48 #include <alsa/asoundlib.h>
50 #include "libavutil/opt.h"
51 #include "libavutil/mathematics.h"
52 #include "libavutil/time.h"
102 snd_pcm_sframes_t delay = 0;
109 if (res == -EAGAIN) {
125 snd_pcm_delay(s->
h, &delay);
156 .priv_class = &alsa_demuxer_class,
void av_free_packet(AVPacket *pkt)
Free a packet.
#define LIBAVUTIL_VERSION_INT
void ff_timefilter_reset(TimeFilter *self)
Reset the filter.
AVCodecContext * codec
Codec context associated with this stream.
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
const char * av_default_item_name(void *ctx)
Return the context name.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
static av_cold int read_close(AVFormatContext *ctx)
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
const OptionDef options[]
static av_cold int audio_read_header(AVFormatContext *s1)
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
AVCodecID
Identify the syntax and semantics of the bitstream.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void * priv_data
Format private data.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int64_t av_gettime(void)
Get the current time in microseconds.
int64_t av_rescale(int64_t a, int64_t b, int64_t c) av_const
Rescale a 64-bit integer with rounding to nearest.
enum AVCodecID audio_codec_id
Forced audio codec_id.
int channels
number of channels set by user
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.
static int read_header(FFV1Context *f)
AVInputFormat ff_alsa_demuxer
TimeFilter * ff_timefilter_new(double time_base, double period, double bandwidth)
Create a new Delay Locked Loop time filter.
enum AVMediaType codec_type
int sample_rate
samples per second
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
Main libavdevice API header.
static int read_packet(AVFormatContext *ctx, AVPacket *pkt)
static const AVClass alsa_demuxer_class
Describe the class of an AVClass context structure.
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
int period_size
preferred size for reads and writes, in frames
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
double ff_timefilter_update(TimeFilter *self, double system_time, double period)
Update the filter.
int channels
number of audio channels
int frame_size
bytes per sample * channels
int sample_rate
sample rate set by user
ALSA input and output: definitions and structures.
This structure stores compressed data.
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...