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libvo-aacenc.c
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1 /*
2  * AAC encoder wrapper
3  * Copyright (c) 2010 Martin Storsjo
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-aacenc/voAAC.h>
23 #include <vo-aacenc/cmnMemory.h>
24 
25 #include "avcodec.h"
26 #include "audio_frame_queue.h"
27 #include "internal.h"
28 #include "mpeg4audio.h"
29 
30 #define FRAME_SIZE 1024
31 #define ENC_DELAY 1600
32 
33 typedef struct AACContext {
34  VO_AUDIO_CODECAPI codec_api;
35  VO_HANDLE handle;
36  VO_MEM_OPERATOR mem_operator;
37  VO_CODEC_INIT_USERDATA user_data;
38  VO_PBYTE end_buffer;
42 } AACContext;
43 
44 
45 static int aac_encode_close(AVCodecContext *avctx)
46 {
47  AACContext *s = avctx->priv_data;
48 
49  s->codec_api.Uninit(s->handle);
50  av_freep(&avctx->extradata);
52  av_freep(&s->end_buffer);
53 
54  return 0;
55 }
56 
58 {
59  AACContext *s = avctx->priv_data;
60  AACENC_PARAM params = { 0 };
61  int index, ret;
62 
63  avctx->frame_size = FRAME_SIZE;
64  avctx->delay = ENC_DELAY;
65  s->last_frame = 2;
66  ff_af_queue_init(avctx, &s->afq);
67 
68  s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 2);
69  if (!s->end_buffer) {
70  ret = AVERROR(ENOMEM);
71  goto error;
72  }
73 
74  voGetAACEncAPI(&s->codec_api);
75 
76  s->mem_operator.Alloc = cmnMemAlloc;
77  s->mem_operator.Copy = cmnMemCopy;
78  s->mem_operator.Free = cmnMemFree;
79  s->mem_operator.Set = cmnMemSet;
80  s->mem_operator.Check = cmnMemCheck;
81  s->user_data.memflag = VO_IMF_USERMEMOPERATOR;
82  s->user_data.memData = &s->mem_operator;
83  s->codec_api.Init(&s->handle, VO_AUDIO_CodingAAC, &s->user_data);
84 
85  params.sampleRate = avctx->sample_rate;
86  params.bitRate = avctx->bit_rate;
87  params.nChannels = avctx->channels;
88  params.adtsUsed = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
89  if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, &params)
90  != VO_ERR_NONE) {
91  av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
92  ret = AVERROR(EINVAL);
93  goto error;
94  }
95 
96  for (index = 0; index < 16; index++)
97  if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[index])
98  break;
99  if (index == 16) {
100  av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
101  avctx->sample_rate);
102  ret = AVERROR(ENOSYS);
103  goto error;
104  }
105  if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
106  avctx->extradata_size = 2;
107  avctx->extradata = av_mallocz(avctx->extradata_size +
109  if (!avctx->extradata) {
110  ret = AVERROR(ENOMEM);
111  goto error;
112  }
113 
114  avctx->extradata[0] = 0x02 << 3 | index >> 1;
115  avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
116  }
117  return 0;
118 error:
119  aac_encode_close(avctx);
120  return ret;
121 }
122 
123 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
124  const AVFrame *frame, int *got_packet_ptr)
125 {
126  AACContext *s = avctx->priv_data;
127  VO_CODECBUFFER input = { 0 }, output = { 0 };
128  VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
129  VO_PBYTE samples;
130  int ret;
131 
132  /* handle end-of-stream small frame and flushing */
133  if (!frame) {
134  if (s->last_frame <= 0)
135  return 0;
136  if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
137  s->last_samples = 0;
138  s->last_frame--;
139  }
140  s->last_frame--;
141  memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
142  samples = s->end_buffer;
143  } else {
144  if (frame->nb_samples < avctx->frame_size) {
145  s->last_samples = frame->nb_samples;
146  memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
147  samples = s->end_buffer;
148  } else {
149  samples = (VO_PBYTE)frame->data[0];
150  }
151  /* add current frame to the queue */
152  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
153  return ret;
154  }
155 
156  if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))) < 0)
157  return ret;
158 
159  input.Buffer = samples;
160  input.Length = 2 * avctx->channels * avctx->frame_size;
161  output.Buffer = avpkt->data;
162  output.Length = avpkt->size;
163 
164  s->codec_api.SetInputData(s->handle, &input);
165  if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
166  != VO_ERR_NONE) {
167  av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
168  return AVERROR(EINVAL);
169  }
170 
171  /* Get the next frame pts/duration */
172  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
173  &avpkt->duration);
174 
175  avpkt->size = output.Length;
176  *got_packet_ptr = 1;
177  return 0;
178 }
179 
180 /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
181  * failures */
182 static const int mpeg4audio_sample_rates[16] = {
183  96000, 88200, 64000, 48000, 44100, 32000,
184  24000, 22050, 16000, 12000, 11025, 8000, 7350
185 };
186 
188  .name = "libvo_aacenc",
189  .long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC (Advanced Audio Coding)"),
190  .type = AVMEDIA_TYPE_AUDIO,
191  .id = AV_CODEC_ID_AAC,
192  .priv_data_size = sizeof(AACContext),
194  .encode2 = aac_encode_frame,
198  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
200 };