FFmpeg  4.4.5
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;
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 
57  if (avctx->channels > 2) {
58  av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
59  return AVERROR(EINVAL);
60  }
61  if (avctx->sample_rate != 22050) {
62  av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
63  return AVERROR(EINVAL);
64  }
65 
66  avctx->frame_size = ROQ_FRAME_SIZE;
67  avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
68  (22050 / ROQ_FRAME_SIZE) * 8;
69 
70  context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
71  sizeof(*context->frame_buffer));
72  if (!context->frame_buffer)
73  return AVERROR(ENOMEM);
74 
75  context->lastSample[0] = context->lastSample[1] = 0;
76 
77  return 0;
78 }
79 
80 static unsigned char dpcm_predict(short *previous, short current)
81 {
82  int diff;
83  int negative;
84  int result;
85  int predicted;
86 
87  diff = current - *previous;
88 
89  negative = diff<0;
90  diff = FFABS(diff);
91 
92  if (diff >= MAX_DPCM)
93  result = 127;
94  else {
95  result = ff_sqrt(diff);
96  result += diff > result*result+result;
97  }
98 
99  /* See if this overflows */
100  retry:
101  diff = result*result;
102  if (negative)
103  diff = -diff;
104  predicted = *previous + diff;
105 
106  /* If it overflows, back off a step */
107  if (predicted > 32767 || predicted < -32768) {
108  result--;
109  goto retry;
110  }
111 
112  /* Add the sign bit */
113  result |= negative << 7; //if (negative) result |= 128;
114 
115  *previous = predicted;
116 
117  return result;
118 }
119 
121  const AVFrame *frame, int *got_packet_ptr)
122 {
123  int i, stereo, data_size, ret;
124  const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
125  uint8_t *out;
126  ROQDPCMContext *context = avctx->priv_data;
127 
128  stereo = (avctx->channels == 2);
129 
130  if (!in && context->input_frames >= 8)
131  return 0;
132 
133  if (in && context->input_frames < 8) {
134  memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
135  in, avctx->frame_size * avctx->channels * sizeof(*in));
136  context->buffered_samples += avctx->frame_size;
137  if (context->input_frames == 0)
138  context->first_pts = frame->pts;
139  if (context->input_frames < 7) {
140  context->input_frames++;
141  return 0;
142  }
143  }
144  if (context->input_frames < 8)
145  in = context->frame_buffer;
146 
147  if (stereo) {
148  context->lastSample[0] &= 0xFF00;
149  context->lastSample[1] &= 0xFF00;
150  }
151 
152  if (context->input_frames == 7)
153  data_size = avctx->channels * context->buffered_samples;
154  else
155  data_size = avctx->channels * avctx->frame_size;
156 
157  if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0)) < 0)
158  return ret;
159  out = avpkt->data;
160 
161  bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
162  bytestream_put_byte(&out, 0x10);
163  bytestream_put_le32(&out, data_size);
164 
165  if (stereo) {
166  bytestream_put_byte(&out, (context->lastSample[1])>>8);
167  bytestream_put_byte(&out, (context->lastSample[0])>>8);
168  } else
169  bytestream_put_le16(&out, context->lastSample[0]);
170 
171  /* Write the actual samples */
172  for (i = 0; i < data_size; i++)
173  *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
174 
175  avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
176  avpkt->duration = data_size / avctx->channels;
177 
178  context->input_frames++;
179  if (!in)
180  context->input_frames = FFMAX(context->input_frames, 8);
181 
182  *got_packet_ptr = 1;
183  return 0;
184 }
185 
187  .name = "roq_dpcm",
188  .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
189  .type = AVMEDIA_TYPE_AUDIO,
190  .id = AV_CODEC_ID_ROQ_DPCM,
191  .priv_data_size = sizeof(ROQDPCMContext),
193  .encode2 = roq_dpcm_encode_frame,
194  .close = roq_dpcm_encode_close,
195  .capabilities = AV_CODEC_CAP_DELAY,
196  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
198 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
#define av_cold
Definition: attributes.h:88
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_log(ac->avr, AV_LOG_TRACE, "%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-> in
uint8_t
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
#define FFMAX(a, b)
Definition: common.h:103
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define NULL
Definition: coverity.c:32
long long int64_t
Definition: coverity.c:34
static AVFrame * frame
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:33
#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: codec.h:77
@ AV_CODEC_ID_ROQ_DPCM
Definition: codec_id.h:414
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
int i
Definition: input.c:407
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define ff_sqrt
Definition: mathops.h:206
static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
Definition: roqaudioenc.c:44
static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
Definition: roqaudioenc.c:53
#define MAX_DPCM
Definition: roqaudioenc.c:32
#define ROQ_HEADER_SIZE
Definition: roqaudioenc.c:30
static unsigned char dpcm_predict(short *previous, short current)
Definition: roqaudioenc.c:80
#define ROQ_FRAME_SIZE
Definition: roqaudioenc.c:29
AVCodec ff_roq_dpcm_encoder
Definition: roqaudioenc.c:186
static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: roqaudioenc.c:120
main external API structure.
Definition: avcodec.h:536
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
int sample_rate
samples per second
Definition: avcodec.h:1196
int channels
number of audio channels
Definition: avcodec.h:1197
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:411
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
This structure stores compressed data.
Definition: packet.h:346
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:387
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:362
uint8_t * data
Definition: packet.h:369
short lastSample[2]
Definition: roqaudioenc.c:36
int buffered_samples
Definition: roqaudioenc.c:38
int64_t first_pts
Definition: roqaudioenc.c:40
int16_t * frame_buffer
Definition: roqaudioenc.c:39
#define av_freep(p)
#define av_malloc(s)
#define av_log(a,...)
FILE * out
Definition: movenc.c:54
static av_always_inline int diff(const uint32_t a, const uint32_t b)