209 lines
6.6 KiB
C
209 lines
6.6 KiB
C
/*
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* AAC encoder wrapper
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* Copyright (c) 2010 Martin Storsjo
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <vo-aacenc/voAAC.h>
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#include <vo-aacenc/cmnMemory.h>
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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#include "mpeg4audio.h"
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#define FRAME_SIZE 1024
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#define ENC_DELAY 1600
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typedef struct AACContext {
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VO_AUDIO_CODECAPI codec_api;
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VO_HANDLE handle;
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VO_MEM_OPERATOR mem_operator;
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VO_CODEC_INIT_USERDATA user_data;
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VO_PBYTE end_buffer;
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AudioFrameQueue afq;
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int last_frame;
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int last_samples;
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} AACContext;
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static int aac_encode_close(AVCodecContext *avctx)
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{
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AACContext *s = avctx->priv_data;
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s->codec_api.Uninit(s->handle);
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#if FF_API_OLD_ENCODE_AUDIO
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av_freep(&avctx->coded_frame);
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#endif
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av_freep(&avctx->extradata);
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ff_af_queue_close(&s->afq);
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av_freep(&s->end_buffer);
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return 0;
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}
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static av_cold int aac_encode_init(AVCodecContext *avctx)
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{
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AACContext *s = avctx->priv_data;
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AACENC_PARAM params = { 0 };
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int index, ret;
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#if FF_API_OLD_ENCODE_AUDIO
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avctx->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame)
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return AVERROR(ENOMEM);
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#endif
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avctx->frame_size = FRAME_SIZE;
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avctx->delay = ENC_DELAY;
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s->last_frame = 2;
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ff_af_queue_init(avctx, &s->afq);
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s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 2);
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if (!s->end_buffer) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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voGetAACEncAPI(&s->codec_api);
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s->mem_operator.Alloc = cmnMemAlloc;
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s->mem_operator.Copy = cmnMemCopy;
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s->mem_operator.Free = cmnMemFree;
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s->mem_operator.Set = cmnMemSet;
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s->mem_operator.Check = cmnMemCheck;
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s->user_data.memflag = VO_IMF_USERMEMOPERATOR;
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s->user_data.memData = &s->mem_operator;
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s->codec_api.Init(&s->handle, VO_AUDIO_CodingAAC, &s->user_data);
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params.sampleRate = avctx->sample_rate;
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params.bitRate = avctx->bit_rate;
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params.nChannels = avctx->channels;
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params.adtsUsed = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
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if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, ¶ms)
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!= VO_ERR_NONE) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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for (index = 0; index < 16; index++)
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if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[index])
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break;
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if (index == 16) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
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avctx->sample_rate);
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ret = AVERROR(ENOSYS);
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goto error;
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}
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if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
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avctx->extradata_size = 2;
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avctx->extradata = av_mallocz(avctx->extradata_size +
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FF_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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avctx->extradata[0] = 0x02 << 3 | index >> 1;
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avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
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}
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return 0;
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error:
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aac_encode_close(avctx);
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return ret;
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}
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static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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AACContext *s = avctx->priv_data;
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VO_CODECBUFFER input = { 0 }, output = { 0 };
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VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
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VO_PBYTE samples;
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int ret;
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/* handle end-of-stream small frame and flushing */
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if (!frame) {
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if (s->last_frame <= 0)
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return 0;
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if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
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s->last_samples = 0;
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s->last_frame--;
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}
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s->last_frame--;
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memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
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samples = s->end_buffer;
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} else {
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if (frame->nb_samples < avctx->frame_size) {
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s->last_samples = frame->nb_samples;
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memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
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samples = s->end_buffer;
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} else {
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samples = (VO_PBYTE)frame->data[0];
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}
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/* add current frame to the queue */
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if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
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return ret;
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}
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if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
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return ret;
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input.Buffer = samples;
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input.Length = 2 * avctx->channels * avctx->frame_size;
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output.Buffer = avpkt->data;
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output.Length = avpkt->size;
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s->codec_api.SetInputData(s->handle, &input);
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if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
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!= VO_ERR_NONE) {
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av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
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return AVERROR(EINVAL);
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}
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/* Get the next frame pts/duration */
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ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
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&avpkt->duration);
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avpkt->size = output.Length;
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*got_packet_ptr = 1;
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return 0;
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}
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/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
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* failures */
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static const int mpeg4audio_sample_rates[16] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000, 7350
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};
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AVCodec ff_libvo_aacenc_encoder = {
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.name = "libvo_aacenc",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_AAC,
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.priv_data_size = sizeof(AACContext),
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.init = aac_encode_init,
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.encode2 = aac_encode_frame,
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.close = aac_encode_close,
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.supported_samplerates = mpeg4audio_sample_rates,
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.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC (Advanced Audio Coding)"),
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};
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