diff libmp3lame.c @ 5101:c3f2379b80db libavcodec

Give all wrappers for external libraries names starting with lib.

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/libmp3lame.c	Wed Jun 06 00:14:18 2007 +0000
@@ -0,0 +1,220 @@
+/*
+ * Interface to libmp3lame for mp3 encoding
+ * Copyright (c) 2002 Lennert Buytenhek <[e-mail]>
+ *
+ * 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>
+
+#define BUFFER_SIZE (2*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);
+    }
+    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)
+{
+    uint32_t header = AV_RB32(data);
+    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;
+}
+
+static 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;
+}
+
+static 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,
+    .capabilities= CODEC_CAP_DELAY,
+};