///< in frames
-} AlsaData;
-
-/**
- * Open an ALSA PCM.
- *
- * @param s media file handle
- * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
- * @param sample_rate in: requested sample rate;
- * out: actually selected sample rate
- * @param channels number of channels
- * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
- * out: actually selected AVCodecID, changed only if
- * AV_CODEC_ID_NONE was requested
- *
- * @return 0 if OK, AVERROR_xxx on error
- */
-int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
- unsigned int *sample_rate,
- int channels, enum AVCodecID *codec_id);
-
-/**
- * Close the ALSA PCM.
- *
- * @param s1 media file handle
- *
- * @return 0
- */
-int ff_alsa_close(AVFormatContext *s1);
-
-/**
- * Try 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_H */
diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c
deleted file mode 100644
index 58bf1dd6a16..00000000000
--- a/libavdevice/alsa_dec.c
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * 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: input
- * @author Luca Abeni ( lucabe72 email it )
- * @author Benoit Fouet ( benoit fouet free fr )
- * @author Nicolas George ( nicolas george normalesup org )
- *
- * This avdevice decoder allows to capture audio from an ALSA (Advanced
- * Linux Sound Architecture) device.
- *
- * The filename parameter is the name of an ALSA PCM device capable of
- * capture, for example "default" or "plughw:1"; see the ALSA documentation
- * for naming conventions. The empty string is equivalent to "default".
- *
- * The capture period is set to the lower value available for the device,
- * which gives a low latency suitable for real-time capture.
- *
- * The PTS are an Unix time in microsecond.
- *
- * Due to a bug in the ALSA library
- * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
- * decoder does not work with certain ALSA plugins, especially the dsnoop
- * plugin.
- */
-
-#include <alsa/asoundlib.h>
-
-#include "libavutil/internal.h"
-#include "libavutil/opt.h"
-
-#include "libavformat/avformat.h"
-#include "libavformat/internal.h"
-
-#include "alsa.h"
-
-static av_cold int audio_read_header(AVFormatContext *s1)
-{
- AlsaData *s = s1->priv_data;
- AVStream *st;
- int ret;
- enum AVCodecID codec_id;
- snd_pcm_sw_params_t *sw_params;
-
- st = avformat_new_stream(s1, NULL);
- if (!st) {
- av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
-
- return AVERROR(ENOMEM);
- }
- codec_id = s1->audio_codec_id;
-
- ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
- &codec_id);
- if (ret < 0) {
- return AVERROR(EIO);
- }
-
- if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
- av_log(s1, AV_LOG_WARNING,
- "capture with some ALSA plugins, especially dsnoop, "
- "may hang.\n");
-
- 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));
- goto fail;
- }
-
- snd_pcm_sw_params_current(s->h, sw_params);
- snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
-
- 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));
- goto fail;
- }
-
- /* take real parameters */
- st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
- st->codecpar->codec_id = codec_id;
- st->codecpar->sample_rate = s->sample_rate;
- st->codecpar->channels = s->channels;
- avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
-
- return 0;
-
-fail:
- 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,
-};
--
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