X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fmpegaudioenc.c;h=a1e2023ba1e96466afa549cfe294867b11010006;hb=257c7147e1fde196a3f649af6003eff8ddba787f;hp=767f823c01d843f6fefac9595f3633d35152446f;hpb=7356aaa78626b34c3b2779c7af84c92111025ebf;p=ffmpeg diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c index 767f823c01d..a1e2023ba1e 100644 --- a/libavcodec/mpegaudioenc.c +++ b/libavcodec/mpegaudioenc.c @@ -1,6 +1,6 @@ /* * The simplest mpeg audio layer 2 encoder - * Copyright (c) 2000, 2001 Fabrice Bellard. + * Copyright (c) 2000, 2001 Fabrice Bellard * * This file is part of FFmpeg. * @@ -20,12 +20,12 @@ */ /** - * @file mpegaudio.c + * @file libavcodec/mpegaudio.c * The simplest mpeg audio layer 2 encoder. */ #include "avcodec.h" -#include "bitstream.h" +#include "put_bits.h" #undef CONFIG_MPEGAUDIO_HP #define CONFIG_MPEGAUDIO_HP 0 @@ -59,7 +59,7 @@ typedef struct MpegAudioContext { } MpegAudioContext; /* define it to use floats in quantization (I don't like floats !) */ -//#define USE_FLOATS +#define USE_FLOATS #include "mpegaudiodata.h" #include "mpegaudiotab.h" @@ -126,10 +126,8 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx) s->sblimit = ff_mpa_sblimit_table[table]; s->alloc_table = ff_mpa_alloc_tables[table]; -#ifdef DEBUG - av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", - bitrate, freq, s->frame_size, table, s->frame_frac_incr); -#endif + dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", + bitrate, freq, s->frame_size, table, s->frame_frac_incr); for(i=0;inb_channels;i++) s->samples_offset[i] = 0; @@ -781,7 +779,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, encode_frame(s, bit_alloc, padding); s->nb_samples += MPA_FRAME_SIZE; - return pbBufPtr(&s->pb) - s->pb.buf; + return put_bits_ptr(&s->pb) - s->pb.buf; } static av_cold int MPA_encode_close(AVCodecContext *avctx) @@ -799,7 +797,7 @@ AVCodec mp2_encoder = { MPA_encode_frame, MPA_encode_close, NULL, - .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, + .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), };