/* * Audio Toolbox system codecs * * copyright (c) 2016 rcombs * * 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 */ #include <AudioToolbox/AudioToolbox.h> #define FF_BUFQUEUE_SIZE 256#include "libavfilter/bufferqueue.h" #include "config.h"#include "audio_frame_queue.h"#include "avcodec.h"#include "bytestream.h"#include "codec_internal.h"#include "encode.h"#include "internal.h"#include "libavformat/isom.h"#include "libavutil/avassert.h"#include "libavutil/channel_layout.h"#include "libavutil/mem.h"#include "libavutil/opt.h"#include "libavutil/log.h" typedef struct ATDecodeContext { AVClass *av_class; int mode; int quality; AudioConverterRef converter; struct FFBufQueue frame_queue; struct FFBufQueue used_frame_queue; unsigned pkt_size; AudioFrameQueue afq; int eof; int frame_size; AVFrame* encoding_frame;} ATDecodeContext; static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile){ switch (codec) { case AV_CODEC_ID_AAC: switch (profile) { case AV_PROFILE_AAC_LOW: default: return kAudioFormatMPEG4AAC; case AV_PROFILE_AAC_HE: return kAudioFormatMPEG4AAC_HE; case AV_PROFILE_AAC_HE_V2: return kAudioFormatMPEG4AAC_HE_V2; case AV_PROFILE_AAC_LD: return kAudioFormatMPEG4AAC_LD;#if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060 case AV_PROFILE_AAC_ELD: return kAudioFormatMPEG4AAC_ELD;#endif } case AV_CODEC_ID_ADPCM_IMA_QT: return kAudioFormatAppleIMA4; case AV_CODEC_ID_ALAC: return kAudioFormatAppleLossless;#if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060 case AV_CODEC_ID_ILBC: return kAudioFormatiLBC;#endif case AV_CODEC_ID_PCM_ALAW: return kAudioFormatALaw; case AV_CODEC_ID_PCM_MULAW: return kAudioFormatULaw; default: av_assert0(!"Invalid codec ID!"); return 0; }} static int ffat_update_ctx(AVCodecContext *avctx){ ATDecodeContext *at = avctx->priv_data; UInt32 size = sizeof(unsigned); AudioConverterPrimeInfo prime_info; AudioStreamBasicDescription out_format; AudioConverterGetProperty(at->converter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &at->pkt_size); if (at->pkt_size <= 0) at->pkt_size = 1024 * 50; size = sizeof(prime_info); if (!AudioConverterGetProperty(at->converter, kAudioConverterPrimeInfo, &size, &prime_info)) { avctx->initial_padding = prime_info.leadingFrames; } size = sizeof(out_format); if (!AudioConverterGetProperty(at->converter, kAudioConverterCurrentOutputStreamDescription, &size, &out_format)) { if (out_format.mFramesPerPacket) { avctx->frame_size = out_format.mFramesPerPacket; } else { /* The doc on mFramesPerPacket says: * For formats with a variable number of frames per packet, such as * Ogg Vorbis, set this field to 0. * Looks like it means for decoding. There is no known case that * mFramesPerPacket is zero for encoding. Use a default value for safety. */ avctx->frame_size = 1024; av_log(avctx, AV_LOG_WARNING, "Missing mFramesPerPacket\n"); } if (out_format.mBytesPerPacket && avctx->codec_id == AV_CODEC_ID_ILBC) avctx->block_align = out_format.mBytesPerPacket; } else { av_log(avctx, AV_LOG_ERROR, "Get OutputStreamDescription failed\n"); return AVERROR_EXTERNAL; } at->frame_size = avctx->frame_size; if (avctx->codec_id == AV_CODEC_ID_PCM_MULAW || avctx->codec_id == AV_CODEC_ID_PCM_ALAW) { at->pkt_size *= 1024; avctx->frame_size *= 1024; } return 0;} static int read_descr(GetByteContext *gb, int *tag){ int len = 0; int count = 4; *tag = bytestream2_get_byte(gb); while (count--) { int c = bytestream2_get_byte(gb); len = (len << 7) | (c & 0x7f); if (!(c & 0x80)) break; } return len;} static int get_ilbc_mode(AVCodecContext *avctx){ if (avctx->block_align == 38) return 20; else if (avctx->block_align == 50) return 30; else if (avctx->bit_rate > 0) return avctx->bit_rate <= 14000 ? 30 : 20; else return 30;} static av_cold int get_channel_label(int channel){ uint64_t map = 1 << channel; if (map <= AV_CH_LOW_FREQUENCY) return channel + 1; else if (map <= AV_CH_BACK_RIGHT) return channel + 29; else if (map <= AV_CH_BACK_CENTER) return channel - 1; else if (map <= AV_CH_SIDE_RIGHT) return channel - 4; else if (map <= AV_CH_TOP_BACK_RIGHT) return channel + 1; else if (map <= AV_CH_STEREO_RIGHT) return -1; else if (map <= AV_CH_WIDE_RIGHT) return channel + 4; else if (map <= AV_CH_SURROUND_DIRECT_RIGHT) return channel - 23; else if (map == AV_CH_LOW_FREQUENCY_2) return kAudioChannelLabel_LFE2; else return -1;} static int remap_layout(AudioChannelLayout *layout, const AVChannelLayout *in_layout){ int i; layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions; layout->mNumberChannelDescriptions = in_layout->nb_channels; for (i = 0; i < in_layout->nb_channels; i++) { int c, label; c = av_channel_layout_channel_from_index(in_layout, i); if (c < 0 || c >= 64) return AVERROR(EINVAL); label = get_channel_label(c); layout->mChannelDescriptions[i].mChannelLabel = label; if (label < 0) return AVERROR(EINVAL); c++; } return 0;} static int get_aac_tag(const AVChannelLayout *in_layout){ static const struct { AVChannelLayout chl; int tag; } map[] = { { AV_CHANNEL_LAYOUT_MONO, kAudioChannelLayoutTag_Mono }, { AV_CHANNEL_LAYOUT_STEREO, kAudioChannelLayoutTag_Stereo }, { AV_CHANNEL_LAYOUT_QUAD, kAudioChannelLayoutTag_AAC_Quadraphonic }, { AV_CHANNEL_LAYOUT_OCTAGONAL, kAudioChannelLayoutTag_AAC_Octagonal }, { AV_CHANNEL_LAYOUT_SURROUND, kAudioChannelLayoutTag_AAC_3_0 }, { AV_CHANNEL_LAYOUT_4POINT0, kAudioChannelLayoutTag_AAC_4_0 }, { AV_CHANNEL_LAYOUT_5POINT0, kAudioChannelLayoutTag_AAC_5_0 }, { AV_CHANNEL_LAYOUT_5POINT1, kAudioChannelLayoutTag_AAC_5_1 }, { AV_CHANNEL_LAYOUT_6POINT0, kAudioChannelLayoutTag_AAC_6_0 }, { AV_CHANNEL_LAYOUT_6POINT1, kAudioChannelLayoutTag_AAC_6_1 }, { AV_CHANNEL_LAYOUT_7POINT0, kAudioChannelLayoutTag_AAC_7_0 }, { AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK, kAudioChannelLayoutTag_AAC_7_1 }, { AV_CHANNEL_LAYOUT_7POINT1, kAudioChannelLayoutTag_MPEG_7_1_C }, }; int i; for (i = 0; i < FF_ARRAY_ELEMS(map); i++) if (!av_channel_layout_compare(in_layout, &map[i].chl)) return map[i].tag; return 0;} static av_cold int ffat_init_encoder(AVCodecContext *avctx){ ATDecodeContext *at = avctx->priv_data; OSStatus status; int ret; AudioStreamBasicDescription in_format = { .mSampleRate = avctx->sample_rate, .mFormatID = kAudioFormatLinearPCM, .mFormatFlags = ((avctx->sample_fmt == AV_SAMPLE_FMT_FLT || avctx->sample_fmt == AV_SAMPLE_FMT_DBL) ? kAudioFormatFlagIsFloat : avctx->sample_fmt == AV_SAMPLE_FMT_U8 ? 0 : kAudioFormatFlagIsSignedInteger) | kAudioFormatFlagIsPacked, .mBytesPerPacket = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->ch_layout.nb_channels, .mFramesPerPacket = 1, .mBytesPerFrame = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->ch_layout.nb_channels, .mChannelsPerFrame = avctx->ch_layout.nb_channels, .mBitsPerChannel = av_get_bytes_per_sample(avctx->sample_fmt) * 8, }; AudioStreamBasicDescription out_format = { .mSampleRate = avctx->sample_rate, .mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile), .mChannelsPerFrame = in_format.mChannelsPerFrame, }; UInt32 layout_size = sizeof(AudioChannelLayout) + sizeof(AudioChannelDescription) * avctx->ch_layout.nb_channels; AudioChannelLayout *channel_layout = av_malloc(layout_size); if (!channel_layout) return AVERROR(ENOMEM); if (avctx->codec_id == AV_CODEC_ID_ILBC) { int mode = get_ilbc_mode(avctx); out_format.mFramesPerPacket = 8000 * mode / 1000; out_format.mBytesPerPacket = (mode == 20 ? 38 : 50); } status = AudioConverterNew(&in_format, &out_format, &at->converter); if (status != 0) { av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status); av_free(channel_layout); return AVERROR_UNKNOWN; } if (avctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) av_channel_layout_default(&avctx->ch_layout, avctx->ch_layout.nb_channels); if ((status = remap_layout(channel_layout, &avctx->ch_layout)) < 0) { av_log(avctx, AV_LOG_ERROR, "Invalid channel layout\n"); av_free(channel_layout); return status; } if (AudioConverterSetProperty(at->converter, kAudioConverterInputChannelLayout, layout_size, channel_layout)) { av_log(avctx, AV_LOG_ERROR, "Unsupported input channel layout\n"); av_free(channel_layout); return AVERROR(EINVAL); } if (avctx->codec_id == AV_CODEC_ID_AAC) { int tag = get_aac_tag(&avctx->ch_layout); if (tag) { channel_layout->mChannelLayoutTag = tag; channel_layout->mNumberChannelDescriptions = 0; } } if (AudioConverterSetProperty(at->converter, kAudioConverterOutputChannelLayout, layout_size, channel_layout)) { av_log(avctx, AV_LOG_ERROR, "Unsupported output channel layout\n"); av_free(channel_layout); return AVERROR(EINVAL); } av_free(channel_layout); if (avctx->bits_per_raw_sample) AudioConverterSetProperty(at->converter, kAudioConverterPropertyBitDepthHint, sizeof(avctx->bits_per_raw_sample), &avctx->bits_per_raw_sample); #if !TARGET_OS_IPHONE if (at->mode == -1) at->mode = (avctx->flags & AV_CODEC_FLAG_QSCALE) ? kAudioCodecBitRateControlMode_Variable : kAudioCodecBitRateControlMode_Constant; AudioConverterSetProperty(at->converter, kAudioCodecPropertyBitRateControlMode, sizeof(at->mode), &at->mode); if (at->mode == kAudioCodecBitRateControlMode_Variable) { int q = avctx->global_quality / FF_QP2LAMBDA; if (q < 0 || q > 14) { av_log(avctx, AV_LOG_WARNING, "VBR quality %d out of range, should be 0-14\n", q); q = av_clip(q, 0, 14); } q = 127 - q * 9; AudioConverterSetProperty(at->converter, kAudioCodecPropertySoundQualityForVBR, sizeof(q), &q); } else#endif if (avctx->bit_rate > 0) { UInt32 rate = avctx->bit_rate; UInt32 size; status = AudioConverterGetPropertyInfo(at->converter, kAudioConverterApplicableEncodeBitRates, &size, NULL); if (!status && size) { UInt32 new_rate = rate; int count; int i; AudioValueRange *ranges = av_malloc(size); if (!ranges) return AVERROR(ENOMEM); AudioConverterGetProperty(at->converter, kAudioConverterApplicableEncodeBitRates, &size, ranges); count = size / sizeof(AudioValueRange); for (i = 0; i < count; i++) { AudioValueRange *range = &ranges[i]; if (rate >= range->mMinimum && rate <= range->mMaximum) { new_rate = rate; break; } else if (rate > range->mMaximum) { new_rate = range->mMaximum; } else { new_rate = range->mMinimum; break; } } if (new_rate != rate) { av_log(avctx, AV_LOG_WARNING, "Bitrate %u not allowed; changing to %u\n", rate, new_rate); rate = new_rate; } av_free(ranges); } AudioConverterSetProperty(at->converter, kAudioConverterEncodeBitRate, sizeof(rate), &rate); } at->quality = 96 - at->quality * 32; AudioConverterSetProperty(at->converter, kAudioConverterCodecQuality, sizeof(at->quality), &at->quality); if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterCompressionMagicCookie, &avctx->extradata_size, NULL) && avctx->extradata_size) { int extradata_size = avctx->extradata_size; uint8_t *extradata; if (!(avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE))) return AVERROR(ENOMEM); if (avctx->codec_id == AV_CODEC_ID_ALAC) { avctx->extradata_size = 0x24; AV_WB32(avctx->extradata, 0x24); AV_WB32(avctx->extradata + 4, MKBETAG('a','l','a','c')); extradata = avctx->extradata + 12; avctx->extradata_size = 0x24; } else { extradata = avctx->extradata; } status = AudioConverterGetProperty(at->converter, kAudioConverterCompressionMagicCookie, &extradata_size, extradata); if (status != 0) { av_log(avctx, AV_LOG_ERROR, "AudioToolbox cookie error: %i\n", (int)status); return AVERROR_UNKNOWN; } else if (avctx->codec_id == AV_CODEC_ID_AAC) { GetByteContext gb; int tag, len; bytestream2_init(&gb, extradata, extradata_size); do { len = read_descr(&gb, &tag); if (tag == MP4DecConfigDescrTag) { bytestream2_skip(&gb, 13); len = read_descr(&gb, &tag); if (tag == MP4DecSpecificDescrTag) { len = FFMIN(gb.buffer_end - gb.buffer, len); memmove(extradata, gb.buffer, len); avctx->extradata_size = len; break; } } else if (tag == MP4ESDescrTag) { int flags; bytestream2_skip(&gb, 2); flags = bytestream2_get_byte(&gb); if (flags & 0x80) //streamDependenceFlag bytestream2_skip(&gb, 2); if (flags & 0x40) //URL_Flag bytestream2_skip(&gb, bytestream2_get_byte(&gb)); if (flags & 0x20) //OCRstreamFlag bytestream2_skip(&gb, 2); } } while (bytestream2_get_bytes_left(&gb)); } else if (avctx->codec_id != AV_CODEC_ID_ALAC) { avctx->extradata_size = extradata_size; } } ret = ffat_update_ctx(avctx); if (ret < 0) return ret; #if !TARGET_OS_IPHONE && defined(__MAC_10_9) 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;Lines 1–500 of 683. 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/* * Audio Toolbox system codecs * * copyright (c) 2016 rcombs * * 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 */ #include <AudioToolbox/AudioToolbox.h> #define FF_BUFQUEUE_SIZE 256 #include "libavfilter/bufferqueue.h" #include "config.h" #include "audio_frame_queue.h" #include "avcodec.h" #include "bytestream.h" #include "codec_internal.h" #include "encode.h" #include "internal.h" #include "libavformat/isom.h" #include "libavutil/avassert.h" #include "libavutil/channel_layout.h" #include "libavutil/mem.h" #include "libavutil/opt.h" #include "libavutil/log.h" typedef struct ATDecodeContext { AVClass *av_class; int mode; int quality; AudioConverterRef converter; struct FFBufQueue frame_queue; struct FFBufQueue used_frame_queue; unsigned pkt_size; AudioFrameQueue afq; int eof; int frame_size; AVFrame* encoding_frame; } ATDecodeContext; static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile) { switch (codec) { case AV_CODEC_ID_AAC: switch (profile) { case AV_PROFILE_AAC_LOW: default: return kAudioFormatMPEG4AAC; case AV_PROFILE_AAC_HE: return kAudioFormatMPEG4AAC_HE; case AV_PROFILE_AAC_HE_V2: return kAudioFormatMPEG4AAC_HE_V2; case AV_PROFILE_AAC_LD: return kAudioFormatMPEG4AAC_LD; #if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060 case AV_PROFILE_AAC_ELD: return kAudioFormatMPEG4AAC_ELD; #endif } case AV_CODEC_ID_ADPCM_IMA_QT: return kAudioFormatAppleIMA4; case AV_CODEC_ID_ALAC: return kAudioFormatAppleLossless; #if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060 case AV_CODEC_ID_ILBC: return kAudioFormatiLBC; #endif case AV_CODEC_ID_PCM_ALAW: return kAudioFormatALaw; case AV_CODEC_ID_PCM_MULAW: return kAudioFormatULaw; default: av_assert0(!"Invalid codec ID!"); return 0; } } static int ffat_update_ctx(AVCodecContext *avctx) { ATDecodeContext *at = avctx->priv_data; UInt32 size = sizeof(unsigned); AudioConverterPrimeInfo prime_info; AudioStreamBasicDescription out_format; AudioConverterGetProperty(at->converter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &at->pkt_size); if (at->pkt_size <= 0) at->pkt_size = 1024 * 50; size = sizeof(prime_info); if (!AudioConverterGetProperty(at->converter, kAudioConverterPrimeInfo, &size, &prime_info)) { avctx->initial_padding = prime_info.leadingFrames; } size = sizeof(out_format); if (!AudioConverterGetProperty(at->converter, kAudioConverterCurrentOutputStreamDescription, &size, &out_format)) { if (out_format.mFramesPerPacket) { avctx->frame_size = out_format.mFramesPerPacket; } else { /* The doc on mFramesPerPacket says: * For formats with a variable number of frames per packet, such as * Ogg Vorbis, set this field to 0. * Looks like it means for decoding. There is no known case that * mFramesPerPacket is zero for encoding. Use a default value for safety. */ avctx->frame_size = 1024; av_log(avctx, AV_LOG_WARNING, "Missing mFramesPerPacket\n"); } if (out_format.mBytesPerPacket && avctx->codec_id == AV_CODEC_ID_ILBC) avctx->block_align = out_format.mBytesPerPacket; } else { av_log(avctx, AV_LOG_ERROR, "Get OutputStreamDescription failed\n"); return AVERROR_EXTERNAL; } at->frame_size = avctx->frame_size; if (avctx->codec_id == AV_CODEC_ID_PCM_MULAW || avctx->codec_id == AV_CODEC_ID_PCM_ALAW) { at->pkt_size *= 1024; avctx->frame_size *= 1024; } return 0; } static int read_descr(GetByteContext *gb, int *tag) { int len = 0; int count = 4; *tag = bytestream2_get_byte(gb); while (count--) { int c = bytestream2_get_byte(gb); len = (len << 7) | (c & 0x7f); if (!(c & 0x80)) break; } return len; } static int get_ilbc_mode(AVCodecContext *avctx) { if (avctx->block_align == 38) return 20; else if (avctx->block_align == 50) return 30; else if (avctx->bit_rate > 0) return avctx->bit_rate <= 14000 ? 30 : 20; else return 30; } static av_cold int get_channel_label(int channel) { uint64_t map = 1 << channel; if (map <= AV_CH_LOW_FREQUENCY) return channel + 1; else if (map <= AV_CH_BACK_RIGHT) return channel + 29; else if (map <= AV_CH_BACK_CENTER) return channel - 1; else if (map <= AV_CH_SIDE_RIGHT) return channel - 4; else if (map <= AV_CH_TOP_BACK_RIGHT) return channel + 1; else if (map <= AV_CH_STEREO_RIGHT) return -1; else if (map <= AV_CH_WIDE_RIGHT) return channel + 4; else if (map <= AV_CH_SURROUND_DIRECT_RIGHT) return channel - 23; else if (map == AV_CH_LOW_FREQUENCY_2) return kAudioChannelLabel_LFE2; else return -1; } static int remap_layout(AudioChannelLayout *layout, const AVChannelLayout *in_layout) { int i; layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions; layout->mNumberChannelDescriptions = in_layout->nb_channels; for (i = 0; i < in_layout->nb_channels; i++) { int c, label; c = av_channel_layout_channel_from_index(in_layout, i); if (c < 0 || c >= 64) return AVERROR(EINVAL); label = get_channel_label(c); layout->mChannelDescriptions[i].mChannelLabel = label; if (label < 0) return AVERROR(EINVAL); c++; } return 0; } static int get_aac_tag(const AVChannelLayout *in_layout) { static const struct { AVChannelLayout chl; int tag; } map[] = { { AV_CHANNEL_LAYOUT_MONO, kAudioChannelLayoutTag_Mono }, { AV_CHANNEL_LAYOUT_STEREO, kAudioChannelLayoutTag_Stereo }, { AV_CHANNEL_LAYOUT_QUAD, kAudioChannelLayoutTag_AAC_Quadraphonic }, { AV_CHANNEL_LAYOUT_OCTAGONAL, kAudioChannelLayoutTag_AAC_Octagonal }, { AV_CHANNEL_LAYOUT_SURROUND, kAudioChannelLayoutTag_AAC_3_0 }, { AV_CHANNEL_LAYOUT_4POINT0, kAudioChannelLayoutTag_AAC_4_0 }, { AV_CHANNEL_LAYOUT_5POINT0, kAudioChannelLayoutTag_AAC_5_0 }, { AV_CHANNEL_LAYOUT_5POINT1, kAudioChannelLayoutTag_AAC_5_1 }, { AV_CHANNEL_LAYOUT_6POINT0, kAudioChannelLayoutTag_AAC_6_0 }, { AV_CHANNEL_LAYOUT_6POINT1, kAudioChannelLayoutTag_AAC_6_1 }, { AV_CHANNEL_LAYOUT_7POINT0, kAudioChannelLayoutTag_AAC_7_0 }, { AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK, kAudioChannelLayoutTag_AAC_7_1 }, { AV_CHANNEL_LAYOUT_7POINT1, kAudioChannelLayoutTag_MPEG_7_1_C }, }; int i; for (i = 0; i < FF_ARRAY_ELEMS(map); i++) if (!av_channel_layout_compare(in_layout, &map[i].chl)) return map[i].tag; return 0; } static av_cold int ffat_init_encoder(AVCodecContext *avctx) { ATDecodeContext *at = avctx->priv_data; OSStatus status; int ret; AudioStreamBasicDescription in_format = { .mSampleRate = avctx->sample_rate, .mFormatID = kAudioFormatLinearPCM, .mFormatFlags = ((avctx->sample_fmt == AV_SAMPLE_FMT_FLT || avctx->sample_fmt == AV_SAMPLE_FMT_DBL) ? kAudioFormatFlagIsFloat : avctx->sample_fmt == AV_SAMPLE_FMT_U8 ? 0 : kAudioFormatFlagIsSignedInteger) | kAudioFormatFlagIsPacked, .mBytesPerPacket = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->ch_layout.nb_channels, .mFramesPerPacket = 1, .mBytesPerFrame = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->ch_layout.nb_channels, .mChannelsPerFrame = avctx->ch_layout.nb_channels, .mBitsPerChannel = av_get_bytes_per_sample(avctx->sample_fmt) * 8, }; AudioStreamBasicDescription out_format = { .mSampleRate = avctx->sample_rate, .mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile), .mChannelsPerFrame = in_format.mChannelsPerFrame, }; UInt32 layout_size = sizeof(AudioChannelLayout) + sizeof(AudioChannelDescription) * avctx->ch_layout.nb_channels; AudioChannelLayout *channel_layout = av_malloc(layout_size); if (!channel_layout) return AVERROR(ENOMEM); if (avctx->codec_id == AV_CODEC_ID_ILBC) { int mode = get_ilbc_mode(avctx); out_format.mFramesPerPacket = 8000 * mode / 1000; out_format.mBytesPerPacket = (mode == 20 ? 38 : 50); } status = AudioConverterNew(&in_format, &out_format, &at->converter); if (status != 0) { av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status); av_free(channel_layout); return AVERROR_UNKNOWN; } if (avctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) av_channel_layout_default(&avctx->ch_layout, avctx->ch_layout.nb_channels); if ((status = remap_layout(channel_layout, &avctx->ch_layout)) < 0) { av_log(avctx, AV_LOG_ERROR, "Invalid channel layout\n"); av_free(channel_layout); return status; } if (AudioConverterSetProperty(at->converter, kAudioConverterInputChannelLayout, layout_size, channel_layout)) { av_log(avctx, AV_LOG_ERROR, "Unsupported input channel layout\n"); av_free(channel_layout); return AVERROR(EINVAL); } if (avctx->codec_id == AV_CODEC_ID_AAC) { int tag = get_aac_tag(&avctx->ch_layout); if (tag) { channel_layout->mChannelLayoutTag = tag; channel_layout->mNumberChannelDescriptions = 0; } } if (AudioConverterSetProperty(at->converter, kAudioConverterOutputChannelLayout, layout_size, channel_layout)) { av_log(avctx, AV_LOG_ERROR, "Unsupported output channel layout\n"); av_free(channel_layout); return AVERROR(EINVAL); } av_free(channel_layout); if (avctx->bits_per_raw_sample) AudioConverterSetProperty(at->converter, kAudioConverterPropertyBitDepthHint, sizeof(avctx->bits_per_raw_sample), &avctx->bits_per_raw_sample); #if !TARGET_OS_IPHONE if (at->mode == -1) at->mode = (avctx->flags & AV_CODEC_FLAG_QSCALE) ? kAudioCodecBitRateControlMode_Variable : kAudioCodecBitRateControlMode_Constant; AudioConverterSetProperty(at->converter, kAudioCodecPropertyBitRateControlMode, sizeof(at->mode), &at->mode); if (at->mode == kAudioCodecBitRateControlMode_Variable) { int q = avctx->global_quality / FF_QP2LAMBDA; if (q < 0 || q > 14) { av_log(avctx, AV_LOG_WARNING, "VBR quality %d out of range, should be 0-14\n", q); q = av_clip(q, 0, 14); } q = 127 - q * 9; AudioConverterSetProperty(at->converter, kAudioCodecPropertySoundQualityForVBR, sizeof(q), &q); } else #endif if (avctx->bit_rate > 0) { UInt32 rate = avctx->bit_rate; UInt32 size; status = AudioConverterGetPropertyInfo(at->converter, kAudioConverterApplicableEncodeBitRates, &size, NULL); if (!status && size) { UInt32 new_rate = rate; int count; int i; AudioValueRange *ranges = av_malloc(size); if (!ranges) return AVERROR(ENOMEM); AudioConverterGetProperty(at->converter, kAudioConverterApplicableEncodeBitRates, &size, ranges); count = size / sizeof(AudioValueRange); for (i = 0; i < count; i++) { AudioValueRange *range = &ranges[i]; if (rate >= range->mMinimum && rate <= range->mMaximum) { new_rate = rate; break; } else if (rate > range->mMaximum) { new_rate = range->mMaximum; } else { new_rate = range->mMinimum; break; } } if (new_rate != rate) { av_log(avctx, AV_LOG_WARNING, "Bitrate %u not allowed; changing to %u\n", rate, new_rate); rate = new_rate; } av_free(ranges); } AudioConverterSetProperty(at->converter, kAudioConverterEncodeBitRate, sizeof(rate), &rate); } at->quality = 96 - at->quality * 32; AudioConverterSetProperty(at->converter, kAudioConverterCodecQuality, sizeof(at->quality), &at->quality); if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterCompressionMagicCookie, &avctx->extradata_size, NULL) && avctx->extradata_size) { int extradata_size = avctx->extradata_size; uint8_t *extradata; if (!(avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE))) return AVERROR(ENOMEM); if (avctx->codec_id == AV_CODEC_ID_ALAC) { avctx->extradata_size = 0x24; AV_WB32(avctx->extradata, 0x24); AV_WB32(avctx->extradata + 4, MKBETAG('a','l','a','c')); extradata = avctx->extradata + 12; avctx->extradata_size = 0x24; } else { extradata = avctx->extradata; } status = AudioConverterGetProperty(at->converter, kAudioConverterCompressionMagicCookie, &extradata_size, extradata); if (status != 0) { av_log(avctx, AV_LOG_ERROR, "AudioToolbox cookie error: %i\n", (int)status); return AVERROR_UNKNOWN; } else if (avctx->codec_id == AV_CODEC_ID_AAC) { GetByteContext gb; int tag, len; bytestream2_init(&gb, extradata, extradata_size); do { len = read_descr(&gb, &tag); if (tag == MP4DecConfigDescrTag) { bytestream2_skip(&gb, 13); len = read_descr(&gb, &tag); if (tag == MP4DecSpecificDescrTag) { len = FFMIN(gb.buffer_end - gb.buffer, len); memmove(extradata, gb.buffer, len); avctx->extradata_size = len; break; } } else if (tag == MP4ESDescrTag) { int flags; bytestream2_skip(&gb, 2); flags = bytestream2_get_byte(&gb); if (flags & 0x80) //streamDependenceFlag bytestream2_skip(&gb, 2); if (flags & 0x40) //URL_Flag bytestream2_skip(&gb, bytestream2_get_byte(&gb)); if (flags & 0x20) //OCRstreamFlag bytestream2_skip(&gb, 2); } } while (bytestream2_get_bytes_left(&gb)); } else if (avctx->codec_id != AV_CODEC_ID_ALAC) { avctx->extradata_size = extradata_size; } } ret = ffat_update_ctx(avctx); if (ret < 0) return ret; #if !TARGET_OS_IPHONE && defined(__MAC_10_9) 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;