[Bf-blender-cvs] SVN commit: /data/svn/bf-blender [52438] trunk/blender/source/blender/ blenkernel/intern/writeffmpeg.c: Patch #33242: ffmpeg AAC/AC3 encoding

[Sergey Sharybin]

Revision: 52438
          http://projects.blender.org/scm/viewvc.php?view=rev&root=bf-blender&revision=52438
Author:   nazgul
Date:     2012-11-21 11:57:35 +0000 (Wed, 21 Nov 2012)
Log Message:
-----------
Patch #33242: ffmpeg AAC/AC3 encoding

Patch by David M (erwin94), thanks!

Also made Vorbis codec using float sample_fmt, otherwise it didn't work
with new FFmpeg.

Perhaps we can make it more clear by explicitly separating audio_input_buffer
for float and integer buffers, but as far as it works i'm not so fussed about
this atm.

Modified Paths:
--------------
    trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c

Modified: trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c
===================================================================
--- trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c	2012-11-21 11:10:37 UTC (rev 52437)
+++ trunk/blender/source/blender/blenkernel/intern/writeffmpeg.c	2012-11-21 11:57:35 UTC (rev 52438)
@@ -111,6 +111,11 @@
 	}
 }
 
+static int use_float_audio_buffer(int codec_id)
+{
+	return codec_id == CODEC_ID_AAC || codec_id == CODEC_ID_AC3 || codec_id == CODEC_ID_VORBIS;
+}
+
 #ifdef WITH_AUDASPACE
 static int write_audio_frame(void) 
 {
@@ -618,6 +623,10 @@
 	c->bit_rate = ffmpeg_audio_bitrate * 1000;
 	c->sample_fmt = AV_SAMPLE_FMT_S16;
 	c->channels = rd->ffcodecdata.audio_channels;
+	if (use_float_audio_buffer(codec_id)) {
+		c->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+		c->sample_fmt = AV_SAMPLE_FMT_FLT;
+	}
 	codec = avcodec_find_encoder(c->codec_id);
 	if (!codec) {
 		//XXX error("Couldn't find a valid audio codec");
@@ -649,7 +658,12 @@
 
 	audio_output_buffer = (uint8_t *) av_malloc(audio_outbuf_size);
 
-	audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * sizeof(int16_t));
+	if (use_float_audio_buffer(codec_id)) {
+		audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * sizeof(float));
+	}
+	else {
+		audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * sizeof(int16_t));
+	}
 
 	audio_time = 0.0f;
 
@@ -949,7 +963,12 @@
 		AVCodecContext *c = audio_stream->codec;
 		AUD_DeviceSpecs specs;
 		specs.channels = c->channels;
-		specs.format = AUD_FORMAT_S16;
+		if (use_float_audio_buffer(c->codec_id)) {
+			specs.format = AUD_FORMAT_FLOAT32;
+		}
+		else {
+			specs.format = AUD_FORMAT_S16;
+		}
 		specs.rate = rd->ffcodecdata.audio_mixrate;
 		audio_mixdown_device = sound_mixdown(scene, specs, rd->sfra, rd->ffcodecdata.audio_volume);
 #ifdef FFMPEG_CODEC_TIME_BASE