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authorAnton Khirnov <anton@khirnov.net>2016-11-24 23:05:20 +0100
committerAnton Khirnov <anton@khirnov.net>2016-12-14 09:06:45 +0100
commit730c02326094bcfb1fa67f10a7e7b22f03f5a88f (patch)
treed33291279c561c9f31a7e2ffbbd306681e8ffd6d
parentfa1749dd34c55fb997c97dfc4da9383c9976ab91 (diff)
downloadffmpeg-730c02326094bcfb1fa67f10a7e7b22f03f5a88f.tar.gz
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.
-rw-r--r--libavcodec/binkaudio.c58
1 files changed, 32 insertions, 26 deletions
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
index cf61135529..51fb6c83ce 100644
--- a/libavcodec/binkaudio.c
+++ b/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,
};