[Drdivx-commits] SF.net SVN: drdivx: [1313] DrDivX/trunk/drffmpeg/libavcodec/mp3lameaudio.c

[<har...@us...>]

Revision: 1313
          http://svn.sourceforge.net/drdivx/?rev=1313&view=rev
Author:   harikrishnan_v
Date:     2007-07-11 16:32:13 -0700 (Wed, 11 Jul 2007)

Log Message:
-----------
mp3lameaudio.c -> libmp3lame.c (renamed in ffmpeg trunk)

Removed Paths:
-------------
    DrDivX/trunk/drffmpeg/libavcodec/mp3lameaudio.c

Deleted: DrDivX/trunk/drffmpeg/libavcodec/mp3lameaudio.c
===================================================================
--- DrDivX/trunk/drffmpeg/libavcodec/mp3lameaudio.c	2007-07-11 23:31:11 UTC (rev 1312)
+++ DrDivX/trunk/drffmpeg/libavcodec/mp3lameaudio.c	2007-07-11 23:32:13 UTC (rev 1313)
@@ -1,231 +0,0 @@
-/*
- * Interface to libmp3lame for mp3 encoding
- * Copyright (c) 2002 Lennert Buytenhek <bu...@gn...>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file mp3lameaudio.c
- * Interface to libmp3lame for mp3 encoding.
- */
-
-#include "avcodec.h"
-#include "mpegaudio.h"
-#include <lame/lame.h>
-
-/* DivX, Inc. FIXME */
-#define BUFFER_SIZE (8*MPA_FRAME_SIZE)
-typedef struct Mp3AudioContext {
-    lame_global_flags *gfp;
-    int stereo;
-    uint8_t buffer[BUFFER_SIZE];
-    int buffer_index;
-} Mp3AudioContext;
-
-static int MP3lame_encode_init(AVCodecContext *avctx)
-{
-    Mp3AudioContext *s = avctx->priv_data;
-
-    if (avctx->channels > 2)
-        return -1;
-
-    s->stereo = avctx->channels > 1 ? 1 : 0;
-
-    if ((s->gfp = lame_init()) == NULL)
-        goto err;
-    lame_set_in_samplerate(s->gfp, avctx->sample_rate);
-    lame_set_out_samplerate(s->gfp, avctx->sample_rate);
-    lame_set_num_channels(s->gfp, avctx->channels);
-    /* lame 3.91 dies on quality != 5 */
-    lame_set_quality(s->gfp, 5);
-    /* lame 3.91 doesn't work in mono */
-    lame_set_mode(s->gfp, JOINT_STEREO);
-    lame_set_brate(s->gfp, avctx->bit_rate/1000);
-    if(avctx->flags & CODEC_FLAG_QSCALE) {
-        lame_set_brate(s->gfp, 0);
-        lame_set_VBR(s->gfp, vbr_default);
-        lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
-    }
-    
-    /* DivX, Inc. FIXME */
-    if(avctx->flags & CODEC_FLAG_ABR) {
-        lame_set_brate(s->gfp, 0);
-        lame_set_VBR(s->gfp, vbr_abr);
-        lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate/1000);
-        lame_set_VBR_q(s->gfp, 2);
-    }
-    lame_set_bWriteVbrTag(s->gfp,0);
-    if (lame_init_params(s->gfp) < 0)
-        goto err_close;
-
-    avctx->frame_size = lame_get_framesize(s->gfp);
-
-    avctx->coded_frame= avcodec_alloc_frame();
-    avctx->coded_frame->key_frame= 1;
-
-    return 0;
-
-err_close:
-    lame_close(s->gfp);
-err:
-    return -1;
-}
-
-static const int sSampleRates[3] = {
-    44100, 48000,  32000
-};
-
-static const int sBitRates[2][3][15] = {
-    {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
-        {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
-        {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
-    },
-    {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
-        {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
-        {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
-    },
-};
-
-static const int sSamplesPerFrame[2][3] =
-{
-    {  384,     1152,    1152 },
-    {  384,     1152,     576 }
-};
-
-static const int sBitsPerSlot[3] = {
-    32,
-    8,
-    8
-};
-
-static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
-{
-    uint8_t *dataTmp = (uint8_t *)data;
-    uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3];
-    int layerID = 3 - ((header >> 17) & 0x03);
-    int bitRateID = ((header >> 12) & 0x0f);
-    int sampleRateID = ((header >> 10) & 0x03);
-    int bitsPerSlot = sBitsPerSlot[layerID];
-    int isPadded = ((header >> 9) & 0x01);
-    static int const mode_tab[4]= {2,3,1,0};
-    int mode= mode_tab[(header >> 19) & 0x03];
-    int mpeg_id= mode>0;
-    int temp0, temp1, bitRate;
-
-    if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
-        return -1;
-    }
-
-    if(!samplesPerFrame) samplesPerFrame= &temp0;
-    if(!sampleRate     ) sampleRate     = &temp1;
-
-//    *isMono = ((header >>  6) & 0x03) == 0x03;
-
-    *sampleRate = sSampleRates[sampleRateID]>>mode;
-    bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
-    *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
-//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
-
-    return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
-}
-
-int MP3lame_encode_frame(AVCodecContext *avctx,
-                     unsigned char *frame, int buf_size, void *data)
-{
-    Mp3AudioContext *s = avctx->priv_data;
-    int len;
-    int lame_result;
-
-    /* lame 3.91 dies on '1-channel interleaved' data */
-
-    if(data){
-        if (s->stereo) {
-            lame_result = lame_encode_buffer_interleaved(
-                s->gfp,
-                data,
-                avctx->frame_size,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
-        } else {
-            lame_result = lame_encode_buffer(
-                s->gfp,
-                data,
-                data,
-                avctx->frame_size,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
-        }
-    }else{
-        lame_result= lame_encode_flush(
-                s->gfp,
-                s->buffer + s->buffer_index,
-                BUFFER_SIZE - s->buffer_index
-                );
-    }
-
-    if(lame_result==-1) {
-        /* output buffer too small */
-        av_log(avctx, AV_LOG_ERROR, "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", s->buffer_index, BUFFER_SIZE - s->buffer_index);
-        return 0;
-    }
-
-    s->buffer_index += lame_result;
-
-    if(s->buffer_index<4)
-        return 0;
-
-        len= mp3len(s->buffer, NULL, NULL);
-//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
-        if(len <= s->buffer_index){
-            memcpy(frame, s->buffer, len);
-            s->buffer_index -= len;
-
-            memmove(s->buffer, s->buffer+len, s->buffer_index);
-            //FIXME fix the audio codec API, so we dont need the memcpy()
-/*for(i=0; i<len; i++){
-    av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
-}*/
-            return len;
-        }else
-            return 0;
-}
-
-int MP3lame_encode_close(AVCodecContext *avctx)
-{
-    Mp3AudioContext *s = avctx->priv_data;
-
-    av_freep(&avctx->coded_frame);
-
-    lame_close(s->gfp);
-    return 0;
-}
-
-
-AVCodec mp3lame_encoder = {
-    "mp3",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_MP3,
-    sizeof(Mp3AudioContext),
-    MP3lame_encode_init,
-    MP3lame_encode_frame,
-    MP3lame_encode_close,
-    NULL,
-    /*.capabilities= */CODEC_CAP_DELAY
-};


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