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alsa-audio-common.c
Go to the documentation of this file.
319 av_log(s1, AV_LOG_ERROR, "cannot recover from underrun (snd_pcm_prepare failed: %s)\n", snd_strerror(err));
Definition: libavcodec/avcodec.h:312
Definition: libavcodec/avcodec.h:100
Definition: libavcodec/avcodec.h:303
Definition: libavcodec/avcodec.h:301
void ff_timefilter_destroy(TimeFilter *self)
Free all resources associated with the filter.
Definition: timefilter.c:58
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
void * av_realloc(void *ptr, size_t size)
Allocate or reallocate a block of memory.
Definition: mem.c:141
#define MAKE_REORDER_FUNCS(CHANNELS, LAYOUT, MAP)
Definition: alsa-audio-common.c:80
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
#define AVFMT_FLAG_NONBLOCK
Do not block when reading packets from input.
Definition: avformat.h:1024
Definition: libavcodec/avcodec.h:313
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
Definition: alsa-audio-common.c:299
int ff_alsa_extend_reorder_buf(AlsaData *s, int min_size)
Definition: alsa-audio-common.c:331
Main libavdevice API header.
Definition: libavcodec/avcodec.h:306
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: libavcodec/utils.c:2690
Definition: alsa-audio.h:48
simple assert() macros that are a bit more flexible than ISO C assert().
audio channel layout utility functions
Definition: libavcodec/avcodec.h:314
Definition: libavcodec/avcodec.h:302
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample format(the sample packing is implied by the sample format) and sample rate.The lists are not just lists
Definition: libavcodec/avcodec.h:321
static av_cold int find_reorder_func(AlsaData *s, int codec_id, uint64_t layout, int out)
Definition: alsa-audio-common.c:127
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.
Definition: alsa-audio-common.c:167
Definition: libavcodec/avcodec.h:300
Definition: libavcodec/avcodec.h:304
Definition: libavcodec/avcodec.h:305
Definition: libavcodec/avcodec.h:309
Definition: libavcodec/avcodec.h:319
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:73
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
Definition: alsa-audio-common.c:310
Definition: libavcodec/avcodec.h:311
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Buffer references ownership and permissions
Definition: libavcodec/avcodec.h:322
Definition: libavcodec/avcodec.h:320
Definition: libavcodec/avcodec.h:310
Definition: libavcodec/avcodec.h:299
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel layout
Definition: filter_design.txt:12
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
Definition: audio_convert.c:194
Definition: libavcodec/avcodec.h:307
ALSA input and output: definitions and structures.
static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
Definition: alsa-audio-common.c:38
Definition: libavcodec/avcodec.h:308
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
Definition: channel_layout.c:182
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