Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead

[ffmpeg] / libavcodec / libmp3lame.c

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 13a658a1f17cf685a9dcf8e5327b91da32f75de8..62b55c933799713638a247973d28ec6153ec9cf5 100644 (file)
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -2,25 +2,25 @@
  * Interface to libmp3lame for mp3 encoding
  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
- * @file mp3lameaudio.c
+ * @file
  * Interface to libmp3lame for mp3 encoding.
  */
 
@@ -28,7 +28,7 @@
 #include "mpegaudio.h"
 #include <lame/lame.h>
 
-#define BUFFER_SIZE (7200 + MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
+#define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
 typedef struct Mp3AudioContext {
     lame_global_flags *gfp;
     int stereo;
@@ -50,15 +50,17 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
     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);
+    if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
+        lame_set_quality(s->gfp, 5);
+    } else {
+        lame_set_quality(s->gfp, avctx->compression_level);
+    }
+    lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
     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_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
     }
     lame_set_bWriteVbrTag(s->gfp,0);
     lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
@@ -78,8 +80,8 @@ err:
     return -1;
 }
 
-static const int sSampleRates[3] = {
-    44100, 48000,  32000
+static const int sSampleRates[] = {
+    44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
 };
 
 static const int sBitRates[2][3][15] = {
@@ -171,10 +173,12 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
                 );
     }
 
-    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;
+    if(lame_result < 0){
+        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 -1;
     }
 
     s->buffer_index += lame_result;
@@ -182,20 +186,20 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
     if(s->buffer_index<4)
         return 0;
 
-        len= mp3len(s->buffer, NULL, NULL);
+    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;
+    if(len <= s->buffer_index){
+        memcpy(frame, s->buffer, len);
+        s->buffer_index -= len;
 
-            memmove(s->buffer, s->buffer+len, s->buffer_index);
+        memmove(s->buffer, s->buffer+len, s->buffer_index);
             //FIXME fix the audio codec API, so we do not need the memcpy()
 /*for(i=0; i<len; i++){
     av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
 }*/
-            return len;
-        }else
-            return 0;
+        return len;
+    }else
+        return 0;
 }
 
 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
@@ -209,14 +213,16 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
 }
 
 
-AVCodec libmp3lame_encoder = {
+AVCodec ff_libmp3lame_encoder = {
     "libmp3lame",
-    CODEC_TYPE_AUDIO,
+    AVMEDIA_TYPE_AUDIO,
     CODEC_ID_MP3,
     sizeof(Mp3AudioContext),
     MP3lame_encode_init,
     MP3lame_encode_frame,
     MP3lame_encode_close,
     .capabilities= CODEC_CAP_DELAY,
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+    .supported_samplerates= sSampleRates,
     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
 };