Go to the documentation of this file.
28 #include "config_components.h"
43 #if !CONFIG_HARDCODED_TABLES
45 #if CONFIG_PCM_ALAW_ENCODER
52 #if CONFIG_PCM_MULAW_ENCODER
59 #if CONFIG_PCM_VIDC_ENCODER
88 #define ENCODE(type, endian, src, dst, n, shift, offset) \
89 samples_ ## type = (const type *) src; \
90 for (; n > 0; n--) { \
91 register type v = (*samples_ ## type++ >> shift) + offset; \
92 bytestream_put_ ## endian(&dst, v); \
95 #define ENCODE_PLANAR(type, endian, dst, n, shift, offset) \
96 n /= avctx->ch_layout.nb_channels; \
97 for (c = 0; c < avctx->ch_layout.nb_channels; c++) { \
99 samples_ ## type = (const type *) frame->extended_data[c]; \
100 for (i = n; i > 0; i--) { \
101 register type v = (*samples_ ## type++ >> shift) + offset; \
102 bytestream_put_ ## endian(&dst, v); \
109 int n,
c, sample_size,
ret;
112 const uint8_t *samples_uint8_t;
113 const int16_t *samples_int16_t;
114 const int32_t *samples_int32_t;
115 const int64_t *samples_int64_t;
116 const uint16_t *samples_uint16_t;
117 const uint32_t *samples_uint32_t;
154 bytestream_put_be24(&
dst,
tmp);
225 const uint8_t *
src =
frame->extended_data[
c];
229 #if CONFIG_PCM_ALAW_ENCODER
233 *
dst++ = linear_to_alaw[(v + 32768) >> 2];
237 #if CONFIG_PCM_MULAW_ENCODER
241 *
dst++ = linear_to_ulaw[(v + 32768) >> 2];
245 #if CONFIG_PCM_VIDC_ENCODER
249 *
dst++ = linear_to_vidc[(v + 32768) >> 2];
268 static const struct {
272 uint8_t bits_per_sample;
273 } codec_id_to_samplefmt[] = {
274 #define ENTRY(CODEC_ID, SAMPLE_FMT, BITS_PER_SAMPLE) \
275 { AV_CODEC_ID_PCM_ ## CODEC_ID, AV_SAMPLE_FMT_ ## SAMPLE_FMT, \
276 BITS_PER_SAMPLE / 8, BITS_PER_SAMPLE }
278 ENTRY(S16BE, S16, 16),
ENTRY(S16BE_PLANAR, S16P, 16),
279 ENTRY(S16LE, S16, 16),
ENTRY(S16LE_PLANAR, S16P, 16),
280 ENTRY(S24DAUD, S16, 24),
ENTRY(S24BE, S32, 24),
281 ENTRY(S24LE, S32, 24),
ENTRY(S24LE_PLANAR, S32P, 24),
283 ENTRY(S32LE_PLANAR, S32P, 32),
296 s->sample_size = codec_id_to_samplefmt[
i].sample_size;
297 avctx->
sample_fmt = codec_id_to_samplefmt[
i].sample_fmt;
321 s->base.sample_size = 4;
347 av_unreachable(
"pcm_lut_decode_init() only used with alaw, mulaw and vidc");
348 #if CONFIG_PCM_ALAW_DECODER
350 for (
int i = 0;
i < 256;
i++)
351 s->table[
i] = alaw2linear(
i);
354 #if CONFIG_PCM_MULAW_DECODER
356 for (
int i = 0;
i < 256;
i++)
357 s->table[
i] = ulaw2linear(
i);
360 #if CONFIG_PCM_VIDC_DECODER
362 for (
int i = 0;
i < 256;
i++)
363 s->table[
i] = vidc2linear(
i);
369 s->base.sample_size = 1;
384 #define DECODE(size, endian, src, dst, n, shift, offset) \
385 for (; n > 0; n--) { \
386 uint ## size ## _t v = bytestream_get_ ## endian(&src); \
387 AV_WN ## size ## A(dst, (uint ## size ## _t)(v - offset) << shift); \
391 #define DECODE_PLANAR(size, endian, src, dst, n, shift, offset) \
393 for (c = 0; c < avctx->ch_layout.nb_channels; c++) { \
395 dst = frame->extended_data[c]; \
396 for (i = n; i > 0; i--) { \
397 uint ## size ## _t v = bytestream_get_ ## endian(&src); \
398 AV_WN ## size ## A(dst, (uint ## size ##_t)(v - offset) << shift); \
404 int *got_frame_ptr,
AVPacket *avpkt)
406 const uint8_t *
src = avpkt->
data;
407 int buf_size = avpkt->
size;
410 int sample_size =
s->sample_size;
411 int c, n,
ret, samples_per_block;
415 samples_per_block = 1;
418 samples_per_block = 2;
433 if (n && buf_size % n) {
436 "Invalid PCM packet, data has size %d but at least a size of %d was expected\n",
440 buf_size -= buf_size % n;
443 n = buf_size / sample_size;
475 uint32_t v = bytestream_get_be24(&
src);
494 int sign = *
src >> 7;
495 int magn = *
src & 0x7f;
496 *
samples++ = sign ? 128 - magn : 128 + magn;
505 for (
i = n;
i > 0;
i--)
572 #if CONFIG_PCM_ALAW_DECODER || CONFIG_PCM_MULAW_DECODER || \
573 CONFIG_PCM_VIDC_DECODER
578 int16_t *restrict samples_16 = (int16_t*)
samples;
581 *samples_16++ = lut[*
src++];
593 *dst_int32_t++ = ((uint32_t)
src[2]<<28) |
596 ((
src[2] & 0x0F) << 8) |
599 *dst_int32_t++ = ((uint32_t)
src[4]<<24) |
601 ((
src[2] & 0xF0) << 8) |
617 (
const float *)
frame->extended_data[0],
626 #define PCM_ENCODER_0(id_, sample_fmt_, name_, long_name_)
627 #define PCM_ENCODER_1(id_, sample_fmt_, name_, long_name_) \
628 const FFCodec ff_ ## name_ ## _encoder = { \
630 CODEC_LONG_NAME(long_name_), \
631 .p.type = AVMEDIA_TYPE_AUDIO, \
633 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_VARIABLE_FRAME_SIZE | \
634 AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE, \
635 .init = pcm_encode_init, \
636 FF_CODEC_ENCODE_CB(pcm_encode_frame), \
637 CODEC_SAMPLEFMTS(sample_fmt_), \
640 #define PCM_ENCODER_2(cf, id, sample_fmt, name, long_name) \
641 PCM_ENCODER_ ## cf(id, sample_fmt, name, long_name)
642 #define PCM_ENCODER_3(cf, id, sample_fmt, name, long_name) \
643 PCM_ENCODER_2(cf, id, sample_fmt, name, long_name)
644 #define PCM_ENCODER(id, sample_fmt, name, long_name) \
645 PCM_ENCODER_3(CONFIG_PCM_ ## id ## _ENCODER, AV_CODEC_ID_PCM_ ## id, \
646 AV_SAMPLE_FMT_ ## sample_fmt, pcm_ ## name, long_name)
648 #define PCM_DECODER_0(id, sample_fmt, name, long_name, Context, init_func)
649 #define PCM_DECODER_1(id_, sample_fmt, name_, long_name, Context, init_func)\
650 const FFCodec ff_ ## name_ ## _decoder = { \
652 CODEC_LONG_NAME(long_name), \
653 .p.type = AVMEDIA_TYPE_AUDIO, \
655 .priv_data_size = sizeof(Context), \
657 FF_CODEC_DECODE_CB(pcm_decode_frame), \
658 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_PARAM_CHANGE, \
661 #define PCM_DECODER_2(cf, id, sample_fmt, name, long_name, Context, init_func) \
662 PCM_DECODER_ ## cf(id, sample_fmt, name, long_name, Context, init_func)
663 #define PCM_DECODER_3(cf, id, sample_fmt, name, long_name, Context, init_func) \
664 PCM_DECODER_2(cf, id, sample_fmt, name, long_name, Context, init_func)
665 #define PCM_DEC_EXT(id, sample_fmt, name, long_name, Context, init_func) \
666 PCM_DECODER_3(CONFIG_PCM_ ## id ## _DECODER, AV_CODEC_ID_PCM_ ## id, \
667 AV_SAMPLE_FMT_ ## sample_fmt, pcm_ ## name, long_name, \
670 #define PCM_DECODER(id, sample_fmt, name, long_name) \
671 PCM_DEC_EXT(id, sample_fmt, name, long_name, PCMDecode, pcm_decode_init)
673 #define PCM_CODEC(id, sample_fmt_, name, long_name_) \
674 PCM_ENCODER(id, sample_fmt_, name, long_name_); \
675 PCM_DECODER(id, sample_fmt_, name, long_name_)
677 #define PCM_CODEC_EXT(id, sample_fmt, name, long_name, DecContext, dec_init_func) \
678 PCM_DEC_EXT(id, sample_fmt, name, long_name, DecContext, dec_init_func); \
679 PCM_ENCODER(id, sample_fmt, name, long_name)
688 PCM_CODEC (F32BE, FLT, f32be,
"PCM 32-bit floating point big-endian");
689 PCM_CODEC (F32LE, FLT, f32le,
"PCM 32-bit floating point little-endian");
690 PCM_CODEC (F64BE, DBL, f64be,
"PCM 64-bit floating point big-endian");
691 PCM_CODEC (F64LE, DBL, f64le,
"PCM 64-bit floating point little-endian");
692 PCM_DECODER (LXF, S32P,lxf,
"PCM signed 20-bit little-endian planar");
695 PCM_CODEC (S8_PLANAR, U8P, s8_planar,
"PCM signed 8-bit planar");
696 PCM_CODEC (S16BE, S16, s16be,
"PCM signed 16-bit big-endian");
697 PCM_CODEC (S16BE_PLANAR, S16P,s16be_planar,
"PCM signed 16-bit big-endian planar");
698 PCM_CODEC (S16LE, S16, s16le,
"PCM signed 16-bit little-endian");
699 PCM_CODEC (S16LE_PLANAR, S16P,s16le_planar,
"PCM signed 16-bit little-endian planar");
700 PCM_CODEC (S24BE, S32, s24be,
"PCM signed 24-bit big-endian");
701 PCM_CODEC (S24DAUD, S16, s24daud,
"PCM D-Cinema audio signed 24-bit");
702 PCM_CODEC (S24LE, S32, s24le,
"PCM signed 24-bit little-endian");
703 PCM_CODEC (S24LE_PLANAR, S32P,s24le_planar,
"PCM signed 24-bit little-endian planar");
704 PCM_CODEC (S32BE, S32, s32be,
"PCM signed 32-bit big-endian");
705 PCM_CODEC (S32LE, S32, s32le,
"PCM signed 32-bit little-endian");
706 PCM_CODEC (S32LE_PLANAR, S32P,s32le_planar,
"PCM signed 32-bit little-endian planar");
708 PCM_CODEC (U16BE, S16, u16be,
"PCM unsigned 16-bit big-endian");
709 PCM_CODEC (U16LE, S16, u16le,
"PCM unsigned 16-bit little-endian");
710 PCM_CODEC (U24BE, S32, u24be,
"PCM unsigned 24-bit big-endian");
711 PCM_CODEC (U24LE, S32, u24le,
"PCM unsigned 24-bit little-endian");
712 PCM_CODEC (U32BE, S32, u32be,
"PCM unsigned 32-bit big-endian");
713 PCM_CODEC (U32LE, S32, u32le,
"PCM unsigned 32-bit little-endian");
714 PCM_CODEC (S64BE, S64, s64be,
"PCM signed 64-bit big-endian");
715 PCM_CODEC (S64LE, S64, s64le,
"PCM signed 64-bit little-endian");
#define PCM_CODEC_EXT(id, sample_fmt, name, long_name, DecContext, dec_init_func)
#define PCM_CODEC(id, sample_fmt_, name, long_name_)
uint64_t_TMPL AV_WL64 unsigned int_TMPL le32
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame 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. Frame references ownership and permissions
#define ENCODE(type, endian, src, dst, n, shift, offset)
Write PCM samples macro.
int sample_rate
samples per second
@ AV_CODEC_ID_PCM_S32LE_PLANAR
This structure describes decoded (raw) audio or video data.
av_unused static av_cold int pcm_scale_decode_init(AVCodecContext *avctx)
@ AV_CODEC_ID_PCM_S16BE_PLANAR
const uint8_t ff_reverse[256]
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
int nb_channels
Number of channels in this layout.
@ AV_CODEC_ID_PCM_S16LE_PLANAR
av_unused static av_cold int pcm_decode_init(AVCodecContext *avctx)
const struct AVCodec * codec
AVChannelLayout ch_layout
Audio channel layout.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL be24
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL le24
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
#define PCM_DECODER(id, sample_fmt, name, long_name)
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
static int ff_thread_once(char *control, void(*routine)(void))
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define FF_ARRAY_ELEMS(a)
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
static av_unused int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
av_unused static av_cold int pcm_lut_decode_init(AVCodecContext *avctx)
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
#define ENTRY(CODEC_ID, SAMPLE_FMT, BITS_PER_SAMPLE)
#define av_unreachable(msg)
Asserts that are used as compiler optimization hints depending upon ASSERT_LEVEL and NBDEBUG.
int64_t bit_rate
the average bitrate
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL be64
#define DECODE_PLANAR(size, endian, src, dst, n, shift, offset)
static int pcm_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL be32
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
@ AV_CODEC_ID_PCM_S24LE_PLANAR
AVCodecID
Identify the syntax and semantics of the bitstream.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
#define i(width, name, range_min, range_max)
enum AVSampleFormat sample_fmt
audio sample format
#define DECODE(size, endian, src, dst, n, shift, offset)
Read PCM samples macro.
#define ENCODE_PLANAR(type, endian, dst, n, shift, offset)
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
@ AV_SAMPLE_FMT_S16
signed 16 bits
static av_always_inline unsigned int bytestream_get_buffer(const uint8_t **b, uint8_t *dst, unsigned int size)
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL be16
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL le16
#define PCM_DEC_EXT(id, sample_fmt, name, long_name, Context, init_func)
main external API structure.
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Filter the word “frame” indicates either a video frame or a group of audio samples
av_unused static av_cold int pcm_encode_init(AVCodecContext *avctx)
@ AV_CODEC_ID_PCM_S24DAUD
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
@ AV_CODEC_ID_PCM_S8_PLANAR
static const uint32_t S8[256]
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
@ AV_SAMPLE_FMT_S32
signed 32 bits