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authorMichael Niedermayer <[email protected]>2011-10-02 21:37:59 +0200
committerMichael Niedermayer <[email protected]>2011-10-02 21:37:59 +0200
commit7fc85451fd47a4607f3cc47d1daa84ae122f5a46 (patch)
treef94ae472ddeea47eee50ef33ee6ecdd3e7e1e501 /libavcodec/libaacplus.c
parent42c8fdb943b210b2f79e2510a91ca0f542c9bad0 (diff)
parentb89a0c9d7f4c4a23d709761033ad5e2f9c2881fa (diff)
Merge branch 'release/0.8' into release/0.7
* release/0.8: (185 commits) h264: fix intra 16x16 mode check when using mbaff and constrained_intra_pred. h264: check for invalid bit depth value. h264: add entries for 11 and 12 bits in ff_h264_chroma_qp[][] h264: fix the check for invalid SPS:num_ref_frames. h264: do not let invalid values in h->ref_count on ff_h264_decode_ref_pic_list_reordering() errors. Reject video with non multiple of 16 width/height in the 4xm decoder. 4xm decoder: fix data size for i2 frames. 4xm decoder: print some error messages in case of errors. Check for out of bound accesses in the 4xm decoder. Prevent block size from inreasing in the shorten decoder. Check for out of bound reads in PTX decoder. Clear FF_INPUT_BUFFER_PADDING_SIZE bytes at the end of the temporary buffers used in 4xm decoder. Fix the check for missing references in ff_er_frame_end() for H264. Prevent NULL dereference when the huffman table is invalid in the 4xm decoder. Fix use of uninitialized memory in 4X Technologies demuxer. h264: increase ref_poc size to 32 as it can be per field. h264: set unused ref_counts to 0 as a precautionary meassure. Remove Chnagelog it has nothing to do with reality fate: fix motion pixels checksum change caused by backported bugfix avienc: Add a limit on the number of skiped frames muxed in a row. ... Conflicts: Doxyfile RELEASE VERSION libavformat/utils.c Merged-by: Michael Niedermayer <[email protected]>
Diffstat (limited to 'libavcodec/libaacplus.c')
-rw-r--r--libavcodec/libaacplus.c136
1 files changed, 136 insertions, 0 deletions
diff --git a/libavcodec/libaacplus.c b/libavcodec/libaacplus.c
new file mode 100644
index 0000000000..c8c87be549
--- /dev/null
+++ b/libavcodec/libaacplus.c
@@ -0,0 +1,136 @@
+/*
+ * Interface to libaacplus for aac+ (sbr+ps) encoding
+ * Copyright (c) 2010 tipok <[email protected]>
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * Interface to libaacplus for aac+ (sbr+ps) encoding.
+ */
+
+#include "avcodec.h"
+#include <aacplus.h>
+
+typedef struct aacPlusAudioContext {
+ aacplusEncHandle aacplus_handle;
+} aacPlusAudioContext;
+
+static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
+{
+ aacPlusAudioContext *s = avctx->priv_data;
+ aacplusEncConfiguration *aacplus_cfg;
+ unsigned long samples_input, max_bytes_output;
+
+ /* number of channels */
+ if (avctx->channels < 1 || avctx->channels > 2) {
+ av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
+ return -1;
+ }
+
+ s->aacplus_handle = aacplusEncOpen(avctx->sample_rate,
+ avctx->channels,
+ &samples_input, &max_bytes_output);
+ if(!s->aacplus_handle) {
+ av_log(avctx, AV_LOG_ERROR, "can't open encoder\n");
+ return -1;
+ }
+
+ /* check aacplus version */
+ aacplus_cfg = aacplusEncGetCurrentConfiguration(s->aacplus_handle);
+
+ /* put the options in the configuration struct */
+ if(avctx->profile != FF_PROFILE_AAC_LOW && avctx->profile != FF_PROFILE_UNKNOWN) {
+ av_log(avctx, AV_LOG_ERROR, "invalid AAC profile: %d, only LC supported\n", avctx->profile);
+ aacplusEncClose(s->aacplus_handle);
+ return -1;
+ }
+
+ aacplus_cfg->bitRate = avctx->bit_rate;
+ aacplus_cfg->bandWidth = avctx->cutoff;
+ if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
+ aacplus_cfg->outputFormat = 0; //raw aac
+ }
+ aacplus_cfg->inputFormat = AACPLUS_INPUT_16BIT;
+ if (!aacplusEncSetConfiguration(s->aacplus_handle, aacplus_cfg)) {
+ av_log(avctx, AV_LOG_ERROR, "libaacplus doesn't support this output format!\n");
+ return -1;
+ }
+
+ avctx->frame_size = samples_input / avctx->channels;
+
+ avctx->coded_frame= avcodec_alloc_frame();
+ avctx->coded_frame->key_frame= 1;
+
+ /* Set decoder specific info */
+ avctx->extradata_size = 0;
+ if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
+
+ unsigned char *buffer = NULL;
+ unsigned long decoder_specific_info_size;
+
+ if (aacplusEncGetDecoderSpecificInfo(s->aacplus_handle, &buffer,
+ &decoder_specific_info_size) == 1) {
+ avctx->extradata = av_malloc(decoder_specific_info_size + FF_INPUT_BUFFER_PADDING_SIZE);
+ avctx->extradata_size = decoder_specific_info_size;
+ memcpy(avctx->extradata, buffer, avctx->extradata_size);
+ }
+#undef free
+ free(buffer);
+#define free please_use_av_free
+ }
+ return 0;
+}
+
+static int aacPlus_encode_frame(AVCodecContext *avctx,
+ unsigned char *frame, int buf_size, void *data)
+{
+ aacPlusAudioContext *s = avctx->priv_data;
+ int bytes_written;
+
+ bytes_written = aacplusEncEncode(s->aacplus_handle,
+ data,
+ avctx->frame_size * avctx->channels,
+ frame,
+ buf_size);
+
+ return bytes_written;
+}
+
+static av_cold int aacPlus_encode_close(AVCodecContext *avctx)
+{
+ aacPlusAudioContext *s = avctx->priv_data;
+
+ av_freep(&avctx->coded_frame);
+ av_freep(&avctx->extradata);
+
+ aacplusEncClose(s->aacplus_handle);
+ return 0;
+}
+
+AVCodec ff_libaacplus_encoder = {
+ "libaacplus",
+ AVMEDIA_TYPE_AUDIO,
+ CODEC_ID_AAC,
+ sizeof(aacPlusAudioContext),
+ aacPlus_encode_init,
+ aacPlus_encode_frame,
+ aacPlus_encode_close,
+ .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+ .long_name = NULL_IF_CONFIG_SMALL("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"),
+};