42 const int *samples_per_frame,
47 if (!samples_per_frame)
74 int stream_index,
int flush)
const int * samples
current samples per frame, pointer to samples_per_frame
const int * samples_per_frame
must be 0-terminated
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=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);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_AARCH64) ff_audio_convert_init_aarch64(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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
int av_fifo_generic_write(AVFifoBuffer *f, void *src, int size, int(*func)(void *, void *, int))
Feed data from a user-supplied callback to an AVFifoBuffer.
int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base)
AVStream ** streams
A list of all streams in the file.
unsigned fifo_size
size of currently allocated FIFO
void av_fifo_free(AVFifoBuffer *f)
Free an AVFifoBuffer.
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, int(*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), int(*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame and interleave them correctly...
int av_fifo_generic_read(AVFifoBuffer *f, void *dest, int buf_size, void(*func)(void *, void *, int))
Feed data from an AVFifoBuffer to a user-supplied callback.
void av_log(void *avcl, int level, const char *fmt,...)
AVCodecContext * codec
Codec context associated with this stream.
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
AVRational time_base
time base of output audio packets
enum AVMediaType codec_type
int av_fifo_realloc2(AVFifoBuffer *f, unsigned int new_size)
Resize an AVFifoBuffer.
a very simple circular buffer FIFO implementation
rational number numerator/denominator
int sample_size
size of one sample all channels included
int av_fifo_size(AVFifoBuffer *f)
Return the amount of data in bytes in the AVFifoBuffer, that is the amount of data you can read from ...
void ff_audio_interleave_close(AVFormatContext *s)
static av_cold void flush(AVCodecContext *avctx)
Flush (reset) the frame ID after seeking.
static int get_packet(URLContext *s, int for_header)
Interact with the server by receiving and sending RTMP packets until there is some significant data (...
AVFifoBuffer * av_fifo_alloc(unsigned int size)
Initialize an AVFifoBuffer.
static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, int stream_index, int flush)
int channels
number of audio channels
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
This structure stores compressed data.
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...