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The aim of this is twofold: a) Clang warns when setting a deprecated
field in a definition and because several of the widely set
AVCodec fields are deprecated, one gets several hundred warnings
from Clang for an ordinary build. Yet fortunately Clang (unlike GCC)
allows to disable deprecation warnings inside a definition, so
that one can create simple macros to set these fields that also suppress
deprecation warnings for Clang. This has already been done in
fdff1b9cbf for AVCodec.channel_layouts.
b) Using macros will allow to easily migrate these fields to internal ones.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
681 lines
25 KiB
C
681 lines
25 KiB
C
/*
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* Audio Toolbox system codecs
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*
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* copyright (c) 2016 rcombs
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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 <AudioToolbox/AudioToolbox.h>
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#define FF_BUFQUEUE_SIZE 256
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#include "libavfilter/bufferqueue.h"
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#include "config.h"
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#include "audio_frame_queue.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#include "codec_internal.h"
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#include "encode.h"
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#include "internal.h"
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#include "libavformat/isom.h"
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "libavutil/log.h"
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typedef struct ATDecodeContext {
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AVClass *av_class;
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int mode;
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int quality;
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AudioConverterRef converter;
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struct FFBufQueue frame_queue;
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struct FFBufQueue used_frame_queue;
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unsigned pkt_size;
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AudioFrameQueue afq;
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int eof;
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int frame_size;
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AVFrame* encoding_frame;
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} ATDecodeContext;
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static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
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{
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switch (codec) {
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case AV_CODEC_ID_AAC:
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switch (profile) {
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case AV_PROFILE_AAC_LOW:
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default:
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return kAudioFormatMPEG4AAC;
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case AV_PROFILE_AAC_HE:
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return kAudioFormatMPEG4AAC_HE;
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case AV_PROFILE_AAC_HE_V2:
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return kAudioFormatMPEG4AAC_HE_V2;
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case AV_PROFILE_AAC_LD:
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return kAudioFormatMPEG4AAC_LD;
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#if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060
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case AV_PROFILE_AAC_ELD:
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return kAudioFormatMPEG4AAC_ELD;
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#endif
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}
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case AV_CODEC_ID_ADPCM_IMA_QT:
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return kAudioFormatAppleIMA4;
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case AV_CODEC_ID_ALAC:
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return kAudioFormatAppleLossless;
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#if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060
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case AV_CODEC_ID_ILBC:
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return kAudioFormatiLBC;
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#endif
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case AV_CODEC_ID_PCM_ALAW:
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return kAudioFormatALaw;
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case AV_CODEC_ID_PCM_MULAW:
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return kAudioFormatULaw;
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default:
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av_assert0(!"Invalid codec ID!");
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return 0;
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}
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}
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static int ffat_update_ctx(AVCodecContext *avctx)
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{
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ATDecodeContext *at = avctx->priv_data;
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UInt32 size = sizeof(unsigned);
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AudioConverterPrimeInfo prime_info;
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AudioStreamBasicDescription out_format;
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AudioConverterGetProperty(at->converter,
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kAudioConverterPropertyMaximumOutputPacketSize,
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&size, &at->pkt_size);
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if (at->pkt_size <= 0)
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at->pkt_size = 1024 * 50;
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size = sizeof(prime_info);
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if (!AudioConverterGetProperty(at->converter,
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kAudioConverterPrimeInfo,
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&size, &prime_info)) {
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avctx->initial_padding = prime_info.leadingFrames;
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}
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size = sizeof(out_format);
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if (!AudioConverterGetProperty(at->converter,
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kAudioConverterCurrentOutputStreamDescription,
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&size, &out_format)) {
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if (out_format.mFramesPerPacket) {
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avctx->frame_size = out_format.mFramesPerPacket;
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} else {
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/* The doc on mFramesPerPacket says:
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* For formats with a variable number of frames per packet, such as
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* Ogg Vorbis, set this field to 0.
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* Looks like it means for decoding. There is no known case that
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* mFramesPerPacket is zero for encoding. Use a default value for safety.
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*/
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avctx->frame_size = 1024;
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av_log(avctx, AV_LOG_WARNING, "Missing mFramesPerPacket\n");
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}
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if (out_format.mBytesPerPacket && avctx->codec_id == AV_CODEC_ID_ILBC)
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avctx->block_align = out_format.mBytesPerPacket;
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} else {
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av_log(avctx, AV_LOG_ERROR, "Get OutputStreamDescription failed\n");
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return AVERROR_EXTERNAL;
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}
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at->frame_size = avctx->frame_size;
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if (avctx->codec_id == AV_CODEC_ID_PCM_MULAW ||
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avctx->codec_id == AV_CODEC_ID_PCM_ALAW) {
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at->pkt_size *= 1024;
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avctx->frame_size *= 1024;
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}
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return 0;
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}
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static int read_descr(GetByteContext *gb, int *tag)
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{
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int len = 0;
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int count = 4;
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*tag = bytestream2_get_byte(gb);
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while (count--) {
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int c = bytestream2_get_byte(gb);
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len = (len << 7) | (c & 0x7f);
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if (!(c & 0x80))
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break;
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}
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return len;
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}
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static int get_ilbc_mode(AVCodecContext *avctx)
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{
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if (avctx->block_align == 38)
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return 20;
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else if (avctx->block_align == 50)
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return 30;
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else if (avctx->bit_rate > 0)
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return avctx->bit_rate <= 14000 ? 30 : 20;
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else
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return 30;
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}
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static av_cold int get_channel_label(int channel)
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{
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uint64_t map = 1 << channel;
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if (map <= AV_CH_LOW_FREQUENCY)
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return channel + 1;
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else if (map <= AV_CH_BACK_RIGHT)
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return channel + 29;
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else if (map <= AV_CH_BACK_CENTER)
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return channel - 1;
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else if (map <= AV_CH_SIDE_RIGHT)
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return channel - 4;
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else if (map <= AV_CH_TOP_BACK_RIGHT)
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return channel + 1;
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else if (map <= AV_CH_STEREO_RIGHT)
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return -1;
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else if (map <= AV_CH_WIDE_RIGHT)
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return channel + 4;
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else if (map <= AV_CH_SURROUND_DIRECT_RIGHT)
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return channel - 23;
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else if (map == AV_CH_LOW_FREQUENCY_2)
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return kAudioChannelLabel_LFE2;
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else
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return -1;
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}
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static int remap_layout(AudioChannelLayout *layout, const AVChannelLayout *in_layout)
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{
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int i;
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layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
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layout->mNumberChannelDescriptions = in_layout->nb_channels;
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for (i = 0; i < in_layout->nb_channels; i++) {
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int c, label;
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c = av_channel_layout_channel_from_index(in_layout, i);
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if (c < 0 || c >= 64)
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return AVERROR(EINVAL);
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label = get_channel_label(c);
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layout->mChannelDescriptions[i].mChannelLabel = label;
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if (label < 0)
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return AVERROR(EINVAL);
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c++;
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}
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return 0;
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}
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static int get_aac_tag(const AVChannelLayout *in_layout)
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{
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static const struct {
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AVChannelLayout chl;
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int tag;
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} map[] = {
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{ AV_CHANNEL_LAYOUT_MONO, kAudioChannelLayoutTag_Mono },
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{ AV_CHANNEL_LAYOUT_STEREO, kAudioChannelLayoutTag_Stereo },
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{ AV_CHANNEL_LAYOUT_QUAD, kAudioChannelLayoutTag_AAC_Quadraphonic },
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{ AV_CHANNEL_LAYOUT_OCTAGONAL, kAudioChannelLayoutTag_AAC_Octagonal },
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{ AV_CHANNEL_LAYOUT_SURROUND, kAudioChannelLayoutTag_AAC_3_0 },
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{ AV_CHANNEL_LAYOUT_4POINT0, kAudioChannelLayoutTag_AAC_4_0 },
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{ AV_CHANNEL_LAYOUT_5POINT0, kAudioChannelLayoutTag_AAC_5_0 },
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{ AV_CHANNEL_LAYOUT_5POINT1, kAudioChannelLayoutTag_AAC_5_1 },
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{ AV_CHANNEL_LAYOUT_6POINT0, kAudioChannelLayoutTag_AAC_6_0 },
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{ AV_CHANNEL_LAYOUT_6POINT1, kAudioChannelLayoutTag_AAC_6_1 },
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{ AV_CHANNEL_LAYOUT_7POINT0, kAudioChannelLayoutTag_AAC_7_0 },
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{ AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK, kAudioChannelLayoutTag_AAC_7_1 },
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{ AV_CHANNEL_LAYOUT_7POINT1, kAudioChannelLayoutTag_MPEG_7_1_C },
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};
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int i;
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for (i = 0; i < FF_ARRAY_ELEMS(map); i++)
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if (!av_channel_layout_compare(in_layout, &map[i].chl))
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return map[i].tag;
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return 0;
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}
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static av_cold int ffat_init_encoder(AVCodecContext *avctx)
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{
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ATDecodeContext *at = avctx->priv_data;
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OSStatus status;
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int ret;
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AudioStreamBasicDescription in_format = {
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.mSampleRate = avctx->sample_rate,
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.mFormatID = kAudioFormatLinearPCM,
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.mFormatFlags = ((avctx->sample_fmt == AV_SAMPLE_FMT_FLT ||
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avctx->sample_fmt == AV_SAMPLE_FMT_DBL) ? kAudioFormatFlagIsFloat
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: avctx->sample_fmt == AV_SAMPLE_FMT_U8 ? 0
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: kAudioFormatFlagIsSignedInteger)
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| kAudioFormatFlagIsPacked,
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.mBytesPerPacket = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->ch_layout.nb_channels,
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.mFramesPerPacket = 1,
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.mBytesPerFrame = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->ch_layout.nb_channels,
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.mChannelsPerFrame = avctx->ch_layout.nb_channels,
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.mBitsPerChannel = av_get_bytes_per_sample(avctx->sample_fmt) * 8,
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};
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AudioStreamBasicDescription out_format = {
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.mSampleRate = avctx->sample_rate,
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.mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile),
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.mChannelsPerFrame = in_format.mChannelsPerFrame,
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};
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UInt32 layout_size = sizeof(AudioChannelLayout) +
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sizeof(AudioChannelDescription) * avctx->ch_layout.nb_channels;
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AudioChannelLayout *channel_layout = av_malloc(layout_size);
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if (!channel_layout)
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return AVERROR(ENOMEM);
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if (avctx->codec_id == AV_CODEC_ID_ILBC) {
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int mode = get_ilbc_mode(avctx);
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out_format.mFramesPerPacket = 8000 * mode / 1000;
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out_format.mBytesPerPacket = (mode == 20 ? 38 : 50);
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}
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status = AudioConverterNew(&in_format, &out_format, &at->converter);
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if (status != 0) {
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av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status);
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av_free(channel_layout);
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return AVERROR_UNKNOWN;
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}
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if (avctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
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av_channel_layout_default(&avctx->ch_layout, avctx->ch_layout.nb_channels);
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if ((status = remap_layout(channel_layout, &avctx->ch_layout)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "Invalid channel layout\n");
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av_free(channel_layout);
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return status;
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}
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if (AudioConverterSetProperty(at->converter, kAudioConverterInputChannelLayout,
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layout_size, channel_layout)) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported input channel layout\n");
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av_free(channel_layout);
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return AVERROR(EINVAL);
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}
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if (avctx->codec_id == AV_CODEC_ID_AAC) {
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int tag = get_aac_tag(&avctx->ch_layout);
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if (tag) {
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channel_layout->mChannelLayoutTag = tag;
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channel_layout->mNumberChannelDescriptions = 0;
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}
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}
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if (AudioConverterSetProperty(at->converter, kAudioConverterOutputChannelLayout,
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layout_size, channel_layout)) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported output channel layout\n");
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av_free(channel_layout);
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return AVERROR(EINVAL);
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}
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av_free(channel_layout);
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if (avctx->bits_per_raw_sample)
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AudioConverterSetProperty(at->converter,
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kAudioConverterPropertyBitDepthHint,
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sizeof(avctx->bits_per_raw_sample),
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&avctx->bits_per_raw_sample);
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#if !TARGET_OS_IPHONE
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if (at->mode == -1)
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at->mode = (avctx->flags & AV_CODEC_FLAG_QSCALE) ?
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kAudioCodecBitRateControlMode_Variable :
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kAudioCodecBitRateControlMode_Constant;
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AudioConverterSetProperty(at->converter, kAudioCodecPropertyBitRateControlMode,
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sizeof(at->mode), &at->mode);
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if (at->mode == kAudioCodecBitRateControlMode_Variable) {
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int q = avctx->global_quality / FF_QP2LAMBDA;
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if (q < 0 || q > 14) {
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av_log(avctx, AV_LOG_WARNING,
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"VBR quality %d out of range, should be 0-14\n", q);
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q = av_clip(q, 0, 14);
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}
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q = 127 - q * 9;
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AudioConverterSetProperty(at->converter, kAudioCodecPropertySoundQualityForVBR,
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sizeof(q), &q);
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} else
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#endif
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if (avctx->bit_rate > 0) {
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UInt32 rate = avctx->bit_rate;
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UInt32 size;
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status = AudioConverterGetPropertyInfo(at->converter,
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kAudioConverterApplicableEncodeBitRates,
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&size, NULL);
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if (!status && size) {
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UInt32 new_rate = rate;
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int count;
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int i;
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AudioValueRange *ranges = av_malloc(size);
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if (!ranges)
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return AVERROR(ENOMEM);
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AudioConverterGetProperty(at->converter,
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kAudioConverterApplicableEncodeBitRates,
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&size, ranges);
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count = size / sizeof(AudioValueRange);
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for (i = 0; i < count; i++) {
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AudioValueRange *range = &ranges[i];
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if (rate >= range->mMinimum && rate <= range->mMaximum) {
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new_rate = rate;
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break;
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} else if (rate > range->mMaximum) {
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new_rate = range->mMaximum;
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} else {
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new_rate = range->mMinimum;
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break;
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}
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}
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if (new_rate != rate) {
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av_log(avctx, AV_LOG_WARNING,
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"Bitrate %u not allowed; changing to %u\n", rate, new_rate);
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rate = new_rate;
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}
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av_free(ranges);
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}
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AudioConverterSetProperty(at->converter, kAudioConverterEncodeBitRate,
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sizeof(rate), &rate);
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}
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at->quality = 96 - at->quality * 32;
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AudioConverterSetProperty(at->converter, kAudioConverterCodecQuality,
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sizeof(at->quality), &at->quality);
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if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterCompressionMagicCookie,
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&avctx->extradata_size, NULL) &&
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avctx->extradata_size) {
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int extradata_size = avctx->extradata_size;
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uint8_t *extradata;
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if (!(avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE)))
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return AVERROR(ENOMEM);
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if (avctx->codec_id == AV_CODEC_ID_ALAC) {
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avctx->extradata_size = 0x24;
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AV_WB32(avctx->extradata, 0x24);
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AV_WB32(avctx->extradata + 4, MKBETAG('a','l','a','c'));
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extradata = avctx->extradata + 12;
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avctx->extradata_size = 0x24;
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} else {
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extradata = avctx->extradata;
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}
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status = AudioConverterGetProperty(at->converter,
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kAudioConverterCompressionMagicCookie,
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&extradata_size, extradata);
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if (status != 0) {
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av_log(avctx, AV_LOG_ERROR, "AudioToolbox cookie error: %i\n", (int)status);
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return AVERROR_UNKNOWN;
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} else if (avctx->codec_id == AV_CODEC_ID_AAC) {
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GetByteContext gb;
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int tag, len;
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bytestream2_init(&gb, extradata, extradata_size);
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do {
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len = read_descr(&gb, &tag);
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if (tag == MP4DecConfigDescrTag) {
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bytestream2_skip(&gb, 13);
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len = read_descr(&gb, &tag);
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if (tag == MP4DecSpecificDescrTag) {
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len = FFMIN(gb.buffer_end - gb.buffer, len);
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memmove(extradata, gb.buffer, len);
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avctx->extradata_size = len;
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break;
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}
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} else if (tag == MP4ESDescrTag) {
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int flags;
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bytestream2_skip(&gb, 2);
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flags = bytestream2_get_byte(&gb);
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if (flags & 0x80) //streamDependenceFlag
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bytestream2_skip(&gb, 2);
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if (flags & 0x40) //URL_Flag
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bytestream2_skip(&gb, bytestream2_get_byte(&gb));
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if (flags & 0x20) //OCRstreamFlag
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bytestream2_skip(&gb, 2);
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}
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} while (bytestream2_get_bytes_left(&gb));
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} else if (avctx->codec_id != AV_CODEC_ID_ALAC) {
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avctx->extradata_size = extradata_size;
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}
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}
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ret = ffat_update_ctx(avctx);
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if (ret < 0)
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return ret;
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#if !TARGET_OS_IPHONE && defined(__MAC_10_9)
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if (at->mode == kAudioCodecBitRateControlMode_Variable && avctx->rc_max_rate) {
|
|
UInt32 max_size = avctx->rc_max_rate * avctx->frame_size / avctx->sample_rate;
|
|
if (max_size)
|
|
AudioConverterSetProperty(at->converter, kAudioCodecPropertyPacketSizeLimitForVBR,
|
|
sizeof(max_size), &max_size);
|
|
}
|
|
#endif
|
|
|
|
ff_af_queue_init(avctx, &at->afq);
|
|
|
|
at->encoding_frame = av_frame_alloc();
|
|
if (!at->encoding_frame)
|
|
return AVERROR(ENOMEM);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static OSStatus ffat_encode_callback(AudioConverterRef converter, UInt32 *nb_packets,
|
|
AudioBufferList *data,
|
|
AudioStreamPacketDescription **packets,
|
|
void *inctx)
|
|
{
|
|
AVCodecContext *avctx = inctx;
|
|
ATDecodeContext *at = avctx->priv_data;
|
|
AVFrame *frame;
|
|
int ret;
|
|
|
|
if (!at->frame_queue.available) {
|
|
if (at->eof) {
|
|
*nb_packets = 0;
|
|
return 0;
|
|
} else {
|
|
*nb_packets = 0;
|
|
return 1;
|
|
}
|
|
}
|
|
|
|
frame = ff_bufqueue_get(&at->frame_queue);
|
|
|
|
data->mNumberBuffers = 1;
|
|
data->mBuffers[0].mNumberChannels = avctx->ch_layout.nb_channels;
|
|
data->mBuffers[0].mDataByteSize = frame->nb_samples *
|
|
av_get_bytes_per_sample(avctx->sample_fmt) *
|
|
avctx->ch_layout.nb_channels;
|
|
data->mBuffers[0].mData = frame->data[0];
|
|
if (*nb_packets > frame->nb_samples)
|
|
*nb_packets = frame->nb_samples;
|
|
|
|
ret = av_frame_replace(at->encoding_frame, frame);
|
|
if (ret < 0) {
|
|
*nb_packets = 0;
|
|
return ret;
|
|
}
|
|
|
|
ff_bufqueue_add(avctx, &at->used_frame_queue, frame);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ffat_encode(AVCodecContext *avctx, AVPacket *avpkt,
|
|
const AVFrame *frame, int *got_packet_ptr)
|
|
{
|
|
ATDecodeContext *at = avctx->priv_data;
|
|
OSStatus ret;
|
|
|
|
AudioBufferList out_buffers = {
|
|
.mNumberBuffers = 1,
|
|
.mBuffers = {
|
|
{
|
|
.mNumberChannels = avctx->ch_layout.nb_channels,
|
|
.mDataByteSize = at->pkt_size,
|
|
}
|
|
}
|
|
};
|
|
AudioStreamPacketDescription out_pkt_desc = {0};
|
|
|
|
if (frame) {
|
|
AVFrame *in_frame;
|
|
|
|
if (ff_bufqueue_is_full(&at->frame_queue)) {
|
|
/*
|
|
* The frame queue is significantly larger than needed in practice,
|
|
* but no clear way to determine the minimum number of samples to
|
|
* get output from AudioConverterFillComplexBuffer().
|
|
*/
|
|
av_log(avctx, AV_LOG_ERROR, "Bug: frame queue is too small.\n");
|
|
return AVERROR_BUG;
|
|
}
|
|
|
|
if ((ret = ff_af_queue_add(&at->afq, frame)) < 0)
|
|
return ret;
|
|
|
|
in_frame = av_frame_clone(frame);
|
|
if (!in_frame)
|
|
return AVERROR(ENOMEM);
|
|
|
|
ff_bufqueue_add(avctx, &at->frame_queue, in_frame);
|
|
} else {
|
|
at->eof = 1;
|
|
}
|
|
|
|
if ((ret = ff_alloc_packet(avctx, avpkt, at->pkt_size)) < 0)
|
|
return ret;
|
|
|
|
|
|
out_buffers.mBuffers[0].mData = avpkt->data;
|
|
|
|
*got_packet_ptr = avctx->frame_size / at->frame_size;
|
|
|
|
ret = AudioConverterFillComplexBuffer(at->converter, ffat_encode_callback, avctx,
|
|
got_packet_ptr, &out_buffers,
|
|
(avctx->frame_size > at->frame_size) ? NULL : &out_pkt_desc);
|
|
|
|
ff_bufqueue_discard_all(&at->used_frame_queue);
|
|
|
|
if ((!ret || ret == 1) && *got_packet_ptr) {
|
|
avpkt->size = out_buffers.mBuffers[0].mDataByteSize;
|
|
ff_af_queue_remove(&at->afq, out_pkt_desc.mVariableFramesInPacket ?
|
|
out_pkt_desc.mVariableFramesInPacket :
|
|
avctx->frame_size,
|
|
&avpkt->pts,
|
|
&avpkt->duration);
|
|
avpkt->flags |= AV_PKT_FLAG_KEY;
|
|
} else if (ret && ret != 1) {
|
|
av_log(avctx, AV_LOG_ERROR, "Encode error: %i\n", ret);
|
|
return AVERROR_EXTERNAL;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold void ffat_encode_flush(AVCodecContext *avctx)
|
|
{
|
|
ATDecodeContext *at = avctx->priv_data;
|
|
AudioConverterReset(at->converter);
|
|
ff_bufqueue_discard_all(&at->frame_queue);
|
|
ff_bufqueue_discard_all(&at->used_frame_queue);
|
|
}
|
|
|
|
static av_cold int ffat_close_encoder(AVCodecContext *avctx)
|
|
{
|
|
ATDecodeContext *at = avctx->priv_data;
|
|
AudioConverterDispose(at->converter);
|
|
ff_bufqueue_discard_all(&at->frame_queue);
|
|
ff_bufqueue_discard_all(&at->used_frame_queue);
|
|
ff_af_queue_close(&at->afq);
|
|
av_frame_free(&at->encoding_frame);
|
|
return 0;
|
|
}
|
|
|
|
static const AVProfile aac_profiles[] = {
|
|
{ AV_PROFILE_AAC_LOW, "LC" },
|
|
{ AV_PROFILE_AAC_HE, "HE-AAC" },
|
|
{ AV_PROFILE_AAC_HE_V2, "HE-AACv2" },
|
|
{ AV_PROFILE_AAC_LD, "LD" },
|
|
{ AV_PROFILE_AAC_ELD, "ELD" },
|
|
{ AV_PROFILE_UNKNOWN },
|
|
};
|
|
|
|
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
|
|
static const AVOption options[] = {
|
|
#if !TARGET_OS_IPHONE
|
|
{"aac_at_mode", "ratecontrol mode", offsetof(ATDecodeContext, mode), AV_OPT_TYPE_INT, {.i64 = -1}, -1, kAudioCodecBitRateControlMode_Variable, AE, .unit = "mode"},
|
|
{"auto", "VBR if global quality is given; CBR otherwise", 0, AV_OPT_TYPE_CONST, {.i64 = -1}, INT_MIN, INT_MAX, AE, .unit = "mode"},
|
|
{"cbr", "constant bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_Constant}, INT_MIN, INT_MAX, AE, .unit = "mode"},
|
|
{"abr", "long-term average bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_LongTermAverage}, INT_MIN, INT_MAX, AE, .unit = "mode"},
|
|
{"cvbr", "constrained variable bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_VariableConstrained}, INT_MIN, INT_MAX, AE, .unit = "mode"},
|
|
{"vbr" , "variable bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_Variable}, INT_MIN, INT_MAX, AE, .unit = "mode"},
|
|
#endif
|
|
{"aac_at_quality", "quality vs speed control", offsetof(ATDecodeContext, quality), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 2, AE},
|
|
{ NULL },
|
|
};
|
|
|
|
#define FFAT_ENC_CLASS(NAME) \
|
|
static const AVClass ffat_##NAME##_enc_class = { \
|
|
.class_name = "at_" #NAME "_enc", \
|
|
.item_name = av_default_item_name, \
|
|
.option = options, \
|
|
.version = LIBAVUTIL_VERSION_INT, \
|
|
};
|
|
|
|
#define FFAT_ENC(NAME, ID, PROFILES, CAPS, CHANNEL_LAYOUTS, CH_LAYOUTS) \
|
|
FFAT_ENC_CLASS(NAME) \
|
|
const FFCodec ff_##NAME##_at_encoder = { \
|
|
.p.name = #NAME "_at", \
|
|
CODEC_LONG_NAME(#NAME " (AudioToolbox)"), \
|
|
.p.type = AVMEDIA_TYPE_AUDIO, \
|
|
.p.id = ID, \
|
|
.priv_data_size = sizeof(ATDecodeContext), \
|
|
.init = ffat_init_encoder, \
|
|
.close = ffat_close_encoder, \
|
|
FF_CODEC_ENCODE_CB(ffat_encode), \
|
|
.flush = ffat_encode_flush, \
|
|
.p.priv_class = &ffat_##NAME##_enc_class, \
|
|
.p.capabilities = AV_CODEC_CAP_DELAY | \
|
|
AV_CODEC_CAP_ENCODER_FLUSH CAPS, \
|
|
.p.profiles = PROFILES, \
|
|
.p.wrapper_name = "at", \
|
|
CODEC_CH_LAYOUTS_ARRAY(CH_LAYOUTS), \
|
|
CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), \
|
|
};
|
|
|
|
static const AVChannelLayout aac_at_ch_layouts[] = {
|
|
AV_CHANNEL_LAYOUT_MONO,
|
|
AV_CHANNEL_LAYOUT_STEREO,
|
|
AV_CHANNEL_LAYOUT_SURROUND,
|
|
AV_CHANNEL_LAYOUT_4POINT0,
|
|
AV_CHANNEL_LAYOUT_5POINT0,
|
|
AV_CHANNEL_LAYOUT_5POINT1,
|
|
AV_CHANNEL_LAYOUT_6POINT0,
|
|
AV_CHANNEL_LAYOUT_6POINT1,
|
|
AV_CHANNEL_LAYOUT_7POINT0,
|
|
AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK,
|
|
AV_CHANNEL_LAYOUT_QUAD,
|
|
AV_CHANNEL_LAYOUT_OCTAGONAL,
|
|
{ 0 },
|
|
};
|
|
|
|
FFAT_ENC(aac, AV_CODEC_ID_AAC, aac_profiles, , aac_at_channel_layouts, aac_at_ch_layouts)
|
|
//FFAT_ENC(adpcm_ima_qt, AV_CODEC_ID_ADPCM_IMA_QT, NULL)
|
|
FFAT_ENC(alac, AV_CODEC_ID_ALAC, NULL, , NULL, NULL)
|
|
FFAT_ENC(ilbc, AV_CODEC_ID_ILBC, NULL, , NULL, NULL)
|
|
FFAT_ENC(pcm_alaw, AV_CODEC_ID_PCM_ALAW, NULL, , NULL, NULL)
|
|
FFAT_ENC(pcm_mulaw, AV_CODEC_ID_PCM_MULAW, NULL, , NULL, NULL)
|