3 * Copyright (c) 2010 Martin Storsjo
5 * This file is part of FFmpeg.
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.
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.
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
22 #include <vo-aacenc/voAAC.h>
23 #include <vo-aacenc/cmnMemory.h>
26 #include "audio_frame_queue.h"
28 #include "mpeg4audio.h"
30 #define FRAME_SIZE 1024
31 #define ENC_DELAY 1600
33 typedef struct AACContext {
34 VO_AUDIO_CODECAPI codec_api;
36 VO_MEM_OPERATOR mem_operator;
37 VO_CODEC_INIT_USERDATA user_data;
45 static int aac_encode_close(AVCodecContext *avctx)
47 AACContext *s = avctx->priv_data;
49 s->codec_api.Uninit(s->handle);
50 #if FF_API_OLD_ENCODE_AUDIO
51 av_freep(&avctx->coded_frame);
53 av_freep(&avctx->extradata);
54 ff_af_queue_close(&s->afq);
55 av_freep(&s->end_buffer);
60 static av_cold int aac_encode_init(AVCodecContext *avctx)
62 AACContext *s = avctx->priv_data;
63 AACENC_PARAM params = { 0 };
66 #if FF_API_OLD_ENCODE_AUDIO
67 avctx->coded_frame = avcodec_alloc_frame();
68 if (!avctx->coded_frame)
69 return AVERROR(ENOMEM);
71 avctx->frame_size = FRAME_SIZE;
72 avctx->delay = ENC_DELAY;
74 ff_af_queue_init(avctx, &s->afq);
76 s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 2);
78 ret = AVERROR(ENOMEM);
82 voGetAACEncAPI(&s->codec_api);
84 s->mem_operator.Alloc = cmnMemAlloc;
85 s->mem_operator.Copy = cmnMemCopy;
86 s->mem_operator.Free = cmnMemFree;
87 s->mem_operator.Set = cmnMemSet;
88 s->mem_operator.Check = cmnMemCheck;
89 s->user_data.memflag = VO_IMF_USERMEMOPERATOR;
90 s->user_data.memData = &s->mem_operator;
91 s->codec_api.Init(&s->handle, VO_AUDIO_CodingAAC, &s->user_data);
93 params.sampleRate = avctx->sample_rate;
94 params.bitRate = avctx->bit_rate;
95 params.nChannels = avctx->channels;
96 params.adtsUsed = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
97 if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, ¶ms)
99 av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
100 ret = AVERROR(EINVAL);
104 for (index = 0; index < 16; index++)
105 if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[index])
108 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
110 ret = AVERROR(ENOSYS);
113 if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
114 avctx->extradata_size = 2;
115 avctx->extradata = av_mallocz(avctx->extradata_size +
116 FF_INPUT_BUFFER_PADDING_SIZE);
117 if (!avctx->extradata) {
118 ret = AVERROR(ENOMEM);
122 avctx->extradata[0] = 0x02 << 3 | index >> 1;
123 avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
127 aac_encode_close(avctx);
131 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
132 const AVFrame *frame, int *got_packet_ptr)
134 AACContext *s = avctx->priv_data;
135 VO_CODECBUFFER input = { 0 }, output = { 0 };
136 VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
140 /* handle end-of-stream small frame and flushing */
142 if (s->last_frame <= 0)
144 if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
149 memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
150 samples = s->end_buffer;
152 if (frame->nb_samples < avctx->frame_size) {
153 s->last_samples = frame->nb_samples;
154 memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
155 samples = s->end_buffer;
157 samples = (VO_PBYTE)frame->data[0];
159 /* add current frame to the queue */
160 if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
164 if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
167 input.Buffer = samples;
168 input.Length = 2 * avctx->channels * avctx->frame_size;
169 output.Buffer = avpkt->data;
170 output.Length = avpkt->size;
172 s->codec_api.SetInputData(s->handle, &input);
173 if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
175 av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
176 return AVERROR(EINVAL);
179 /* Get the next frame pts/duration */
180 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
183 avpkt->size = output.Length;
188 AVCodec ff_libvo_aacenc_encoder = {
189 .name = "libvo_aacenc",
190 .type = AVMEDIA_TYPE_AUDIO,
192 .priv_data_size = sizeof(AACContext),
193 .init = aac_encode_init,
194 .encode2 = aac_encode_frame,
195 .close = aac_encode_close,
196 .supported_samplerates = avpriv_mpeg4audio_sample_rates,
197 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
198 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
199 AV_SAMPLE_FMT_NONE },
200 .long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC"),