FFmpeg: libavcodec/mpc7.c Source File
Go to the documentation of this file. 1 /*
2 * Musepack SV7 decoder
3 * Copyright (c) 2006 Konstantin Shishkov
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
25 * divided into 32 subbands.
26 */
27
33
39
42
44
46 {
49
60 for (int j = 0; j < 2; j++) {
64 &raw_quant_table[1], 2,
65 &raw_quant_table[0], 2, 1,
70 }
71 }
73 }
74
76 {
81
82 /* Musepack SV7 is always stereo */
86 }
87
91 }
92 memset(
c->oldDSCF, 0,
sizeof(
c->oldDSCF));
96 c->bdsp.bswap_buf((uint32_t *) buf, (
const uint32_t *) avctx->
extradata, 4);
98
105 }
110 c->IS,
c->MSS,
c->gapless,
c->lastframelen,
c->maxbands);
111 c->frames_to_skip = 0;
112
116
118
119 return 0;
120 }
121
122 /**
123 * Fill samples for given subband
124 */
126 {
128 switch(idx){
129 case -1:
132 }
133 break;
134 case 1:
141 }
142 break;
143 case 2:
149 }
150 break;
151 case 3: case 4: case 5: case 6: case 7:
155 break;
156 case 8: case 9: case 10: case 11: case 12:
157 case 13: case 14: case 15: case 16: case 17:
158 t = (1 << (idx - 2)) - 1;
161 break;
162 default: // case 0 and -2..-17
163 return;
164 }
165 }
166
168 {
170 if (t == 8)
173 }
174
176 int *got_frame_ptr,
AVPacket *avpkt)
177 {
178 const uint8_t *buf = avpkt->
data;
179 int buf_size;
185 int off,
ret, last_frame, skip;
186 int bits_used, bits_avail;
187
188 memset(
bands, 0,
sizeof(*
bands) * (
c->maxbands + 1));
189
190 buf_size = avpkt->
size & ~3;
191 if (buf_size <= 0) {
195 }
196 if (buf_size != avpkt->
size) {
198 "extra bytes at the end will be skipped.\n");
199 }
200
201 skip = buf[0];
202 last_frame = buf[1];
203 buf += 4;
204 buf_size -= 4;
205
206 /* get output buffer */
210
214 c->bdsp.bswap_buf((uint32_t *)
c->bits, (
const uint32_t *) buf,
215 buf_size >> 2);
219
220 /* read subband indexes */
221 for(
i = 0;
i <=
c->maxbands;
i++){
222 for(ch = 0; ch < 2; ch++){
229 }
230 }
231
235 }
236 }
237 /* get scale indexes coding method */
238 for(
i = 0;
i <=
mb;
i++)
239 for(ch = 0; ch < 2; ch++)
241 /* get scale indexes */
242 for(
i = 0;
i <=
mb;
i++){
243 for(ch = 0; ch < 2; ch++){
245 bands[
i].scf_idx[ch][2] =
c->oldDSCF[ch][
i];
247 switch(
bands[
i].scfi[ch]){
248 case 0:
251 break;
252 case 1:
255 break;
256 case 2:
259 break;
260 case 3:
262 break;
263 }
264 c->oldDSCF[ch][
i] =
bands[
i].scf_idx[ch][2];
265 }
266 }
267 }
268 /* get quantizers */
269 memset(
c->Q, 0,
sizeof(
c->Q));
270 off = 0;
272 for(ch = 0; ch < 2; ch++)
274
276 if(last_frame)
277 frame->nb_samples =
c->lastframelen;
278
280 bits_avail = buf_size * 8;
281 if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
282 av_log(avctx,
AV_LOG_ERROR,
"Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
284 }
285 if(
c->frames_to_skip){
287 *got_frame_ptr = 0;
289 }
290
291 *got_frame_ptr = 1;
292
294 }
295
297 {
299
300 memset(
c->oldDSCF, 0,
sizeof(
c->oldDSCF));
301 c->frames_to_skip = 32;
302 }
303
305 {
309 return 0;
310 }
311
326 };
static const uint8_t mpc7_scfi[MPC7_SCFI_SIZE *2]
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static int mpc7_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
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
const FFCodec ff_mpc7_decoder
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
static int get_bits_count(const GetBitContext *s)
av_cold void ff_mpadsp_init(MPADSPContext *s)
This structure describes decoded (raw) audio or video data.
static VLC quant_vlc[MPC7_QUANT_VLC_TABLES][2]
static const uint16_t table[]
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
#define AV_CHANNEL_LAYOUT_STEREO
int nb_channels
Number of channels in this layout.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static const uint8_t mpc7_hdr[MPC7_HDR_SIZE *2]
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
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)
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
#define FF_CODEC_DECODE_CB(func)
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
static const int8_t mpc7_idx32[]
#define LOCAL_ALIGNED_16(t, v,...)
static void mpc7_decode_flush(AVCodecContext *avctx)
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static const float bands[]
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
void ff_mpa_synth_init_fixed(void)
static const int8_t mpc7_idx51[]
static const uint8_t mpc7_quant_vlcs[177 *2 *2]
#define INIT_VLC_STATIC_FROM_LENGTHS(vlc, bits, nb_codes, lens, len_wrap, symbols, symbols_wrap, symbols_size, offset, flags, static_size)
static unsigned int get_bits1(GetBitContext *s)
int ff_init_vlc_from_lengths(VLC *vlc, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags, void *logctx)
Build VLC decoding tables suitable for use with get_vlc2()
static av_cold void mpc7_init_static(void)
static av_always_inline int get_vlc2(GetBitContext *s, const VLCElem *table, int bits, int max_depth)
Parse a vlc code.
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
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
#define MPC7_QUANT_VLC_TABLES
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
An AVChannelLayout holds information about the channel layout of audio data.
enum AVSampleFormat sample_fmt
audio sample format
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES]
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
static int get_scale_idx(GetBitContext *gb, int ref)
#define i(width, name, range_min, range_max)
static const int8_t mpc7_idx31[]
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
AVSampleFormat
Audio sample formats.
void av_fast_padded_malloc(void *ptr, unsigned int *size, size_t min_size)
Same behaviour av_fast_malloc but the buffer has additional AV_INPUT_BUFFER_PADDING_SIZE at the end w...
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
const char * name
Name of the codec implementation.
static const int8_t mpc7_idx30[]
#define INIT_VLC_STATIC_OVERLONG
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
static const uint8_t mpc7_dscf[MPC7_DSCF_SIZE *2]
main external API structure.
static const int8_t mpc7_idx50[]
Subband structure - hold all variables for each subband.
static int ref[MAX_W *MAX_W]
#define avpriv_request_sample(...)
This structure stores compressed data.
static const uint8_t quant_tables[]
static void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
Fill samples for given subband.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static const int8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
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