| /* |
| * 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->delay = ENC_DELAY; |
| s->last_frame = 2; |
| ff_af_queue_init(avctx, &s->afq); |
| |
| s->end_buffer = av_mallocz(avctx->frame_size * avctx->channels * 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 }, |
| }; |