mirror of https://git.ffmpeg.org/ffmpeg.git
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.
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fa1749dd34
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730c023260
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@ -35,6 +35,7 @@
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#include "avcodec.h"
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#include "bitstream.h"
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#include "dct.h"
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#include "decode.h"
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#include "internal.h"
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#include "rdft.h"
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#include "wma_freqs.h"
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@ -57,7 +58,7 @@ typedef struct BinkAudioContext {
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float root;
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DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
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float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
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uint8_t *packet_buffer;
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AVPacket *pkt;
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union {
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RDFTContext rdft;
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DCTContext dct;
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@ -140,6 +141,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
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else
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return -1;
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s->pkt = av_packet_alloc();
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if (!s->pkt)
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return AVERROR(ENOMEM);
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return 0;
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}
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@ -269,12 +274,13 @@ static av_cold int decode_end(AVCodecContext *avctx)
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{
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BinkAudioContext * s = avctx->priv_data;
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av_freep(&s->bands);
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av_freep(&s->packet_buffer);
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if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
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ff_rdft_end(&s->trans.rdft);
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else if (CONFIG_BINKAUDIO_DCT_DECODER)
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ff_dct_end(&s->trans.dct);
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av_packet_free(&s->pkt);
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return 0;
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}
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@ -285,32 +291,26 @@ static void get_bits_align32(BitstreamContext *s)
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bitstream_skip(s, n);
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}
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static int decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame)
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{
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BinkAudioContext *s = avctx->priv_data;
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AVFrame *frame = data;
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BitstreamContext *bc = &s->bc;
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int ret, consumed = 0;
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int ret;
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if (!bitstream_bits_left(bc)) {
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uint8_t *buf;
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/* handle end-of-stream */
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if (!avpkt->size) {
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*got_frame_ptr = 0;
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return 0;
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}
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if (avpkt->size < 4) {
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if (!s->pkt->data) {
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ret = ff_decode_get_packet(avctx, s->pkt);
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if (ret < 0)
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return ret;
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if (s->pkt->size < 4) {
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av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
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return AVERROR_INVALIDDATA;
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ret = AVERROR_INVALIDDATA;
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goto fail;
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}
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buf = av_realloc(s->packet_buffer, avpkt->size + AV_INPUT_BUFFER_PADDING_SIZE);
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if (!buf)
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return AVERROR(ENOMEM);
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s->packet_buffer = buf;
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memcpy(s->packet_buffer, avpkt->data, avpkt->size);
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bitstream_init(bc, s->packet_buffer, avpkt->size * 8);
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consumed = avpkt->size;
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ret = bitstream_init8(bc, s->pkt->data, s->pkt->size);
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if (ret < 0)
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goto fail;
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/* skip reported size */
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bitstream_skip(bc, 32);
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@ -329,11 +329,17 @@ static int decode_frame(AVCodecContext *avctx, void *data,
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return AVERROR_INVALIDDATA;
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}
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get_bits_align32(bc);
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if (!bitstream_bits_left(bc)) {
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memset(bc, 0, sizeof(*bc));
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av_packet_unref(s->pkt);
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}
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frame->nb_samples = s->block_size / avctx->channels;
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*got_frame_ptr = 1;
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return consumed;
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return 0;
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fail:
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av_packet_unref(s->pkt);
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return ret;
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}
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AVCodec ff_binkaudio_rdft_decoder = {
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@ -344,7 +350,7 @@ AVCodec ff_binkaudio_rdft_decoder = {
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.priv_data_size = sizeof(BinkAudioContext),
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.init = decode_init,
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.close = decode_end,
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.decode = decode_frame,
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.receive_frame = binkaudio_receive_frame,
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.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
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};
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@ -356,6 +362,6 @@ AVCodec ff_binkaudio_dct_decoder = {
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.priv_data_size = sizeof(BinkAudioContext),
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.init = decode_init,
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.close = decode_end,
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.decode = decode_frame,
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.receive_frame = binkaudio_receive_frame,
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.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
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};
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