diff --git a/libavcodec/8svx.c b/libavcodec/8svx.c index 4f95d9034e..5d94e005a2 100644 --- a/libavcodec/8svx.c +++ b/libavcodec/8svx.c @@ -1,5 +1,6 @@ /* * Copyright (C) 2008 Jaikrishnan Menon + * Copyright (C) 2011 Stefano Sabatini * * This file is part of FFmpeg. * @@ -38,62 +39,155 @@ /** decoder context */ typedef struct EightSvxContext { - int16_t fib_acc; - const int16_t *table; + const int8_t *table; + + /* buffer used to store the whole audio decoded/interleaved chunk, + * which is sent with the first packet */ + uint8_t *samples; + size_t samples_size; + int samples_idx; } EightSvxContext; -static const int16_t fibonacci[16] = { -34<<8, -21<<8, -13<<8, -8<<8, -5<<8, -3<<8, -2<<8, -1<<8, - 0, 1<<8, 2<<8, 3<<8, 5<<8, 8<<8, 13<<8, 21<<8 }; -static const int16_t exponential[16] = { -128<<8, -64<<8, -32<<8, -16<<8, -8<<8, -4<<8, -2<<8, -1<<8, - 0, 1<<8, 2<<8, 4<<8, 8<<8, 16<<8, 32<<8, 64<<8 }; +static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 }; +static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 }; + +#define MAX_FRAME_SIZE 2048 + +/** + * Interleave samples in buffer containing all left channel samples + * at the beginning, and right channel samples at the end. + * Each sample is assumed to be in signed 8-bit format. + * + * @param size the size in bytes of the dst and src buffer + */ +static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size) +{ + uint8_t *dst_end = dst + size; + size = size>>1; + + while (dst < dst_end) { + *dst++ = *src; + *dst++ = *(src+size); + src++; + } +} + +/** + * Delta decode the compressed values in src, and put the resulting + * decoded n samples in dst. + * + * @param val starting value assumed by the delta sequence + * @param table delta sequence table + * @return size in bytes of the decoded data, must be src_size*2 + */ +static int delta_decode(int8_t *dst, const uint8_t *src, int src_size, + int8_t val, const int8_t *table) +{ + int n = src_size; + int8_t *dst0 = dst; + + while (n--) { + uint8_t d = *src++; + val = av_clip(val + table[d & 0x0f], -127, 128); + *dst++ = val; + val = av_clip(val + table[d >> 4] , -127, 128); + *dst++ = val; + } + + return dst-dst0; +} static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; EightSvxContext *esc = avctx->priv_data; - int16_t *out_data = data; - int consumed = buf_size; - const uint8_t *buf_end = buf + buf_size; + int out_data_size, n; + uint8_t *src, *dst; - if((*data_size >> 2) < buf_size) - return -1; + /* decode and interleave the first packet */ + if (!esc->samples && avpkt) { + uint8_t *deinterleaved_samples; - if(avctx->frame_number == 0) { - esc->fib_acc = buf[1] << 8; - buf_size -= 2; - buf += 2; - } + esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW ? + avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2; + if (!(esc->samples = av_malloc(esc->samples_size))) + return AVERROR(ENOMEM); - *data_size = buf_size << 2; + /* decompress */ + if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) { + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + int n = esc->samples_size; - while(buf < buf_end) { - uint8_t d = *buf++; - esc->fib_acc += esc->table[d & 0x0f]; - *out_data++ = esc->fib_acc; - esc->fib_acc += esc->table[d >> 4]; - *out_data++ = esc->fib_acc; + if (!(deinterleaved_samples = av_mallocz(n))) + return AVERROR(ENOMEM); + + /* the uncompressed starting value is contained in the first byte */ + if (avctx->channels == 2) { + delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table); + buf += buf_size/2; + delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table); + } else + delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table); + } else { + deinterleaved_samples = avpkt->data; + } + + if (avctx->channels == 2) + interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size); + else + memcpy(esc->samples, deinterleaved_samples, esc->samples_size); } - return consumed; + /* return single packed with fixed size */ + out_data_size = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx); + if (*data_size < out_data_size) { + av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", *data_size); + return AVERROR(EINVAL); + } + + *data_size = out_data_size; + dst = data; + src = esc->samples + esc->samples_idx; + for (n = out_data_size; n > 0; n--) + *dst++ = *src++ + 128; + esc->samples_idx += *data_size; + + return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ? + (avctx->frame_number == 0)*2 + out_data_size / 2 : + out_data_size; } static av_cold int eightsvx_decode_init(AVCodecContext *avctx) { EightSvxContext *esc = avctx->priv_data; - switch(avctx->codec->id) { - case CODEC_ID_8SVX_FIB: - esc->table = fibonacci; - break; - case CODEC_ID_8SVX_EXP: - esc->table = exponential; - break; - default: - return -1; + if (avctx->channels > 2) { + av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n"); + return AVERROR_INVALIDDATA; } - avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + switch (avctx->codec->id) { + case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break; + case CODEC_ID_8SVX_EXP: esc->table = exponential; break; + case CODEC_ID_8SVX_RAW: esc->table = NULL; break; + default: + av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id); + return AVERROR_INVALIDDATA; + } + avctx->sample_fmt = AV_SAMPLE_FMT_U8; + + return 0; +} + +static av_cold int eightsvx_decode_close(AVCodecContext *avctx) +{ + EightSvxContext *esc = avctx->priv_data; + + av_freep(&esc->samples); + esc->samples_size = 0; + esc->samples_idx = 0; + return 0; } @@ -104,6 +198,7 @@ AVCodec ff_eightsvx_fib_decoder = { .priv_data_size = sizeof (EightSvxContext), .init = eightsvx_decode_init, .decode = eightsvx_decode_frame, + .close = eightsvx_decode_close, .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"), }; @@ -114,5 +209,17 @@ AVCodec ff_eightsvx_exp_decoder = { .priv_data_size = sizeof (EightSvxContext), .init = eightsvx_decode_init, .decode = eightsvx_decode_frame, + .close = eightsvx_decode_close, .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"), }; + +AVCodec ff_eightsvx_raw_decoder = { + .name = "8svx_raw", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_8SVX_RAW, + .priv_data_size = sizeof(EightSvxContext), + .init = eightsvx_decode_init, + .decode = eightsvx_decode_frame, + .close = eightsvx_decode_close, + .long_name = NULL_IF_CONFIG_SMALL("8SVX rawaudio"), +}; diff --git a/libavcodec/Makefile b/libavcodec/Makefile index e293438e45..ac16e06df8 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -136,6 +136,7 @@ OBJS-$(CONFIG_EATQI_DECODER) += eatqi.o eaidct.o mpeg12.o \ OBJS-$(CONFIG_EIGHTBPS_DECODER) += 8bps.o OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER) += 8svx.o OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER) += 8svx.o +OBJS-$(CONFIG_EIGHTSVX_RAW_DECODER) += 8svx.o OBJS-$(CONFIG_ESCAPE124_DECODER) += escape124.o OBJS-$(CONFIG_FFV1_DECODER) += ffv1.o rangecoder.o OBJS-$(CONFIG_FFV1_ENCODER) += ffv1.o rangecoder.o diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index fc74eeaf8c..ff032dda85 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -104,6 +104,7 @@ void avcodec_register_all(void) REGISTER_DECODER (EIGHTBPS, eightbps); REGISTER_DECODER (EIGHTSVX_EXP, eightsvx_exp); REGISTER_DECODER (EIGHTSVX_FIB, eightsvx_fib); + REGISTER_DECODER (EIGHTSVX_RAW, eightsvx_raw); REGISTER_DECODER (ESCAPE124, escape124); REGISTER_ENCDEC (FFV1, ffv1); REGISTER_ENCDEC (FFVHUFF, ffvhuff); diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index 2fbf9cfc2a..d1a5e6655e 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -204,6 +204,7 @@ enum CodecID { CODEC_ID_PRORES, CODEC_ID_JV, CODEC_ID_DFA, + CODEC_ID_8SVX_RAW, /* various PCM "codecs" */ CODEC_ID_PCM_S16LE= 0x10000, diff --git a/libavcodec/version.h b/libavcodec/version.h index 067cf4af89..471e3aaa9a 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -21,7 +21,7 @@ #define AVCODEC_VERSION_H #define LIBAVCODEC_VERSION_MAJOR 53 -#define LIBAVCODEC_VERSION_MINOR 5 +#define LIBAVCODEC_VERSION_MINOR 6 #define LIBAVCODEC_VERSION_MICRO 0 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ diff --git a/libavformat/iff.c b/libavformat/iff.c index b092de0b08..f6edcdda2e 100644 --- a/libavformat/iff.c +++ b/libavformat/iff.c @@ -60,8 +60,6 @@ #define RIGHT 4 #define STEREO 6 -#define PACKET_SIZE 1024 - /** * This number of bytes if added at the beginning of each AVPacket * which contain additional information about video properties @@ -97,19 +95,6 @@ typedef struct { unsigned masking; ///< masking method used } IffDemuxContext; - -static void interleave_stereo(const uint8_t *src, uint8_t *dest, int size) -{ - uint8_t *end = dest + size; - size = size>>1; - - while(dest < end) { - *dest++ = *src; - *dest++ = *(src+size); - src++; - } -} - /* Metadata string read */ static int get_metadata(AVFormatContext *s, const char *const tag, @@ -255,7 +240,7 @@ static int iff_read_header(AVFormatContext *s, switch (iff->svx8_compression) { case COMP_NONE: - st->codec->codec_id = CODEC_ID_PCM_S8; + st->codec->codec_id = CODEC_ID_8SVX_RAW; break; case COMP_FIB: st->codec->codec_id = CODEC_ID_8SVX_FIB; @@ -330,15 +315,8 @@ static int iff_read_packet(AVFormatContext *s, if(iff->sent_bytes >= iff->body_size) return AVERROR(EIO); - if(st->codec->channels == 2) { - uint8_t sample_buffer[PACKET_SIZE]; - - ret = avio_read(pb, sample_buffer, PACKET_SIZE); - if(av_new_packet(pkt, PACKET_SIZE) < 0) { - av_log(s, AV_LOG_ERROR, "cannot allocate packet\n"); - return AVERROR(ENOMEM); - } - interleave_stereo(sample_buffer, pkt->data, PACKET_SIZE); + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + ret = av_get_packet(pb, pkt, iff->body_size); } else if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { uint8_t *buf; @@ -349,23 +327,13 @@ static int iff_read_packet(AVFormatContext *s, buf = pkt->data; bytestream_put_be16(&buf, 2); ret = avio_read(pb, buf, iff->body_size); - } else { - ret = av_get_packet(pb, pkt, PACKET_SIZE); } if(iff->sent_bytes == 0) pkt->flags |= AV_PKT_FLAG_KEY; + iff->sent_bytes = iff->body_size; - if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - iff->sent_bytes += PACKET_SIZE; - } else { - iff->sent_bytes = iff->body_size; - } pkt->stream_index = 0; - if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - pkt->pts = iff->audio_frame_count; - iff->audio_frame_count += ret / st->codec->channels; - } return ret; } diff --git a/tests/ref/fate/iff-fibonacci b/tests/ref/fate/iff-fibonacci index e452f31e6c..947f78e964 100644 --- a/tests/ref/fate/iff-fibonacci +++ b/tests/ref/fate/iff-fibonacci @@ -1 +1 @@ -e968a853779bb6438339e3b8d69d8d24 +e76b025238a6a27968f8644f4ccc3207