mirror of https://github.com/FFmpeg/FFmpeg.git
Rodger Combs
9 years ago
parent
db1a642cd2
commit
d5d328059e
6 changed files with 389 additions and 2 deletions
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/*
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* Audio Toolbox system codecs |
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* |
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* copyright (c) 2016 Rodger Combs |
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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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#include "config.h" |
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#include "avcodec.h" |
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#include "bytestream.h" |
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#include "internal.h" |
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#include "libavutil/avassert.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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AudioConverterRef converter; |
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AudioStreamPacketDescription pkt_desc; |
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AVPacket in_pkt; |
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AVPacket new_in_pkt; |
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unsigned pkt_size; |
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int64_t last_pts; |
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int eof; |
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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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return kAudioFormatMPEG4AAC; |
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case AV_CODEC_ID_AC3: |
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return kAudioFormatAC3; |
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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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case AV_CODEC_ID_AMR_NB: |
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return kAudioFormatAMR; |
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case AV_CODEC_ID_GSM_MS: |
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return kAudioFormatMicrosoftGSM; |
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case AV_CODEC_ID_ILBC: |
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return kAudioFormatiLBC; |
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case AV_CODEC_ID_MP1: |
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return kAudioFormatMPEGLayer1; |
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case AV_CODEC_ID_MP2: |
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return kAudioFormatMPEGLayer2; |
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case AV_CODEC_ID_MP3: |
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return kAudioFormatMPEGLayer3; |
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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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case AV_CODEC_ID_QDMC: |
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return kAudioFormatQDesign; |
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case AV_CODEC_ID_QDM2: |
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return kAudioFormatQDesign2; |
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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 void ffat_update_ctx(AVCodecContext *avctx) |
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{ |
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ATDecodeContext *at = avctx->priv_data; |
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AudioStreamBasicDescription in_format; |
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UInt32 size = sizeof(in_format); |
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if (!AudioConverterGetProperty(at->converter, |
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kAudioConverterCurrentInputStreamDescription, |
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&size, &in_format)) { |
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avctx->channels = in_format.mChannelsPerFrame; |
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at->pkt_size = in_format.mFramesPerPacket; |
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} |
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if (!at->pkt_size) |
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at->pkt_size = 2048; |
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} |
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static void put_descr(PutByteContext *pb, int tag, unsigned int size) |
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{ |
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int i = 3; |
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bytestream2_put_byte(pb, tag); |
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for (; i > 0; i--) |
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bytestream2_put_byte(pb, (size >> (7 * i)) | 0x80); |
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bytestream2_put_byte(pb, size & 0x7F); |
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} |
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static av_cold int ffat_init_decoder(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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enum AVSampleFormat sample_fmt = (avctx->bits_per_raw_sample == 32) ? |
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AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S16; |
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AudioStreamBasicDescription in_format = { |
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.mSampleRate = avctx->sample_rate ? avctx->sample_rate : 44100, |
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.mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile), |
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.mBytesPerPacket = avctx->block_align, |
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.mChannelsPerFrame = avctx->channels ? avctx->channels : 1, |
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}; |
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AudioStreamBasicDescription out_format = { |
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.mSampleRate = in_format.mSampleRate, |
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.mFormatID = kAudioFormatLinearPCM, |
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.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked, |
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.mFramesPerPacket = 1, |
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.mChannelsPerFrame = in_format.mChannelsPerFrame, |
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.mBitsPerChannel = av_get_bytes_per_sample(sample_fmt) * 8, |
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}; |
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avctx->sample_fmt = sample_fmt; |
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if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT) |
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in_format.mFramesPerPacket = 64; |
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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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return AVERROR_UNKNOWN; |
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} |
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if (avctx->extradata_size) { |
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char *extradata = avctx->extradata; |
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int extradata_size = avctx->extradata_size; |
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if (avctx->codec_id == AV_CODEC_ID_AAC) { |
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PutByteContext pb; |
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extradata_size = 5 + 3 + 5+13 + 5+avctx->extradata_size; |
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if (!(extradata = av_malloc(extradata_size))) |
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return AVERROR(ENOMEM); |
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bytestream2_init_writer(&pb, extradata, extradata_size); |
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// ES descriptor
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put_descr(&pb, 0x03, 3 + 5+13 + 5+avctx->extradata_size); |
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bytestream2_put_be16(&pb, 0); |
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bytestream2_put_byte(&pb, 0x00); // flags (= no flags)
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// DecoderConfig descriptor
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put_descr(&pb, 0x04, 13 + 5+avctx->extradata_size); |
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// Object type indication
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bytestream2_put_byte(&pb, 0x40); |
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bytestream2_put_byte(&pb, 0x15); // flags (= Audiostream)
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bytestream2_put_be24(&pb, 0); // Buffersize DB
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bytestream2_put_be32(&pb, 0); // maxbitrate
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bytestream2_put_be32(&pb, 0); // avgbitrate
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// DecoderSpecific info descriptor
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put_descr(&pb, 0x05, avctx->extradata_size); |
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bytestream2_put_buffer(&pb, avctx->extradata, avctx->extradata_size); |
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} |
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status = AudioConverterSetProperty(at->converter, |
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kAudioConverterDecompressionMagicCookie, |
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extradata_size, extradata); |
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if (status != 0) |
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av_log(avctx, AV_LOG_WARNING, "AudioToolbox cookie error: %i\n", (int)status); |
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} |
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ffat_update_ctx(avctx); |
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at->last_pts = AV_NOPTS_VALUE; |
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return 0; |
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} |
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static OSStatus ffat_decode_callback(AudioConverterRef converter, UInt32 *nb_packets, |
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AudioBufferList *data, |
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AudioStreamPacketDescription **packets, |
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void *inctx) |
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{ |
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AVCodecContext *avctx = inctx; |
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ATDecodeContext *at = avctx->priv_data; |
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if (at->eof) { |
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*nb_packets = 0; |
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if (packets) { |
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*packets = &at->pkt_desc; |
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at->pkt_desc.mDataByteSize = 0; |
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} |
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return 0; |
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} |
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av_packet_move_ref(&at->in_pkt, &at->new_in_pkt); |
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at->new_in_pkt.data = 0; |
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at->new_in_pkt.size = 0; |
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if (!at->in_pkt.data) { |
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*nb_packets = 0; |
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return 1; |
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} |
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data->mNumberBuffers = 1; |
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data->mBuffers[0].mNumberChannels = 0; |
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data->mBuffers[0].mDataByteSize = at->in_pkt.size; |
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data->mBuffers[0].mData = at->in_pkt.data; |
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*nb_packets = 1; |
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if (packets) { |
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*packets = &at->pkt_desc; |
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at->pkt_desc.mDataByteSize = at->in_pkt.size; |
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} |
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return 0; |
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} |
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static int ffat_decode(AVCodecContext *avctx, void *data, |
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int *got_frame_ptr, AVPacket *avpkt) |
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{ |
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ATDecodeContext *at = avctx->priv_data; |
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AVFrame *frame = data; |
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OSStatus ret; |
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AudioBufferList out_buffers = { |
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.mNumberBuffers = 1, |
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.mBuffers = { |
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{ |
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.mNumberChannels = avctx->channels, |
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.mDataByteSize = av_get_bytes_per_sample(avctx->sample_fmt) * at->pkt_size * avctx->channels, |
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} |
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} |
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}; |
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av_packet_unref(&at->new_in_pkt); |
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if (avpkt->size) { |
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if ((ret = av_packet_ref(&at->new_in_pkt, avpkt)) < 0) |
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return ret; |
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} else { |
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at->eof = 1; |
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} |
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frame->sample_rate = avctx->sample_rate; |
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frame->nb_samples = at->pkt_size; |
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ff_get_buffer(avctx, frame, 0); |
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out_buffers.mBuffers[0].mData = frame->data[0]; |
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ret = AudioConverterFillComplexBuffer(at->converter, ffat_decode_callback, avctx, |
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&frame->nb_samples, &out_buffers, NULL); |
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if ((!ret || ret == 1) && frame->nb_samples) { |
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*got_frame_ptr = 1; |
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if (at->last_pts != AV_NOPTS_VALUE) { |
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frame->pts = at->last_pts; |
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at->last_pts = avpkt->pts; |
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} |
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} else if (ret && ret != 1) { |
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av_log(avctx, AV_LOG_WARNING, "Decode error: %i\n", ret); |
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} else { |
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at->last_pts = avpkt->pts; |
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} |
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return avpkt->size; |
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} |
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static av_cold void ffat_decode_flush(AVCodecContext *avctx) |
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{ |
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ATDecodeContext *at = avctx->priv_data; |
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AudioConverterReset(at->converter); |
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av_packet_unref(&at->new_in_pkt); |
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av_packet_unref(&at->in_pkt); |
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} |
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static av_cold int ffat_close_decoder(AVCodecContext *avctx) |
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{ |
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ATDecodeContext *at = avctx->priv_data; |
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AudioConverterDispose(at->converter); |
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av_packet_unref(&at->new_in_pkt); |
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av_packet_unref(&at->in_pkt); |
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return 0; |
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} |
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#define FFAT_DEC_CLASS(NAME) \ |
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static const AVClass ffat_##NAME##_dec_class = { \
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.class_name = "at_" #NAME "_dec", \
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.version = LIBAVUTIL_VERSION_INT, \
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}; |
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#define FFAT_DEC(NAME, ID) \ |
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FFAT_DEC_CLASS(NAME) \
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AVCodec ff_##NAME##_at_decoder = { \
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.name = #NAME "_at", \
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.long_name = NULL_IF_CONFIG_SMALL(#NAME " (AudioToolbox)"), \
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.type = AVMEDIA_TYPE_AUDIO, \
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.id = ID, \
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.priv_data_size = sizeof(ATDecodeContext), \
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.init = ffat_init_decoder, \
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.close = ffat_close_decoder, \
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.decode = ffat_decode, \
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.flush = ffat_decode_flush, \
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.priv_class = &ffat_##NAME##_dec_class, \
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.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, \
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
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}; |
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FFAT_DEC(aac, AV_CODEC_ID_AAC) |
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FFAT_DEC(ac3, AV_CODEC_ID_AC3) |
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FFAT_DEC(adpcm_ima_qt, AV_CODEC_ID_ADPCM_IMA_QT) |
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FFAT_DEC(alac, AV_CODEC_ID_ALAC) |
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FFAT_DEC(amr_nb, AV_CODEC_ID_AMR_NB) |
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FFAT_DEC(gsm_ms, AV_CODEC_ID_GSM_MS) |
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FFAT_DEC(ilbc, AV_CODEC_ID_ILBC) |
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FFAT_DEC(mp1, AV_CODEC_ID_MP1) |
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FFAT_DEC(mp2, AV_CODEC_ID_MP2) |
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FFAT_DEC(mp3, AV_CODEC_ID_MP3) |
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FFAT_DEC(pcm_alaw, AV_CODEC_ID_PCM_ALAW) |
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FFAT_DEC(pcm_mulaw, AV_CODEC_ID_PCM_MULAW) |
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FFAT_DEC(qdmc, AV_CODEC_ID_QDMC) |
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FFAT_DEC(qdm2, AV_CODEC_ID_QDM2) |
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