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lavc/audiotoolboxdec: move to new BSF API

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This commit is contained in:
Rodger Combs 2016-04-19 03:50:28 -05:00
parent 95116bf35f
commit e1c20485c1
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GPG Key ID: E3E54DCDCD3CB843
1 changed files with 54 additions and 28 deletions
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82
libavcodec/audiotoolboxdec.c
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@ -43,10 +43,13 @@ typedef struct ATDecodeContext {
AudioStreamPacketDescription pkt_desc;
AVPacket in_pkt;
AVPacket new_in_pkt;
AVBitStreamFilterContext *bsf;
AVBSFContext *bsf;
char *decoded_data;
int channel_map[64];
uint8_t *extradata;
int extradata_size;
int64_t last_pts;
int eof;
} ATDecodeContext;
@ -224,22 +227,23 @@ static void put_descr(PutByteContext *pb, int tag, unsigned int size)
static uint8_t* ffat_get_magic_cookie(AVCodecContext *avctx, UInt32 *cookie_size)
{
ATDecodeContext *at = avctx->priv_data;
if (avctx->codec_id == AV_CODEC_ID_AAC) {
char *extradata;
PutByteContext pb;
*cookie_size = 5 + 3 + 5+13 + 5+avctx->extradata_size;
*cookie_size = 5 + 3 + 5+13 + 5+at->extradata_size;
if (!(extradata = av_malloc(*cookie_size)))
return NULL;
bytestream2_init_writer(&pb, extradata, *cookie_size);
// ES descriptor
put_descr(&pb, 0x03, 3 + 5+13 + 5+avctx->extradata_size);
put_descr(&pb, 0x03, 3 + 5+13 + 5+at->extradata_size);
bytestream2_put_be16(&pb, 0);
bytestream2_put_byte(&pb, 0x00); // flags (= no flags)
// DecoderConfig descriptor
put_descr(&pb, 0x04, 13 + 5+avctx->extradata_size);
put_descr(&pb, 0x04, 13 + 5+at->extradata_size);
// Object type indication
bytestream2_put_byte(&pb, 0x40);
@ -252,18 +256,19 @@ static uint8_t* ffat_get_magic_cookie(AVCodecContext *avctx, UInt32 *cookie_size
bytestream2_put_be32(&pb, 0); // avgbitrate
// DecoderSpecific info descriptor
put_descr(&pb, 0x05, avctx->extradata_size);
bytestream2_put_buffer(&pb, avctx->extradata, avctx->extradata_size);
put_descr(&pb, 0x05, at->extradata_size);
bytestream2_put_buffer(&pb, at->extradata, at->extradata_size);
return extradata;
} else {
*cookie_size = avctx->extradata_size;
return avctx->extradata;
*cookie_size = at->extradata_size;
return at->extradata;
}
}
static av_cold int ffat_usable_extradata(AVCodecContext *avctx)
{
return avctx->extradata_size &&
ATDecodeContext *at = avctx->priv_data;
return at->extradata_size &&
(avctx->codec_id == AV_CODEC_ID_ALAC ||
avctx->codec_id == AV_CODEC_ID_AAC);
}
@ -284,7 +289,7 @@ static int ffat_set_extradata(AVCodecContext *avctx)
if (status != 0)
av_log(avctx, AV_LOG_WARNING, "AudioToolbox cookie error: %i\n", (int)status);
if (cookie != avctx->extradata)
if (cookie != at->extradata)
av_free(cookie);
}
return 0;
@ -320,7 +325,7 @@ static av_cold int ffat_create_decoder(AVCodecContext *avctx, AVPacket *pkt)
return AVERROR(ENOMEM);
status = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
cookie_size, cookie, &format_size, &in_format);
if (cookie != avctx->extradata)
if (cookie != at->extradata)
av_free(cookie);
if (status != 0) {
av_log(avctx, AV_LOG_ERROR, "AudioToolbox header-parse error: %i\n", (int)status);
@ -391,6 +396,10 @@ static av_cold int ffat_create_decoder(AVCodecContext *avctx, AVPacket *pkt)
static av_cold int ffat_init_decoder(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
at->extradata = avctx->extradata;
at->extradata_size = avctx->extradata_size;
if ((avctx->channels && avctx->sample_rate) || ffat_usable_extradata(avctx))
return ffat_create_decoder(avctx, NULL);
else
@ -416,8 +425,6 @@ static OSStatus ffat_decode_callback(AudioConverterRef converter, UInt32 *nb_pac
av_packet_unref(&at->in_pkt);
av_packet_move_ref(&at->in_pkt, &at->new_in_pkt);
at->new_in_pkt.data = 0;
at->new_in_pkt.size = 0;
if (!at->in_pkt.data) {
*nb_packets = 0;
@ -464,33 +471,49 @@ static int ffat_decode(AVCodecContext *avctx, void *data,
ATDecodeContext *at = avctx->priv_data;
AVFrame *frame = data;
int pkt_size = avpkt->size;
AVPacket filtered_packet;
AVPacket filtered_packet = {0};
OSStatus ret;
AudioBufferList out_buffers;
if (avctx->codec_id == AV_CODEC_ID_AAC && avpkt->size > 2 &&
(AV_RB16(avpkt->data) & 0xfff0) == 0xfff0) {
uint8_t *p_filtered = NULL;
int n_filtered = 0;
AVPacket filter_pkt = {0};
if (!at->bsf) {
if(!(at->bsf = av_bitstream_filter_init("aac_adtstoasc")))
return AVERROR(ENOMEM);
const AVBitStreamFilter *bsf = av_bsf_get_by_name("aac_adtstoasc");
if(!bsf)
return AVERROR_BSF_NOT_FOUND;
if ((ret = av_bsf_alloc(bsf, &at->bsf)))
return ret;
if (((ret = avcodec_parameters_from_context(at->bsf->par_in, avctx)) < 0) ||
((ret = av_bsf_init(at->bsf)) < 0)) {
av_bsf_free(&at->bsf);
return ret;
}
}
ret = av_bitstream_filter_filter(at->bsf, avctx, NULL, &p_filtered, &n_filtered,
avpkt->data, avpkt->size, 0);
if (ret >= 0 && p_filtered != avpkt->data) {
filtered_packet = *avpkt;
avpkt = &filtered_packet;
avpkt->data = p_filtered;
avpkt->size = n_filtered;
if ((ret = av_packet_ref(&filter_pkt, avpkt)) < 0)
return ret;
if ((ret = av_bsf_send_packet(at->bsf, &filter_pkt)) < 0) {
av_packet_unref(&filter_pkt);
return ret;
}
if ((ret = av_bsf_receive_packet(at->bsf, &filtered_packet)) < 0)
return ret;
at->extradata = at->bsf->par_out->extradata;
at->extradata_size = at->bsf->par_out->extradata_size;
avpkt = &filtered_packet;
}
if (!at->converter) {
if ((ret = ffat_create_decoder(avctx, avpkt)) < 0)
if ((ret = ffat_create_decoder(avctx, avpkt)) < 0) {
av_packet_unref(&filtered_packet);
return ret;
}
}
out_buffers = (AudioBufferList){
.mNumberBuffers = 1,
@ -506,9 +529,11 @@ static int ffat_decode(AVCodecContext *avctx, void *data,
av_packet_unref(&at->new_in_pkt);
if (avpkt->size) {
if ((ret = av_packet_ref(&at->new_in_pkt, avpkt)) < 0)
if (filtered_packet.data) {
at->new_in_pkt = filtered_packet;
} else if ((ret = av_packet_ref(&at->new_in_pkt, avpkt)) < 0) {
return ret;
at->new_in_pkt.data = avpkt->data;
}
} else {
at->eof = 1;
}
@ -551,6 +576,7 @@ static av_cold int ffat_close_decoder(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
AudioConverterDispose(at->converter);
av_bsf_free(&at->bsf);
av_packet_unref(&at->new_in_pkt);
av_packet_unref(&at->in_pkt);
av_free(at->decoded_data);