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author | Michael Niedermayer <michaelni@gmx.at> | 2014-07-19 13:39:12 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2014-07-19 13:39:12 +0200 |
commit | 80acedae3ed55902b6a15ba5ea121a8dfc7a880a (patch) | |
tree | acf6807b88f8bcba1f0b1c4ba5455f3a41b67a57 /libavdevice/oss_audio.c | |
parent | 54cba3f53efd80442015a0ba5ba25252e8096290 (diff) | |
parent | d6e1d37100af568211f28ec0bcf7958a3a2a299e (diff) | |
download | ffmpeg-streaming-80acedae3ed55902b6a15ba5ea121a8dfc7a880a.zip ffmpeg-streaming-80acedae3ed55902b6a15ba5ea121a8dfc7a880a.tar.gz |
Merge commit 'd6e1d37100af568211f28ec0bcf7958a3a2a299e'
* commit 'd6e1d37100af568211f28ec0bcf7958a3a2a299e':
oss_audio: Split muxer and demuxer
Conflicts:
libavdevice/Makefile
libavdevice/oss_audio.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavdevice/oss_audio.c')
-rw-r--r-- | libavdevice/oss_audio.c | 219 |
1 files changed, 10 insertions, 209 deletions
diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c index fa338d6..d27da85 100644 --- a/libavdevice/oss_audio.c +++ b/libavdevice/oss_audio.c @@ -20,47 +20,32 @@ */ #include "config.h" -#include <stdlib.h> -#include <stdio.h> -#include <stdint.h> + #include <string.h> -#include <errno.h> + #if HAVE_SOUNDCARD_H #include <soundcard.h> #else #include <sys/soundcard.h> #endif + #if HAVE_UNISTD_H #include <unistd.h> #endif #include <fcntl.h> #include <sys/ioctl.h> -#include "libavutil/internal.h" #include "libavutil/log.h" -#include "libavutil/opt.h" -#include "libavutil/time.h" + #include "libavcodec/avcodec.h" #include "avdevice.h" -#include "libavformat/internal.h" -#define AUDIO_BLOCK_SIZE 4096 +#include "oss_audio.h" -typedef struct AudioData { - AVClass *class; - int fd; - int sample_rate; - int channels; - int frame_size; /* in bytes ! */ - enum AVCodecID codec_id; - unsigned int flip_left : 1; - uint8_t buffer[AUDIO_BLOCK_SIZE]; - int buffer_ptr; -} AudioData; - -static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device) +int ff_oss_audio_open(AVFormatContext *s1, int is_output, + const char *audio_device) { - AudioData *s = s1->priv_data; + OSSAudioData *s = s1->priv_data; int audio_fd; int tmp, err; char *flip = getenv("AUDIO_FLIP_LEFT"); @@ -85,7 +70,7 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi } } - s->frame_size = AUDIO_BLOCK_SIZE; + s->frame_size = OSS_AUDIO_BLOCK_SIZE; #define CHECK_IOCTL_ERROR(event) \ if (err < 0) { \ @@ -149,192 +134,8 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi #undef CHECK_IOCTL_ERROR } -static int audio_close(AudioData *s) +int ff_oss_audio_close(OSSAudioData *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(s1, 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 = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, 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) -{ - AudioData *s = s1->priv_data; - AVStream *st; - int ret; - - st = avformat_new_stream(s1, NULL); - if (!st) { - return AVERROR(ENOMEM); - } - - ret = audio_open(s1, 0, s1->filename); - if (ret < 0) { - return AVERROR(EIO); - } - - /* take real parameters */ - st->codec->codec_type = AVMEDIA_TYPE_AUDIO; - st->codec->codec_id = s->codec_id; - st->codec->sample_rate = s->sample_rate; - st->codec->channels = s->channels; - - avpriv_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 ((ret=av_new_packet(pkt, s->frame_size)) < 0) - return ret; - - ret = read(s->fd, pkt->data, pkt->size); - if (ret <= 0){ - av_free_packet(pkt); - pkt->size = 0; - if (ret<0) return AVERROR(errno); - else return AVERROR_EOF; - } - 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; - } - /* subtract 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; -} - -#if CONFIG_OSS_INDEV -static const AVOption options[] = { - { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, - { "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, - { NULL }, -}; - -static const AVClass oss_demuxer_class = { - .class_name = "OSS demuxer", - .item_name = av_default_item_name, - .option = options, - .version = LIBAVUTIL_VERSION_INT, - .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT, -}; - -AVInputFormat ff_oss_demuxer = { - .name = "oss", - .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"), - .priv_data_size = sizeof(AudioData), - .read_header = audio_read_header, - .read_packet = audio_read_packet, - .read_close = audio_read_close, - .flags = AVFMT_NOFILE, - .priv_class = &oss_demuxer_class, -}; -#endif - -#if CONFIG_OSS_OUTDEV -static const AVClass oss_muxer_class = { - .class_name = "OSS muxer", - .item_name = av_default_item_name, - .version = LIBAVUTIL_VERSION_INT, - .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT, -}; - -AVOutputFormat ff_oss_muxer = { - .name = "oss", - .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"), - .priv_data_size = 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 */ - .audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE), - .video_codec = AV_CODEC_ID_NONE, - .write_header = audio_write_header, - .write_packet = audio_write_packet, - .write_trailer = audio_write_trailer, - .flags = AVFMT_NOFILE, - .priv_class = &oss_muxer_class, -}; -#endif |