FFmpeg
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 "encode.h"
27 #include "internal.h"
28 #include "mathops.h"
29 
30 #define ROQ_FRAME_SIZE 735
31 #define ROQ_HEADER_SIZE 8
32 
33 #define MAX_DPCM (127*127)
34 
35 
36 typedef struct ROQDPCMContext {
37  short lastSample[2];
40  int16_t *frame_buffer;
41  int64_t first_pts;
43 
44 
46 {
48 
49  av_freep(&context->frame_buffer);
50 
51  return 0;
52 }
53 
55 {
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  return AVERROR(ENOMEM);
75 
76  context->lastSample[0] = context->lastSample[1] = 0;
77 
78  return 0;
79 }
80 
81 static unsigned char dpcm_predict(short *previous, short current)
82 {
83  int diff;
84  int negative;
85  int result;
86  int predicted;
87 
88  diff = current - *previous;
89 
90  negative = diff<0;
91  diff = FFABS(diff);
92 
93  if (diff >= MAX_DPCM)
94  result = 127;
95  else {
96  result = ff_sqrt(diff);
98  }
99 
100  /* See if this overflows */
101  retry:
102  diff = result*result;
103  if (negative)
104  diff = -diff;
105  predicted = *previous + diff;
106 
107  /* If it overflows, back off a step */
108  if (predicted > 32767 || predicted < -32768) {
109  result--;
110  goto retry;
111  }
112 
113  /* Add the sign bit */
114  result |= negative << 7; //if (negative) result |= 128;
115 
116  *previous = predicted;
117 
118  return result;
119 }
120 
122  const AVFrame *frame, int *got_packet_ptr)
123 {
124  int i, stereo, data_size, ret;
125  const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
126  uint8_t *out;
127  ROQDPCMContext *context = avctx->priv_data;
128 
129  stereo = (avctx->channels == 2);
130 
131  if (!in && context->input_frames >= 8)
132  return 0;
133 
134  if (in && context->input_frames < 8) {
135  memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
136  in, avctx->frame_size * avctx->channels * sizeof(*in));
137  context->buffered_samples += avctx->frame_size;
138  if (context->input_frames == 0)
139  context->first_pts = frame->pts;
140  if (context->input_frames < 7) {
141  context->input_frames++;
142  return 0;
143  }
144  }
145  if (context->input_frames < 8)
146  in = context->frame_buffer;
147 
148  if (stereo) {
149  context->lastSample[0] &= 0xFF00;
150  context->lastSample[1] &= 0xFF00;
151  }
152 
153  if (context->input_frames == 7)
154  data_size = avctx->channels * context->buffered_samples;
155  else
156  data_size = avctx->channels * avctx->frame_size;
157 
158  ret = ff_get_encode_buffer(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0);
159  if (ret < 0)
160  return ret;
161  out = avpkt->data;
162 
163  bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
164  bytestream_put_byte(&out, 0x10);
165  bytestream_put_le32(&out, data_size);
166 
167  if (stereo) {
168  bytestream_put_byte(&out, (context->lastSample[1])>>8);
169  bytestream_put_byte(&out, (context->lastSample[0])>>8);
170  } else
171  bytestream_put_le16(&out, context->lastSample[0]);
172 
173  /* Write the actual samples */
174  for (i = 0; i < data_size; i++)
175  *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
176 
177  avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
178  avpkt->duration = data_size / avctx->channels;
179 
180  context->input_frames++;
181  if (!in)
182  context->input_frames = FFMAX(context->input_frames, 8);
183 
184  *got_packet_ptr = 1;
185  return 0;
186 }
187 
189  .name = "roq_dpcm",
190  .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
191  .type = AVMEDIA_TYPE_AUDIO,
192  .id = AV_CODEC_ID_ROQ_DPCM,
193  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
194  .priv_data_size = sizeof(ROQDPCMContext),
196  .encode2 = roq_dpcm_encode_frame,
197  .close = roq_dpcm_encode_close,
198  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
200  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
201 };
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1118
AVCodec
AVCodec.
Definition: codec.h:197
FF_CODEC_CAP_INIT_THREADSAFE
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:41
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
AVERROR
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
out
FILE * out
Definition: movenc.c:54
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1098
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:953
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:365
encode.h
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:383
ROQDPCMContext
Definition: roqaudioenc.c:36
ff_roq_dpcm_encoder
const AVCodec ff_roq_dpcm_encoder
Definition: roqaudioenc.c:188
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
ff_sqrt
#define ff_sqrt
Definition: mathops.h:206
ROQDPCMContext::frame_buffer
int16_t * frame_buffer
Definition: roqaudioenc.c:40
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:181
av_cold
#define av_cold
Definition: attributes.h:90
ROQ_HEADER_SIZE
#define ROQ_HEADER_SIZE
Definition: roqaudioenc.c:31
ROQDPCMContext::input_frames
int input_frames
Definition: roqaudioenc.c:38
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
context
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 default minimum maximum flags name is the option keep it simple and lowercase description are in without and describe what they for example set the foo of the bar offset is the offset of the field in your context
Definition: writing_filters.txt:91
result
and forward the result(frame or status change) to the corresponding input. If nothing is possible
NULL
#define NULL
Definition: coverity.c:32
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:551
mathops.h
ROQDPCMContext::first_pts
int64_t first_pts
Definition: roqaudioenc.c:41
AV_CODEC_ID_ROQ_DPCM
@ AV_CODEC_ID_ROQ_DPCM
Definition: codec_id.h:412
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
roq_dpcm_encode_init
static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
Definition: roqaudioenc.c:54
FFMAX
#define FFMAX(a, b)
Definition: common.h:103
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
roq_dpcm_encode_frame
static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: roqaudioenc.c:121
dpcm_predict
static unsigned char dpcm_predict(short *previous, short current)
Definition: roqaudioenc.c:81
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1099
i
int i
Definition: input.c:407
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:358
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:204
avcodec.h
ret
ret
Definition: filter_design.txt:187
frame
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
Definition: filter_design.txt:264
MAX_DPCM
#define MAX_DPCM
Definition: roqaudioenc.c:33
AVCodecContext
main external API structure.
Definition: avcodec.h:501
ROQDPCMContext::buffered_samples
int buffered_samples
Definition: roqaudioenc.c:39
ff_get_encode_buffer
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Definition: encode.c:81
AV_CODEC_CAP_DELAY
#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
roq_dpcm_encode_close
static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
Definition: roqaudioenc.c:45
diff
static av_always_inline int diff(const uint32_t a, const uint32_t b)
Definition: vf_palettegen.c:136
AVPacket
This structure stores compressed data.
Definition: packet.h:342
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:528
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
bytestream.h
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
ROQ_FRAME_SIZE
#define ROQ_FRAME_SIZE
Definition: roqaudioenc.c:30
ROQDPCMContext::lastSample
short lastSample[2]
Definition: roqaudioenc.c:37