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roqaudioenc.c
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1 /*
2  * RoQ audio encoder
3  *
4  * Copyright (c) 2005 Eric Lasota
5  * Based on RoQ specs (c)2001 Tim Ferguson
6  *
7  * This file is part of FFmpeg.
8  *
9  * FFmpeg is free software; you can redistribute it and/or
10  * modify it under the terms of the GNU Lesser General Public
11  * License as published by the Free Software Foundation; either
12  * version 2.1 of the License, or (at your option) any later version.
13  *
14  * FFmpeg is distributed in the hope that it will be useful,
15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17  * Lesser General Public License for more details.
18  *
19  * You should have received a copy of the GNU Lesser General Public
20  * License along with FFmpeg; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22  */
23 
24 #include "avcodec.h"
25 #include "bytestream.h"
26 #include "internal.h"
27 #include "mathops.h"
28 
29 #define ROQ_FRAME_SIZE 735
30 #define ROQ_HEADER_SIZE 8
31 
32 #define MAX_DPCM (127*127)
33 
34 
35 typedef struct ROQDPCMContext {
36  short lastSample[2];
39  int16_t *frame_buffer;
40  int64_t first_pts;
42 
43 
45 {
46  ROQDPCMContext *context = avctx->priv_data;
47 
48  av_freep(&context->frame_buffer);
49 
50  return 0;
51 }
52 
54 {
55  ROQDPCMContext *context = avctx->priv_data;
56  int ret;
57 
58  if (avctx->channels > 2) {
59  av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
60  return AVERROR(EINVAL);
61  }
62  if (avctx->sample_rate != 22050) {
63  av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
64  return AVERROR(EINVAL);
65  }
66 
67  avctx->frame_size = ROQ_FRAME_SIZE;
68  avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
69  (22050 / ROQ_FRAME_SIZE) * 8;
70 
71  context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
72  sizeof(*context->frame_buffer));
73  if (!context->frame_buffer) {
74  ret = AVERROR(ENOMEM);
75  goto error;
76  }
77 
78  context->lastSample[0] = context->lastSample[1] = 0;
79 
80  return 0;
81 error:
82  roq_dpcm_encode_close(avctx);
83  return ret;
84 }
85 
86 static unsigned char dpcm_predict(short *previous, short current)
87 {
88  int diff;
89  int negative;
90  int result;
91  int predicted;
92 
93  diff = current - *previous;
94 
95  negative = diff<0;
96  diff = FFABS(diff);
97 
98  if (diff >= MAX_DPCM)
99  result = 127;
100  else {
101  result = ff_sqrt(diff);
102  result += diff > result*result+result;
103  }
104 
105  /* See if this overflows */
106  retry:
107  diff = result*result;
108  if (negative)
109  diff = -diff;
110  predicted = *previous + diff;
111 
112  /* If it overflows, back off a step */
113  if (predicted > 32767 || predicted < -32768) {
114  result--;
115  goto retry;
116  }
117 
118  /* Add the sign bit */
119  result |= negative << 7; //if (negative) result |= 128;
120 
121  *previous = predicted;
122 
123  return result;
124 }
125 
127  const AVFrame *frame, int *got_packet_ptr)
128 {
129  int i, stereo, data_size, ret;
130  const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
131  uint8_t *out;
132  ROQDPCMContext *context = avctx->priv_data;
133 
134  stereo = (avctx->channels == 2);
135 
136  if (!in && context->input_frames >= 8)
137  return 0;
138 
139  if (in && context->input_frames < 8) {
140  memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
141  in, avctx->frame_size * avctx->channels * sizeof(*in));
142  context->buffered_samples += avctx->frame_size;
143  if (context->input_frames == 0)
144  context->first_pts = frame->pts;
145  if (context->input_frames < 7) {
146  context->input_frames++;
147  return 0;
148  }
149  }
150  if (context->input_frames < 8)
151  in = context->frame_buffer;
152 
153  if (stereo) {
154  context->lastSample[0] &= 0xFF00;
155  context->lastSample[1] &= 0xFF00;
156  }
157 
158  if (context->input_frames == 7)
159  data_size = avctx->channels * context->buffered_samples;
160  else
161  data_size = avctx->channels * avctx->frame_size;
162 
163  if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0)) < 0)
164  return ret;
165  out = avpkt->data;
166 
167  bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
168  bytestream_put_byte(&out, 0x10);
169  bytestream_put_le32(&out, data_size);
170 
171  if (stereo) {
172  bytestream_put_byte(&out, (context->lastSample[1])>>8);
173  bytestream_put_byte(&out, (context->lastSample[0])>>8);
174  } else
175  bytestream_put_le16(&out, context->lastSample[0]);
176 
177  /* Write the actual samples */
178  for (i = 0; i < data_size; i++)
179  *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
180 
181  avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
182  avpkt->duration = data_size / avctx->channels;
183 
184  context->input_frames++;
185  if (!in)
186  context->input_frames = FFMAX(context->input_frames, 8);
187 
188  *got_packet_ptr = 1;
189  return 0;
190 }
191 
193  .name = "roq_dpcm",
194  .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
195  .type = AVMEDIA_TYPE_AUDIO,
196  .id = AV_CODEC_ID_ROQ_DPCM,
197  .priv_data_size = sizeof(ROQDPCMContext),
199  .encode2 = roq_dpcm_encode_frame,
200  .close = roq_dpcm_encode_close,
201  .capabilities = AV_CODEC_CAP_DELAY,
202  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
204 };
#define NULL
Definition: coverity.c:32
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1740
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AVCodec.
Definition: avcodec.h:3599
short lastSample[2]
Definition: roqaudioenc.c:36
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:983
int64_t first_pts
Definition: roqaudioenc.c:40
uint8_t
#define av_cold
Definition: attributes.h:82
static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
Definition: roqaudioenc.c:44
#define av_malloc(s)
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1618
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:268
int buffered_samples
Definition: roqaudioenc.c:38
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1600
#define av_log(a,...)
#define ff_sqrt
Definition: mathops.h:215
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
const char * name
Name of the codec implementation.
Definition: avcodec.h:3606
#define FFMAX(a, b)
Definition: common.h:94
int16_t * frame_buffer
Definition: roqaudioenc.c:39
static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: roqaudioenc.c:126
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
Definition: roqaudioenc.c:53
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2457
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int sample_rate
samples per second
Definition: avcodec.h:2437
main external API structure.
Definition: avcodec.h:1675
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
static unsigned char dpcm_predict(short *previous, short current)
Definition: roqaudioenc.c:86
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1710
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:198
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
void * priv_data
Definition: avcodec.h:1717
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
Definition: avcodec.h:2438
#define ROQ_FRAME_SIZE
Definition: roqaudioenc.c:29
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
AVCodec ff_roq_dpcm_encoder
Definition: roqaudioenc.c:192
#define av_freep(p)
#define ROQ_HEADER_SIZE
Definition: roqaudioenc.c:30
#define MAX_DPCM
Definition: roqaudioenc.c:32
This structure stores compressed data.
Definition: avcodec.h:1577
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1593