binkaudio: switch to the new send/receive API

It is more natural for this codec and allows to avoid awkward constructs
like "consuming 0 bytes from input". Also, keep a reference to the input
packet to avoid unnecessary copying.
pull/254/merge
Anton Khirnov 8 years ago
parent fa1749dd34
commit 730c023260
  1. 58
      libavcodec/binkaudio.c

@ -35,6 +35,7 @@
#include "avcodec.h"
#include "bitstream.h"
#include "dct.h"
#include "decode.h"
#include "internal.h"
#include "rdft.h"
#include "wma_freqs.h"
@ -57,7 +58,7 @@ typedef struct BinkAudioContext {
float root;
DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
uint8_t *packet_buffer;
AVPacket *pkt;
union {
RDFTContext rdft;
DCTContext dct;
@ -140,6 +141,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
else
return -1;
s->pkt = av_packet_alloc();
if (!s->pkt)
return AVERROR(ENOMEM);
return 0;
}
@ -269,12 +274,13 @@ static av_cold int decode_end(AVCodecContext *avctx)
{
BinkAudioContext * s = avctx->priv_data;
av_freep(&s->bands);
av_freep(&s->packet_buffer);
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_end(&s->trans.rdft);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
ff_dct_end(&s->trans.dct);
av_packet_free(&s->pkt);
return 0;
}
@ -285,32 +291,26 @@ static void get_bits_align32(BitstreamContext *s)
bitstream_skip(s, n);
}
static int decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
{
BinkAudioContext *s = avctx->priv_data;
AVFrame *frame = data;
BitstreamContext *bc = &s->bc;
int ret, consumed = 0;
int ret;
if (!bitstream_bits_left(bc)) {
uint8_t *buf;
/* handle end-of-stream */
if (!avpkt->size) {
*got_frame_ptr = 0;
return 0;
}
if (avpkt->size < 4) {
if (!s->pkt->data) {
ret = ff_decode_get_packet(avctx, s->pkt);
if (ret < 0)
return ret;
if (s->pkt->size < 4) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
ret = AVERROR_INVALIDDATA;
goto fail;
}
buf = av_realloc(s->packet_buffer, avpkt->size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!buf)
return AVERROR(ENOMEM);
s->packet_buffer = buf;
memcpy(s->packet_buffer, avpkt->data, avpkt->size);
bitstream_init(bc, s->packet_buffer, avpkt->size * 8);
consumed = avpkt->size;
ret = bitstream_init8(bc, s->pkt->data, s->pkt->size);
if (ret < 0)
goto fail;
/* skip reported size */
bitstream_skip(bc, 32);
@ -329,11 +329,17 @@ static int decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}
get_bits_align32(bc);
if (!bitstream_bits_left(bc)) {
memset(bc, 0, sizeof(*bc));
av_packet_unref(s->pkt);
}
frame->nb_samples = s->block_size / avctx->channels;
*got_frame_ptr = 1;
return consumed;
return 0;
fail:
av_packet_unref(s->pkt);
return ret;
}
AVCodec ff_binkaudio_rdft_decoder = {
@ -344,7 +350,7 @@ AVCodec ff_binkaudio_rdft_decoder = {
.priv_data_size = sizeof(BinkAudioContext),
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.receive_frame = binkaudio_receive_frame,
.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
};
@ -356,6 +362,6 @@ AVCodec ff_binkaudio_dct_decoder = {
.priv_data_size = sizeof(BinkAudioContext),
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.receive_frame = binkaudio_receive_frame,
.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
};

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