The internal encoder is superior to libvo-aacenc. Signed-off-by: Timothy Gu <timothygu99@gmail.com>pull/172/merge
Kieran Kunhya
9 years ago
committed by
Timothy Gu
parent
9ba54c1b82
commit
e07e88cd82
8 changed files with 5 additions and 246 deletions
@ -1,200 +0,0 @@ |
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/*
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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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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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avctx->frame_size = FRAME_SIZE; |
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avctx->initial_padding = 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_array(avctx->channels, avctx->frame_size * 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 & AV_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 & AV_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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AV_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), 0)) < 0) |
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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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.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC (Advanced Audio Coding)"), |
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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 = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_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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}; |
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