vbr audio encode patch by (Justin Ruggles: jruggle, earthlink net)

with changes by me
    int->float as video uses float too
    remove silent cliping to some per codec range, this should result in an error instead
    remove change to utils.c as its inconsistant with video

Originally committed as revision 4533 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/ffmpeg.c b/ffmpeg.c
index 945a81d..d3e59a7 100644 (file)
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -217,6 +217,8 @@ static int gop_size = 12;
 static int intra_only = 0;
 static int audio_sample_rate = 44100;
 static int audio_bit_rate = 64000;
+#define QSCALE_NONE -99999
+static float audio_qscale = QSCALE_NONE;
 static int audio_disable = 0;
 static int audio_channels = 1;
 static int audio_codec_id = CODEC_ID_NONE;
@@ -3501,6 +3503,10 @@ static void new_audio_stream(AVFormatContext *oc)
         audio_enc->codec_id = codec_id;
         
         audio_enc->bit_rate = audio_bit_rate;
+        if (audio_qscale > QSCALE_NONE) {
+            audio_enc->flags |= CODEC_FLAG_QSCALE;
+            audio_enc->global_quality = st->quality = FF_QP2LAMBDA * audio_qscale;
+        }
         audio_enc->strict_std_compliance = strict;
         audio_enc->thread_count = thread_count;
         /* For audio codecs other than AC3 or DTS we limit */
@@ -4348,6 +4354,7 @@ const OptionDef options[] = {
 
     /* audio options */
     { "ab", HAS_ARG | OPT_AUDIO, {(void*)opt_audio_bitrate}, "set audio bitrate (in kbit/s)", "bitrate", },
+    { "aq", OPT_FLOAT | HAS_ARG | OPT_AUDIO, {(void*)&audio_qscale}, "set audio quality (codec-specific)", "quality", },
     { "ar", HAS_ARG | OPT_AUDIO, {(void*)opt_audio_rate}, "set audio sampling rate (in Hz)", "rate" },
     { "ac", HAS_ARG | OPT_AUDIO, {(void*)opt_audio_channels}, "set number of audio channels", "channels" },
     { "an", OPT_BOOL | OPT_AUDIO, {(void*)&audio_disable}, "disable audio" },

diff --git a/libavcodec/faac.c b/libavcodec/faac.c
index a65bdef..a49cce9 100644 (file)
--- a/libavcodec/faac.c
+++ b/libavcodec/faac.c
@@ -56,7 +56,11 @@ static int Faac_encode_init(AVCodecContext *avctx)
     faac_cfg->mpegVersion = MPEG4;
     faac_cfg->useTns = 0;
     faac_cfg->allowMidside = 1;
-    faac_cfg->bitRate = avctx->bit_rate;
+    faac_cfg->bitRate = avctx->bit_rate / avctx->channels;
+    if(avctx->flags & CODEC_FLAG_QSCALE) {
+        faac_cfg->bitRate = 0;
+        faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA;
+    }
     faac_cfg->outputFormat = 0;
     faac_cfg->inputFormat = FAAC_INPUT_16BIT;
 

diff --git a/libavcodec/mp3lameaudio.c b/libavcodec/mp3lameaudio.c
index 3f10a10..26764fc 100644 (file)
--- a/libavcodec/mp3lameaudio.c
+++ b/libavcodec/mp3lameaudio.c
@@ -53,6 +53,11 @@ static int MP3lame_encode_init(AVCodecContext *avctx)
        /* 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;