From 1b9e90ee80be92c8e14349a8fc74b4560a46e648 Mon Sep 17 00:00:00 2001 From: Rodger Combs Date: Thu, 24 Mar 2016 00:49:51 -0500 Subject: lavc/audiotoolboxdec: fix a number of config and timestamp issues - ADTS-formatted AAC didn't work - Channel layouts were never exported - Channel mappings were incorrect beyond stereo - Channel counts weren't updated after packets were decoded - Timestamps were exported incorrectly --- libavcodec/audiotoolboxdec.c | 286 +++++++++++++++++++++++++++++++++---------- 1 file changed, 221 insertions(+), 65 deletions(-) (limited to 'libavcodec') diff --git a/libavcodec/audiotoolboxdec.c b/libavcodec/audiotoolboxdec.c index 1fa6f16e70..4ff46eae3e 100644 --- a/libavcodec/audiotoolboxdec.c +++ b/libavcodec/audiotoolboxdec.c @@ -38,8 +38,9 @@ typedef struct ATDecodeContext { AVPacket in_pkt; AVPacket new_in_pkt; AVBitStreamFilterContext *bsf; + char *decoded_data; + int channel_map[64]; - unsigned pkt_size; int64_t last_pts; int eof; } ATDecodeContext; @@ -81,20 +82,127 @@ static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile) } } -static void ffat_update_ctx(AVCodecContext *avctx) +static int ffat_get_channel_id(AudioChannelLabel label) +{ + if (label == 0) + return -1; + else if (label <= kAudioChannelLabel_LFEScreen) + return label - 1; + else if (label <= kAudioChannelLabel_RightSurround) + return label + 4; + else if (label <= kAudioChannelLabel_CenterSurround) + return label + 1; + else if (label <= kAudioChannelLabel_RightSurroundDirect) + return label + 23; + else if (label <= kAudioChannelLabel_TopBackRight) + return label - 1; + else if (label < kAudioChannelLabel_RearSurroundLeft) + return -1; + else if (label <= kAudioChannelLabel_RearSurroundRight) + return label - 29; + else if (label <= kAudioChannelLabel_RightWide) + return label - 4; + else if (label == kAudioChannelLabel_LFE2) + return ff_ctzll(AV_CH_LOW_FREQUENCY_2); + else if (label == kAudioChannelLabel_Mono) + return ff_ctzll(AV_CH_FRONT_CENTER); + else + return -1; +} + +static int ffat_compare_channel_descriptions(const void* a, const void* b) +{ + const AudioChannelDescription* da = a; + const AudioChannelDescription* db = b; + return ffat_get_channel_id(da->mChannelLabel) - ffat_get_channel_id(db->mChannelLabel); +} + +static AudioChannelLayout *ffat_convert_layout(AudioChannelLayout *layout, UInt32* size) +{ + AudioChannelLayoutTag tag = layout->mChannelLayoutTag; + AudioChannelLayout *new_layout; + if (tag == kAudioChannelLayoutTag_UseChannelDescriptions) + return layout; + else if (tag == kAudioChannelLayoutTag_UseChannelBitmap) + AudioFormatGetPropertyInfo(kAudioFormatProperty_ChannelLayoutForBitmap, + sizeof(UInt32), &layout->mChannelBitmap, size); + else + AudioFormatGetPropertyInfo(kAudioFormatProperty_ChannelLayoutForTag, + sizeof(AudioChannelLayoutTag), &tag, size); + new_layout = av_malloc(*size); + if (!new_layout) { + av_free(layout); + return NULL; + } + if (tag == kAudioChannelLayoutTag_UseChannelBitmap) + AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForBitmap, + sizeof(UInt32), &layout->mChannelBitmap, size, new_layout); + else + AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForTag, + sizeof(AudioChannelLayoutTag), &tag, size, new_layout); + new_layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions; + av_free(layout); + return new_layout; +} + +static int ffat_update_ctx(AVCodecContext *avctx) { ATDecodeContext *at = avctx->priv_data; - AudioStreamBasicDescription in_format; - UInt32 size = sizeof(in_format); + AudioStreamBasicDescription format; + UInt32 size = sizeof(format); if (!AudioConverterGetProperty(at->converter, kAudioConverterCurrentInputStreamDescription, - &size, &in_format)) { - avctx->channels = in_format.mChannelsPerFrame; - at->pkt_size = in_format.mFramesPerPacket; + &size, &format)) { + if (format.mSampleRate) + avctx->sample_rate = format.mSampleRate; + avctx->channels = format.mChannelsPerFrame; + avctx->channel_layout = av_get_default_channel_layout(avctx->channels); + avctx->frame_size = format.mFramesPerPacket; + } + + if (!AudioConverterGetProperty(at->converter, + kAudioConverterCurrentOutputStreamDescription, + &size, &format)) { + format.mSampleRate = avctx->sample_rate; + format.mChannelsPerFrame = avctx->channels; + AudioConverterSetProperty(at->converter, + kAudioConverterCurrentOutputStreamDescription, + size, &format); + } + + if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterOutputChannelLayout, + &size, NULL) && size) { + AudioChannelLayout *layout = av_malloc(size); + uint64_t layout_mask = 0; + int i; + if (!layout) + return AVERROR(ENOMEM); + AudioConverterGetProperty(at->converter, kAudioConverterOutputChannelLayout, + &size, layout); + if (!(layout = ffat_convert_layout(layout, &size))) + return AVERROR(ENOMEM); + for (i = 0; i < layout->mNumberChannelDescriptions; i++) { + int id = ffat_get_channel_id(layout->mChannelDescriptions[i].mChannelLabel); + if (id < 0) + goto done; + if (layout_mask & (1 << id)) + goto done; + layout_mask |= 1 << id; + layout->mChannelDescriptions[i].mChannelFlags = i; // Abusing flags as index + } + avctx->channel_layout = layout_mask; + qsort(layout->mChannelDescriptions, layout->mNumberChannelDescriptions, + sizeof(AudioChannelDescription), &ffat_compare_channel_descriptions); + for (i = 0; i < layout->mNumberChannelDescriptions; i++) + at->channel_map[i] = layout->mChannelDescriptions[i].mChannelFlags; +done: + av_free(layout); } - if (!at->pkt_size) - at->pkt_size = 2048; + if (!avctx->frame_size) + avctx->frame_size = 2048; + + return 0; } static void put_descr(PutByteContext *pb, int tag, unsigned int size) @@ -106,42 +214,11 @@ static void put_descr(PutByteContext *pb, int tag, unsigned int size) bytestream2_put_byte(pb, size & 0x7F); } -static av_cold int ffat_init_decoder(AVCodecContext *avctx) +static int ffat_set_extradata(AVCodecContext *avctx) { ATDecodeContext *at = avctx->priv_data; - OSStatus status; - - enum AVSampleFormat sample_fmt = (avctx->bits_per_raw_sample == 32) ? - AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S16; - - AudioStreamBasicDescription in_format = { - .mSampleRate = avctx->sample_rate ? avctx->sample_rate : 44100, - .mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile), - .mBytesPerPacket = avctx->block_align, - .mChannelsPerFrame = avctx->channels ? avctx->channels : 1, - }; - AudioStreamBasicDescription out_format = { - .mSampleRate = in_format.mSampleRate, - .mFormatID = kAudioFormatLinearPCM, - .mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked, - .mFramesPerPacket = 1, - .mChannelsPerFrame = in_format.mChannelsPerFrame, - .mBitsPerChannel = av_get_bytes_per_sample(sample_fmt) * 8, - }; - - avctx->sample_fmt = sample_fmt; - - if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT) - in_format.mFramesPerPacket = 64; - - status = AudioConverterNew(&in_format, &out_format, &at->converter); - - if (status != 0) { - av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status); - return AVERROR_UNKNOWN; - } - if (avctx->extradata_size) { + OSStatus status; char *extradata = avctx->extradata; int extradata_size = avctx->extradata_size; if (avctx->codec_id == AV_CODEC_ID_AAC) { @@ -180,15 +257,74 @@ static av_cold int ffat_init_decoder(AVCodecContext *avctx) extradata_size, extradata); if (status != 0) av_log(avctx, AV_LOG_WARNING, "AudioToolbox cookie error: %i\n", (int)status); + + if (avctx->codec_id == AV_CODEC_ID_AAC) + av_free(extradata); + } + return 0; +} + +static av_cold int ffat_create_decoder(AVCodecContext *avctx) +{ + ATDecodeContext *at = avctx->priv_data; + OSStatus status; + int i; + + enum AVSampleFormat sample_fmt = (avctx->bits_per_raw_sample == 32) ? + AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S16; + + AudioStreamBasicDescription in_format = { + .mSampleRate = avctx->sample_rate ? avctx->sample_rate : 44100, + .mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile), + .mBytesPerPacket = avctx->block_align, + .mChannelsPerFrame = avctx->channels ? avctx->channels : 1, + }; + AudioStreamBasicDescription out_format = { + .mSampleRate = in_format.mSampleRate, + .mFormatID = kAudioFormatLinearPCM, + .mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked, + .mFramesPerPacket = 1, + .mChannelsPerFrame = in_format.mChannelsPerFrame, + .mBitsPerChannel = av_get_bytes_per_sample(sample_fmt) * 8, + }; + + avctx->sample_fmt = sample_fmt; + + if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT) + in_format.mFramesPerPacket = 64; + + status = AudioConverterNew(&in_format, &out_format, &at->converter); + + if (status != 0) { + av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status); + return AVERROR_UNKNOWN; } + if ((status = ffat_set_extradata(avctx)) < 0) + return status; + + for (i = 0; i < (sizeof(at->channel_map) / sizeof(at->channel_map[0])); i++) + at->channel_map[i] = i; + ffat_update_ctx(avctx); + if(!(at->decoded_data = av_malloc(av_get_bytes_per_sample(avctx->sample_fmt) + * avctx->frame_size * avctx->channels))) + return AVERROR(ENOMEM); + at->last_pts = AV_NOPTS_VALUE; return 0; } +static av_cold int ffat_init_decoder(AVCodecContext *avctx) +{ + if (avctx->channels || avctx->extradata_size) + return ffat_create_decoder(avctx); + else + return 0; +} + static OSStatus ffat_decode_callback(AudioConverterRef converter, UInt32 *nb_packets, AudioBufferList *data, AudioStreamPacketDescription **packets, @@ -229,6 +365,26 @@ static OSStatus ffat_decode_callback(AudioConverterRef converter, UInt32 *nb_pac return 0; } +#define COPY_SAMPLES(type) \ + type *in_ptr = (type*)at->decoded_data; \ + type *end_ptr = in_ptr + frame->nb_samples * avctx->channels; \ + type *out_ptr = (type*)frame->data[0]; \ + for (; in_ptr < end_ptr; in_ptr += avctx->channels, out_ptr += avctx->channels) { \ + int c; \ + for (c = 0; c < avctx->channels; c++) \ + out_ptr[c] = in_ptr[at->channel_map[c]]; \ + } + +static void ffat_copy_samples(AVCodecContext *avctx, AVFrame *frame) +{ + ATDecodeContext *at = avctx->priv_data; + if (avctx->sample_fmt == AV_SAMPLE_FMT_S32) { + COPY_SAMPLES(int32_t); + } else { + COPY_SAMPLES(int16_t); + } +} + static int ffat_decode(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { @@ -237,24 +393,13 @@ static int ffat_decode(AVCodecContext *avctx, void *data, int pkt_size = avpkt->size; AVPacket filtered_packet; OSStatus ret; - - AudioBufferList out_buffers = { - .mNumberBuffers = 1, - .mBuffers = { - { - .mNumberChannels = avctx->channels, - .mDataByteSize = av_get_bytes_per_sample(avctx->sample_fmt) * at->pkt_size * avctx->channels, - } - } - }; + AudioBufferList out_buffers; if (avctx->codec_id == AV_CODEC_ID_AAC && avpkt->size > 2 && (AV_RB16(avpkt->data) & 0xfff0) == 0xfff0) { - int first = 0; uint8_t *p_filtered = NULL; int n_filtered = 0; if (!at->bsf) { - first = 1; if(!(at->bsf = av_bitstream_filter_init("aac_adtstoasc"))) return AVERROR(ENOMEM); } @@ -267,16 +412,24 @@ static int ffat_decode(AVCodecContext *avctx, void *data, avpkt->data = p_filtered; avpkt->size = n_filtered; } + } - if (first) { - if ((ret = ffat_set_extradata(avctx)) < 0) - return ret; - ffat_update_ctx(avctx); - out_buffers.mBuffers[0].mNumberChannels = avctx->channels; - out_buffers.mBuffers[0].mDataByteSize = av_get_bytes_per_sample(avctx->sample_fmt) * at->pkt_size * avctx->channels; - } + if (!at->converter) { + if ((ret = ffat_create_decoder(avctx)) < 0) + return ret; } + out_buffers = (AudioBufferList){ + .mNumberBuffers = 1, + .mBuffers = { + { + .mNumberChannels = avctx->channels, + .mDataByteSize = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->frame_size + * avctx->channels, + } + } + }; + av_packet_unref(&at->new_in_pkt); if (avpkt->size) { @@ -289,17 +442,19 @@ static int ffat_decode(AVCodecContext *avctx, void *data, frame->sample_rate = avctx->sample_rate; - frame->nb_samples = at->pkt_size; - ff_get_buffer(avctx, frame, 0); + frame->nb_samples = avctx->frame_size; - out_buffers.mBuffers[0].mData = frame->data[0]; + out_buffers.mBuffers[0].mData = at->decoded_data; ret = AudioConverterFillComplexBuffer(at->converter, ffat_decode_callback, avctx, &frame->nb_samples, &out_buffers, NULL); if ((!ret || ret == 1) && frame->nb_samples) { + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + ffat_copy_samples(avctx, frame); *got_frame_ptr = 1; if (at->last_pts != AV_NOPTS_VALUE) { - frame->pts = at->last_pts; + frame->pkt_pts = at->last_pts; at->last_pts = avpkt->pts; } } else if (ret && ret != 1) { @@ -325,6 +480,7 @@ static av_cold int ffat_close_decoder(AVCodecContext *avctx) AudioConverterDispose(at->converter); av_packet_unref(&at->new_in_pkt); av_packet_unref(&at->in_pkt); + av_free(at->decoded_data); return 0; } -- cgit v1.2.3