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-rw-r--r--libavformat/audio.c344
1 files changed, 0 insertions, 344 deletions
diff --git a/libavformat/audio.c b/libavformat/audio.c
deleted file mode 100644
index 151cbffd51..0000000000
--- a/libavformat/audio.c
+++ /dev/null
@@ -1,344 +0,0 @@
-/*
- * Linux audio play and grab interface
- * Copyright (c) 2000, 2001 Fabrice Bellard.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg 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.
- *
- * FFmpeg 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 FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-#include "avformat.h"
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#ifdef HAVE_SOUNDCARD_H
-#include <soundcard.h>
-#else
-#include <sys/soundcard.h>
-#endif
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/ioctl.h>
-#include <sys/mman.h>
-#include <sys/time.h>
-
-#define AUDIO_BLOCK_SIZE 4096
-
-typedef struct {
- int fd;
- int sample_rate;
- int channels;
- int frame_size; /* in bytes ! */
- int codec_id;
- int flip_left : 1;
- uint8_t buffer[AUDIO_BLOCK_SIZE];
- int buffer_ptr;
-} AudioData;
-
-static int audio_open(AudioData *s, int is_output, const char *audio_device)
-{
- int audio_fd;
- int tmp, err;
- char *flip = getenv("AUDIO_FLIP_LEFT");
-
- if (is_output)
- audio_fd = open(audio_device, O_WRONLY);
- else
- audio_fd = open(audio_device, O_RDONLY);
- if (audio_fd < 0) {
- av_log(NULL, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
- return AVERROR(EIO);
- }
-
- if (flip && *flip == '1') {
- s->flip_left = 1;
- }
-
- /* non blocking mode */
- if (!is_output)
- fcntl(audio_fd, F_SETFL, O_NONBLOCK);
-
- s->frame_size = AUDIO_BLOCK_SIZE;
-#if 0
- tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
- err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
- if (err < 0) {
- perror("SNDCTL_DSP_SETFRAGMENT");
- }
-#endif
-
- /* select format : favour native format */
- err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
-
-#ifdef WORDS_BIGENDIAN
- if (tmp & AFMT_S16_BE) {
- tmp = AFMT_S16_BE;
- } else if (tmp & AFMT_S16_LE) {
- tmp = AFMT_S16_LE;
- } else {
- tmp = 0;
- }
-#else
- if (tmp & AFMT_S16_LE) {
- tmp = AFMT_S16_LE;
- } else if (tmp & AFMT_S16_BE) {
- tmp = AFMT_S16_BE;
- } else {
- tmp = 0;
- }
-#endif
-
- switch(tmp) {
- case AFMT_S16_LE:
- s->codec_id = CODEC_ID_PCM_S16LE;
- break;
- case AFMT_S16_BE:
- s->codec_id = CODEC_ID_PCM_S16BE;
- break;
- default:
- av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
- close(audio_fd);
- return AVERROR(EIO);
- }
- err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
- if (err < 0) {
- av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
- goto fail;
- }
-
- tmp = (s->channels == 2);
- err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
- if (err < 0) {
- av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
- goto fail;
- }
- if (tmp)
- s->channels = 2;
-
- tmp = s->sample_rate;
- err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
- if (err < 0) {
- av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
- goto fail;
- }
- s->sample_rate = tmp; /* store real sample rate */
- s->fd = audio_fd;
-
- return 0;
- fail:
- close(audio_fd);
- return AVERROR(EIO);
-}
-
-static int audio_close(AudioData *s)
-{
- close(s->fd);
- return 0;
-}
-
-/* sound output support */
-static int audio_write_header(AVFormatContext *s1)
-{
- AudioData *s = s1->priv_data;
- AVStream *st;
- int ret;
-
- st = s1->streams[0];
- s->sample_rate = st->codec->sample_rate;
- s->channels = st->codec->channels;
- ret = audio_open(s, 1, s1->filename);
- if (ret < 0) {
- return AVERROR(EIO);
- } else {
- return 0;
- }
-}
-
-static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
-{
- AudioData *s = s1->priv_data;
- int len, ret;
- int size= pkt->size;
- uint8_t *buf= pkt->data;
-
- while (size > 0) {
- len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
- if (len > size)
- len = size;
- memcpy(s->buffer + s->buffer_ptr, buf, len);
- s->buffer_ptr += len;
- if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
- for(;;) {
- ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
- if (ret > 0)
- break;
- if (ret < 0 && (errno != EAGAIN && errno != EINTR))
- return AVERROR(EIO);
- }
- s->buffer_ptr = 0;
- }
- buf += len;
- size -= len;
- }
- return 0;
-}
-
-static int audio_write_trailer(AVFormatContext *s1)
-{
- AudioData *s = s1->priv_data;
-
- audio_close(s);
- return 0;
-}
-
-/* grab support */
-
-static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
-{
- AudioData *s = s1->priv_data;
- AVStream *st;
- int ret;
-
- if (ap->sample_rate <= 0 || ap->channels <= 0)
- return -1;
-
- st = av_new_stream(s1, 0);
- if (!st) {
- return AVERROR(ENOMEM);
- }
- s->sample_rate = ap->sample_rate;
- s->channels = ap->channels;
-
- ret = audio_open(s, 0, s1->filename);
- if (ret < 0) {
- av_free(st);
- return AVERROR(EIO);
- }
-
- /* take real parameters */
- st->codec->codec_type = CODEC_TYPE_AUDIO;
- st->codec->codec_id = s->codec_id;
- st->codec->sample_rate = s->sample_rate;
- st->codec->channels = s->channels;
-
- av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
- return 0;
-}
-
-static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
-{
- AudioData *s = s1->priv_data;
- int ret, bdelay;
- int64_t cur_time;
- struct audio_buf_info abufi;
-
- if (av_new_packet(pkt, s->frame_size) < 0)
- return AVERROR(EIO);
- for(;;) {
- struct timeval tv;
- fd_set fds;
-
- tv.tv_sec = 0;
- tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
-
- FD_ZERO(&fds);
- FD_SET(s->fd, &fds);
-
- /* This will block until data is available or we get a timeout */
- (void) select(s->fd + 1, &fds, 0, 0, &tv);
-
- ret = read(s->fd, pkt->data, pkt->size);
- if (ret > 0)
- break;
- if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
- av_free_packet(pkt);
- pkt->size = 0;
- pkt->pts = av_gettime();
- return 0;
- }
- if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
- av_free_packet(pkt);
- return AVERROR(EIO);
- }
- }
- pkt->size = ret;
-
- /* compute pts of the start of the packet */
- cur_time = av_gettime();
- bdelay = ret;
- if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
- bdelay += abufi.bytes;
- }
- /* substract time represented by the number of bytes in the audio fifo */
- cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
-
- /* convert to wanted units */
- pkt->pts = cur_time;
-
- if (s->flip_left && s->channels == 2) {
- int i;
- short *p = (short *) pkt->data;
-
- for (i = 0; i < ret; i += 4) {
- *p = ~*p;
- p += 2;
- }
- }
- return 0;
-}
-
-static int audio_read_close(AVFormatContext *s1)
-{
- AudioData *s = s1->priv_data;
-
- audio_close(s);
- return 0;
-}
-
-#ifdef CONFIG_OSS_DEMUXER
-AVInputFormat oss_demuxer = {
- "oss",
- "audio grab and output",
- sizeof(AudioData),
- NULL,
- audio_read_header,
- audio_read_packet,
- audio_read_close,
- .flags = AVFMT_NOFILE,
-};
-#endif
-
-#ifdef CONFIG_OSS_MUXER
-AVOutputFormat oss_muxer = {
- "oss",
- "audio grab and output",
- "",
- "",
- sizeof(AudioData),
- /* XXX: we make the assumption that the soundcard accepts this format */
- /* XXX: find better solution with "preinit" method, needed also in
- other formats */
-#ifdef WORDS_BIGENDIAN
- CODEC_ID_PCM_S16BE,
-#else
- CODEC_ID_PCM_S16LE,
-#endif
- CODEC_ID_NONE,
- audio_write_header,
- audio_write_packet,
- audio_write_trailer,
- .flags = AVFMT_NOFILE,
-};
-#endif
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