FFmpeg  4.3
libvo-amrwbenc.c
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1 /*
2  * AMR Audio encoder stub
3  * Copyright (c) 2003 The FFmpeg project
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 #include <vo-amrwbenc/enc_if.h>
23 #include <stdio.h>
24 #include <stdlib.h>
25 
26 #include "libavutil/avstring.h"
27 #include "libavutil/internal.h"
28 #include "libavutil/mem.h"
29 #include "libavutil/opt.h"
30 #include "avcodec.h"
31 #include "internal.h"
32 
33 #define MAX_PACKET_SIZE (1 + (477 + 7) / 8)
34 
35 typedef struct AMRWBContext {
37  void *state;
38  int mode;
40  int allow_dtx;
41 } AMRWBContext;
42 
43 static const AVOption options[] = {
44  { "dtx", "Allow DTX (generate comfort noise)", offsetof(AMRWBContext, allow_dtx), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
45  { NULL }
46 };
47 
48 static const AVClass amrwb_class = {
49  .class_name = "libvo_amrwbenc",
50  .item_name = av_default_item_name,
51  .option = options,
52  .version = LIBAVUTIL_VERSION_INT,
53 };
54 
55 static int get_wb_bitrate_mode(int bitrate, void *log_ctx)
56 {
57  /* make the correspondence between bitrate and mode */
58  static const int rates[] = { 6600, 8850, 12650, 14250, 15850, 18250,
59  19850, 23050, 23850 };
60  int i, best = -1, min_diff = 0;
61  char log_buf[200];
62 
63  for (i = 0; i < 9; i++) {
64  if (rates[i] == bitrate)
65  return i;
66  if (best < 0 || abs(rates[i] - bitrate) < min_diff) {
67  best = i;
68  min_diff = abs(rates[i] - bitrate);
69  }
70  }
71  /* no bitrate matching exactly, log a warning */
72  snprintf(log_buf, sizeof(log_buf), "bitrate not supported: use one of ");
73  for (i = 0; i < 9; i++)
74  av_strlcatf(log_buf, sizeof(log_buf), "%.2fk, ", rates[i] / 1000.f);
75  av_strlcatf(log_buf, sizeof(log_buf), "using %.2fk", rates[best] / 1000.f);
76  av_log(log_ctx, AV_LOG_WARNING, "%s\n", log_buf);
77 
78  return best;
79 }
80 
82 {
83  AMRWBContext *s = avctx->priv_data;
84 
85  if (avctx->sample_rate != 16000 && avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL) {
86  av_log(avctx, AV_LOG_ERROR, "Only 16000Hz sample rate supported\n");
87  return AVERROR(ENOSYS);
88  }
89 
90  if (avctx->channels != 1) {
91  av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
92  return AVERROR(ENOSYS);
93  }
94 
95  s->mode = get_wb_bitrate_mode(avctx->bit_rate, avctx);
96  s->last_bitrate = avctx->bit_rate;
97 
98  avctx->frame_size = 320;
99  avctx->initial_padding = 80;
100 
101  s->state = E_IF_init();
102 
103  return 0;
104 }
105 
107 {
108  AMRWBContext *s = avctx->priv_data;
109 
110  E_IF_exit(s->state);
111  return 0;
112 }
113 
114 static int amr_wb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
115  const AVFrame *frame, int *got_packet_ptr)
116 {
117  AMRWBContext *s = avctx->priv_data;
118  const int16_t *samples = (const int16_t *)frame->data[0];
119  int size, ret;
120 
121  if ((ret = ff_alloc_packet2(avctx, avpkt, MAX_PACKET_SIZE, 0)) < 0)
122  return ret;
123 
124  if (s->last_bitrate != avctx->bit_rate) {
125  s->mode = get_wb_bitrate_mode(avctx->bit_rate, avctx);
126  s->last_bitrate = avctx->bit_rate;
127  }
128  size = E_IF_encode(s->state, s->mode, samples, avpkt->data, s->allow_dtx);
129  if (size <= 0 || size > MAX_PACKET_SIZE) {
130  av_log(avctx, AV_LOG_ERROR, "Error encoding frame\n");
131  return AVERROR(EINVAL);
132  }
133 
134  if (frame->pts != AV_NOPTS_VALUE)
135  avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
136 
137  avpkt->size = size;
138  *got_packet_ptr = 1;
139  return 0;
140 }
141 
143  .name = "libvo_amrwbenc",
144  .long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AMR-WB "
145  "(Adaptive Multi-Rate Wide-Band)"),
146  .type = AVMEDIA_TYPE_AUDIO,
147  .id = AV_CODEC_ID_AMR_WB,
148  .priv_data_size = sizeof(AMRWBContext),
150  .encode2 = amr_wb_encode_frame,
151  .close = amr_wb_encode_close,
152  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
154  .priv_class = &amrwb_class,
155  .wrapper_name = "libvo_amrwbenc",
156 };
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1206
AVCodec
AVCodec.
Definition: codec.h:190
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
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
opt.h
amr_wb_encode_init
static av_cold int amr_wb_encode_init(AVCodecContext *avctx)
Definition: libvo-amrwbenc.c:81
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1186
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:716
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:355
AVOption
AVOption.
Definition: opt.h:246
AMRWBContext::mode
int mode
Definition: libvo-amrwbenc.c:38
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
av_strlcatf
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:101
FF_COMPLIANCE_UNOFFICIAL
#define FF_COMPLIANCE_UNOFFICIAL
Allow unofficial extensions.
Definition: avcodec.h:1593
AV_CODEC_ID_AMR_WB
@ AV_CODEC_ID_AMR_WB
Definition: codec_id.h:393
rates
static const int rates[]
Definition: avresample.c:176
get_wb_bitrate_mode
static int get_wb_bitrate_mode(int bitrate, void *log_ctx)
Definition: libvo-amrwbenc.c:55
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:2060
ff_samples_to_time_base
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
Definition: internal.h:256
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold
#define av_cold
Definition: attributes.h:90
s
#define s(width, name)
Definition: cbs_vp9.c:257
AV_OPT_FLAG_ENCODING_PARAM
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
Definition: opt.h:276
amr_wb_encode_close
static int amr_wb_encode_close(AVCodecContext *avctx)
Definition: libvo-amrwbenc.c:106
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AMRWBContext::state
void * state
Definition: libvo-amrwbenc.c:37
f
#define f(width, name)
Definition: cbs_vp9.c:255
AMRWBContext::av_class
AVClass * av_class
Definition: libvo-amrwbenc.c:36
AV_OPT_FLAG_AUDIO_PARAM
#define AV_OPT_FLAG_AUDIO_PARAM
Definition: opt.h:278
if
if(ret)
Definition: filter_design.txt:179
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
amrwb_class
static const AVClass amrwb_class
Definition: libvo-amrwbenc.c:48
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:576
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
abs
#define abs(x)
Definition: cuda_runtime.h:35
ff_libvo_amrwbenc_encoder
AVCodec ff_libvo_amrwbenc_encoder
Definition: libvo-amrwbenc.c:142
AMRWBContext::allow_dtx
int allow_dtx
Definition: libvo-amrwbenc.c:40
AVPacket::size
int size
Definition: packet.h:356
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:186
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
size
int size
Definition: twinvq_data.h:11134
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
MAX_PACKET_SIZE
#define MAX_PACKET_SIZE
Definition: libvo-amrwbenc.c:33
bitrate
int64_t bitrate
Definition: h264_levels.c:131
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:348
internal.h
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:197
avcodec.h
ret
ret
Definition: filter_design.txt:187
AVClass::class_name
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
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
AVCodecContext::strict_std_compliance
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:1589
AVCodecContext
main external API structure.
Definition: avcodec.h:526
AMRWBContext
Definition: amrwbdec.c:47
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:223
AMRWBContext::last_bitrate
int last_bitrate
Definition: libvo-amrwbenc.c:39
amr_wb_encode_frame
static int amr_wb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libvo-amrwbenc.c:114
options
static const AVOption options[]
Definition: libvo-amrwbenc.c:43
mem.h
AVPacket
This structure stores compressed data.
Definition: packet.h:332
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
avstring.h
ff_alloc_packet2
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:32
snprintf
#define snprintf
Definition: snprintf.h:34