libfaac.c
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1 /*
2  * Interface to libfaac for aac encoding
3  * Copyright (c) 2002 Gildas Bazin <gbazin@netcourrier.com>
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  * Interface to libfaac for aac encoding.
25  */
26 
27 #include <faac.h>
28 
30 #include "libavutil/common.h"
31 #include "avcodec.h"
32 #include "audio_frame_queue.h"
33 #include "internal.h"
34 
35 
36 /* libfaac has an encoder delay of 1024 samples */
37 #define FAAC_DELAY_SAMPLES 1024
38 
39 typedef struct FaacAudioContext {
40  faacEncHandle faac_handle;
43 
45 {
46  FaacAudioContext *s = avctx->priv_data;
47 
48  av_freep(&avctx->extradata);
50 
51  if (s->faac_handle)
52  faacEncClose(s->faac_handle);
53 
54  return 0;
55 }
56 
57 static const int channel_maps[][6] = {
58  { 2, 0, 1 }, //< C L R
59  { 2, 0, 1, 3 }, //< C L R Cs
60  { 2, 0, 1, 3, 4 }, //< C L R Ls Rs
61  { 2, 0, 1, 4, 5, 3 }, //< C L R Ls Rs LFE
62 };
63 
65 {
66  FaacAudioContext *s = avctx->priv_data;
67  faacEncConfigurationPtr faac_cfg;
68  unsigned long samples_input, max_bytes_output;
69  int ret;
70 
71  /* number of channels */
72  if (avctx->channels < 1 || avctx->channels > 6) {
73  av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
74  ret = AVERROR(EINVAL);
75  goto error;
76  }
77 
78  s->faac_handle = faacEncOpen(avctx->sample_rate,
79  avctx->channels,
80  &samples_input, &max_bytes_output);
81  if (!s->faac_handle) {
82  av_log(avctx, AV_LOG_ERROR, "error in faacEncOpen()\n");
83  ret = AVERROR_UNKNOWN;
84  goto error;
85  }
86 
87  /* check faac version */
88  faac_cfg = faacEncGetCurrentConfiguration(s->faac_handle);
89  if (faac_cfg->version != FAAC_CFG_VERSION) {
90  av_log(avctx, AV_LOG_ERROR, "wrong libfaac version (compiled for: %d, using %d)\n", FAAC_CFG_VERSION, faac_cfg->version);
91  ret = AVERROR(EINVAL);
92  goto error;
93  }
94 
95  /* put the options in the configuration struct */
96  switch(avctx->profile) {
98  faac_cfg->aacObjectType = MAIN;
99  break;
100  case FF_PROFILE_UNKNOWN:
101  case FF_PROFILE_AAC_LOW:
102  faac_cfg->aacObjectType = LOW;
103  break;
104  case FF_PROFILE_AAC_SSR:
105  faac_cfg->aacObjectType = SSR;
106  break;
107  case FF_PROFILE_AAC_LTP:
108  faac_cfg->aacObjectType = LTP;
109  break;
110  default:
111  av_log(avctx, AV_LOG_ERROR, "invalid AAC profile\n");
112  ret = AVERROR(EINVAL);
113  goto error;
114  }
115  faac_cfg->mpegVersion = MPEG4;
116  faac_cfg->useTns = 0;
117  faac_cfg->allowMidside = 1;
118  faac_cfg->bitRate = avctx->bit_rate / avctx->channels;
119  faac_cfg->bandWidth = avctx->cutoff;
120  if(avctx->flags & CODEC_FLAG_QSCALE) {
121  faac_cfg->bitRate = 0;
122  faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA;
123  }
124  faac_cfg->outputFormat = 1;
125  faac_cfg->inputFormat = FAAC_INPUT_16BIT;
126  if (avctx->channels > 2)
127  memcpy(faac_cfg->channel_map, channel_maps[avctx->channels-3],
128  avctx->channels * sizeof(int));
129 
130  avctx->frame_size = samples_input / avctx->channels;
131 
132  /* Set decoder specific info */
133  avctx->extradata_size = 0;
134  if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
135 
136  unsigned char *buffer = NULL;
137  unsigned long decoder_specific_info_size;
138 
139  if (!faacEncGetDecoderSpecificInfo(s->faac_handle, &buffer,
140  &decoder_specific_info_size)) {
141  avctx->extradata = av_malloc(decoder_specific_info_size + FF_INPUT_BUFFER_PADDING_SIZE);
142  if (!avctx->extradata) {
143  ret = AVERROR(ENOMEM);
144  goto error;
145  }
146  avctx->extradata_size = decoder_specific_info_size;
147  memcpy(avctx->extradata, buffer, avctx->extradata_size);
148  faac_cfg->outputFormat = 0;
149  }
150  free(buffer);
151  }
152 
153  if (!faacEncSetConfiguration(s->faac_handle, faac_cfg)) {
154  av_log(avctx, AV_LOG_ERROR, "libfaac doesn't support this output format!\n");
155  ret = AVERROR(EINVAL);
156  goto error;
157  }
158 
159  avctx->delay = FAAC_DELAY_SAMPLES;
160  ff_af_queue_init(avctx, &s->afq);
161 
162  return 0;
163 error:
164  Faac_encode_close(avctx);
165  return ret;
166 }
167 
168 static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
169  const AVFrame *frame, int *got_packet_ptr)
170 {
171  FaacAudioContext *s = avctx->priv_data;
172  int bytes_written, ret;
173  int num_samples = frame ? frame->nb_samples : 0;
174  void *samples = frame ? frame->data[0] : NULL;
175 
176  if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels)) < 0)
177  return ret;
178 
179  bytes_written = faacEncEncode(s->faac_handle, samples,
180  num_samples * avctx->channels,
181  avpkt->data, avpkt->size);
182  if (bytes_written < 0) {
183  av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
184  return bytes_written;
185  }
186 
187  /* add current frame to the queue */
188  if (frame) {
189  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
190  return ret;
191  }
192 
193  if (!bytes_written)
194  return 0;
195 
196  /* Get the next frame pts/duration */
197  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
198  &avpkt->duration);
199 
200  avpkt->size = bytes_written;
201  *got_packet_ptr = 1;
202  return 0;
203 }
204 
205 static const AVProfile profiles[] = {
206  { FF_PROFILE_AAC_MAIN, "Main" },
207  { FF_PROFILE_AAC_LOW, "LC" },
208  { FF_PROFILE_AAC_SSR, "SSR" },
209  { FF_PROFILE_AAC_LTP, "LTP" },
210  { FF_PROFILE_UNKNOWN },
211 };
212 
213 static const uint64_t faac_channel_layouts[] = {
220  0
221 };
222 
224  .name = "libfaac",
225  .type = AVMEDIA_TYPE_AUDIO,
226  .id = AV_CODEC_ID_AAC,
227  .priv_data_size = sizeof(FaacAudioContext),
229  .encode2 = Faac_encode_frame,
232  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
234  .long_name = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio Coding)"),
235  .profiles = NULL_IF_CONFIG_SMALL(profiles),
236  .channel_layouts = faac_channel_layouts,
237 };
faacEncHandle faac_handle
Definition: libfaac.c:40
const char * s
Definition: avisynth_c.h:668
#define FF_PROFILE_AAC_SSR
This structure describes decoded (raw) audio or video data.
Definition: frame.h:76
#define AV_CH_LAYOUT_SURROUND
static av_cold int Faac_encode_init(AVCodecContext *avctx)
Definition: libfaac.c:64
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
#define AV_CH_LAYOUT_4POINT0
#define AV_CH_LAYOUT_STEREO
signed 16 bits
Definition: samplefmt.h:52
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
AudioFrameQueue afq
Definition: libfaac.c:41
#define av_cold
Definition: attributes.h:78
#define CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
#define FF_PROFILE_UNKNOWN
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static av_cold int Faac_encode_close(AVCodecContext *avctx)
Definition: libfaac.c:44
AVCodec ff_libfaac_encoder
Definition: libfaac.c:223
uint8_t * data
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libfaac.c:168
frame
Definition: stft.m:14
#define FF_PROFILE_AAC_MAIN
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define FAAC_DELAY_SAMPLES
Definition: libfaac.c:37
int flags
CODEC_FLAG_*.
#define CODEC_FLAG_QSCALE
Use fixed qscale.
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:246
const char * name
Name of the codec implementation.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
external API header
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
int bit_rate
the average bitrate
audio channel layout utility functions
ret
Definition: avfilter.c:821
static const uint64_t faac_channel_layouts[]
Definition: libfaac.c:213
#define AV_CH_LAYOUT_5POINT1_BACK
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
int frame_size
Number of samples per channel in an audio frame.
NULL
Definition: eval.c:55
int sample_rate
samples per second
#define FF_PROFILE_AAC_LTP
main external API structure.
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:375
#define FF_PROFILE_AAC_LOW
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:148
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:73
#define AV_CH_LAYOUT_5POINT0_BACK
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer 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.Buffer references ownership and permissions
int global_quality
Global quality for codecs which cannot change it per frame.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:87
static const AVProfile profiles[]
Definition: libfaac.c:205
common internal api header.
common internal and external API header
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
static const int channel_maps[][6]
Definition: libfaac.c:57
AVProfile.
the buffer and buffer reference mechanism is intended to as much as expensive copies of that data while still allowing the filters to produce correct results The data is stored in buffers represented by AVFilterBuffer structures They must not be accessed but through references stored in AVFilterBufferRef structures Several references can point to the same buffer
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
int cutoff
Audio cutoff bandwidth (0 means "automatic")
struct FaacAudioContext FaacAudioContext
void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int *duration)
Remove frame(s) from the queue.
int channels
number of audio channels
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:169
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:700
Filter the word “frame” indicates either a video frame or a group of audio samples
#define MAIN
Definition: vf_overlay.c:76
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
int delay
Codec delay.
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:127
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
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