[Ffmpeg-cvslog] r6958 - in trunk/libavcodec: allcodecs.c avcodec.h bitstream_filter.c

[michael]

Author: michael
Date: Fri Nov 10 02:41:53 2006
New Revision: 6958

Modified:
   trunk/libavcodec/allcodecs.c
   trunk/libavcodec/avcodec.h
   trunk/libavcodec/bitstream_filter.c

Log:
mp3 header (de)compression bitstream filter
this will make mp3 frames 4 bytes smaller, it will not give you binary identical mp3 files, but it will give you mp3 files which decode to binary identical output
this will only work in containers providing at least packet size, sample_rate and number of channels 
bugreports about mp3 files for which this fails are welcome
and this is experimental (dont expect compatibility and dont even expect to be able to decompress what you compressed, hell dont even expect this to work without editing the source a little)


Modified: trunk/libavcodec/allcodecs.c
==============================================================================
--- trunk/libavcodec/allcodecs.c	(original)
+++ trunk/libavcodec/allcodecs.c	Fri Nov 10 02:41:53 2006
@@ -869,5 +869,7 @@
     av_register_bitstream_filter(&dump_extradata_bsf);
     av_register_bitstream_filter(&remove_extradata_bsf);
     av_register_bitstream_filter(&noise_bsf);
+    av_register_bitstream_filter(&mp3_header_compress_bsf);
+    av_register_bitstream_filter(&mp3_header_decompress_bsf);
 }
 

Modified: trunk/libavcodec/avcodec.h
==============================================================================
--- trunk/libavcodec/avcodec.h	(original)
+++ trunk/libavcodec/avcodec.h	Fri Nov 10 02:41:53 2006
@@ -2662,6 +2662,8 @@
 extern AVBitStreamFilter dump_extradata_bsf;
 extern AVBitStreamFilter remove_extradata_bsf;
 extern AVBitStreamFilter noise_bsf;
+extern AVBitStreamFilter mp3_header_compress_bsf;
+extern AVBitStreamFilter mp3_header_decompress_bsf;
 
 
 /* memory */

Modified: trunk/libavcodec/bitstream_filter.c
==============================================================================
--- trunk/libavcodec/bitstream_filter.c	(original)
+++ trunk/libavcodec/bitstream_filter.c	Fri Nov 10 02:41:53 2006
@@ -19,6 +19,7 @@
  */
 
 #include "avcodec.h"
+#include "mpegaudio.h"
 
 AVBitStreamFilter *first_bitstream_filter= NULL;
 
@@ -124,6 +125,112 @@
     return 1;
 }
 
+static int mp3_header_compress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
+                     uint8_t **poutbuf, int *poutbuf_size,
+                     const uint8_t *buf, int buf_size, int keyframe){
+    uint32_t header;
+    int mode_extension;
+
+    if(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL){
+        av_log(avctx, AV_LOG_ERROR, "not standards compliant\n");
+        return -1;
+    }
+
+    header = (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
+    mode_extension= (header>>4)&3;
+
+    if(ff_mpa_check_header(header) < 0 || (header&0x70000) != 0x30000){
+        *poutbuf= (uint8_t *) buf;
+        *poutbuf_size= buf_size;
+
+        av_log(avctx, AV_LOG_INFO, "cannot compress %08X\n", header);
+        return 0;
+    }
+
+    *poutbuf_size= buf_size - 4;
+    *poutbuf= av_malloc(buf_size - 4 + FF_INPUT_BUFFER_PADDING_SIZE);
+    memcpy(*poutbuf, buf + 4, buf_size - 4 + FF_INPUT_BUFFER_PADDING_SIZE);
+
+    if(avctx->channels==2){
+        if((header & (3<<19)) != 3<<19){
+            (*poutbuf)[1] &= 0x3F;
+            (*poutbuf)[1] |= mode_extension<<6;
+            FFSWAP(int, (*poutbuf)[1], (*poutbuf)[2]);
+        }else{
+            (*poutbuf)[1] &= 0x8F;
+            (*poutbuf)[1] |= mode_extension<<4;
+        }
+    }
+
+    return 1;
+}
+
+static int mp3_header_decompress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
+                     uint8_t **poutbuf, int *poutbuf_size,
+                     const uint8_t *buf, int buf_size, int keyframe){
+    uint32_t header;
+    int sample_rate= avctx->sample_rate;
+    int sample_rate_index=0;
+    int lsf, mpeg25, bitrate_index, frame_size;
+
+    header = (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
+    if(ff_mpa_check_header(header) >= 0){
+        *poutbuf= (uint8_t *) buf;
+        *poutbuf_size= buf_size;
+
+        return 0;
+    }
+
+    header= 0xFFE00000 | ((4-3)<<17) | (1<<16); //FIXME simplify
+
+    lsf     = sample_rate < (24000+32000)/2;
+    mpeg25  = sample_rate < (12000+16000)/2;
+    header |= (!mpeg25)<<20;
+    header |= (!lsf   )<<19;
+    if(sample_rate<<(lsf+mpeg25) < (44100+32000)/2)
+        sample_rate_index |= 2;
+    else if(sample_rate<<(lsf+mpeg25) > (44100+48000)/2)
+        sample_rate_index |= 1;
+
+    header |= sample_rate_index<<10;
+    sample_rate= mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
+
+    for(bitrate_index=2; bitrate_index<30; bitrate_index++){
+        frame_size = mpa_bitrate_tab[lsf][2][bitrate_index>>1];
+        frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
+        if(frame_size == buf_size + 4)
+            break;
+    }
+    if(bitrate_index == 30){
+        av_log(avctx, AV_LOG_ERROR, "couldnt find bitrate_index\n");
+        return -1;
+    }
+
+    header |= (bitrate_index&1)<<9;
+    header |= (bitrate_index>>1)<<12;
+    header |= (avctx->channels==1 ? MPA_MONO : MPA_JSTEREO)<<6;
+
+    *poutbuf_size= buf_size + 4;
+    *poutbuf= av_malloc(buf_size + 4 + FF_INPUT_BUFFER_PADDING_SIZE);
+    memcpy(*poutbuf + 4, buf, buf_size + FF_INPUT_BUFFER_PADDING_SIZE);
+
+    if(avctx->channels==2){
+        if(lsf){
+            FFSWAP(int, (*poutbuf)[5], (*poutbuf)[6]);
+            header |= ((*poutbuf)[5] & 0xC0)>>2;
+        }else{
+            header |= (*poutbuf)[5] & 0x30;
+        }
+    }
+
+    (*poutbuf)[0]= header>>24;
+    (*poutbuf)[1]= header>>16;
+    (*poutbuf)[2]= header>> 8;
+    (*poutbuf)[3]= header    ;
+
+    return 1;
+}
+
 AVBitStreamFilter dump_extradata_bsf={
     "dump_extra",
     0,
@@ -141,3 +248,15 @@
     sizeof(int),
     noise,
 };
+
+AVBitStreamFilter mp3_header_compress_bsf={
+    "mp3comp",
+    0,
+    mp3_header_compress,
+};
+
+AVBitStreamFilter mp3_header_decompress_bsf={
+    "mp3decomp",
+    0,
+    mp3_header_decompress,
+};