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