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Diffstat (limited to 'libavcodec/libvo-aacenc.c')
-rw-r--r--libavcodec/libvo-aacenc.c194
1 files changed, 0 insertions, 194 deletions
diff --git a/libavcodec/libvo-aacenc.c b/libavcodec/libvo-aacenc.c
deleted file mode 100644
index 876ef4c197..0000000000
--- a/libavcodec/libvo-aacenc.c
+++ /dev/null
@@ -1,194 +0,0 @@
-/*
- * AAC encoder wrapper
- * Copyright (c) 2010 Martin Storsjo
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; 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(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 & AV_CODEC_FLAG_GLOBAL_HEADER);
- if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, &params)
- != 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 & AV_CODEC_FLAG_GLOBAL_HEADER) {
- avctx->extradata_size = 2;
- avctx->extradata = av_mallocz(avctx->extradata_size +
- AV_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_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {
- av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
- 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;
-}
-
-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,
- .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
- AV_SAMPLE_FMT_NONE },
-};