3D Audio GSoC:

- Sequencer dynamics: Now it's possible to change the output channels and the resampling quality also increased (previously maximum quality was 44,1 kHz)
- Changed two buffers to use ffmpeg allocation, not sure if that helps somehow.

3 files changed, 13 insertions, 9 deletions

diff --git a/source/blender/blenkernel/intern/scene.c b/source/blender/blenkernel/intern/scene.c
index 51eaba3c05b..e045bd0fb82 100644
--- a/source/blender/blenkernel/intern/scene.c
+++ b/source/blender/blenkernel/intern/scene.c
@@ -469,6 +469,7 @@ Scene *add_scene(const char *name)
 	sce->r.ffcodecdata.audio_mixrate = 44100;
 	sce->r.ffcodecdata.audio_volume = 1.0f;
 	sce->r.ffcodecdata.audio_bitrate = 192;
+	sce->r.ffcodecdata.audio_channels = 2;
 
 	BLI_strncpy(sce->r.engine, "BLENDER_RENDER", sizeof(sce->r.engine));
 
diff --git a/source/blender/blenkernel/intern/sound.c b/source/blender/blenkernel/intern/sound.c
index 60a24c2eace..9b86b8752db 100644
--- a/source/blender/blenkernel/intern/sound.c
+++ b/source/blender/blenkernel/intern/sound.c
@@ -346,6 +346,7 @@ AUD_Device* sound_mixdown(struct Scene *scene, AUD_DeviceSpecs specs, int start,
 
 	AUD_setDeviceVolume(mixdown, volume);
 
+	AUD_setSequencerSpecs(scene->sound_scene, specs.specs);
 	AUD_freeHandle(AUD_playDevice(mixdown, scene->sound_scene, start / FPS));
 
 	return mixdown;
@@ -405,6 +406,9 @@ static void sound_start_play_scene(struct Scene *scene)
 {
 	if(scene->sound_scene_handle)
 		AUD_stop(scene->sound_scene_handle);
+
+	AUD_setSequencerDeviceSpecs(scene->sound_scene);
+
 	if((scene->sound_scene_handle = AUD_play(scene->sound_scene, 1)))
 		AUD_setLoop(scene->sound_scene_handle, -1);
 }
diff --git a/source/blender/blenkernel/intern/writeffmpeg.c b/source/blender/blenkernel/intern/writeffmpeg.c
index c729565533f..26de31b2b03 100644
--- a/source/blender/blenkernel/intern/writeffmpeg.c
+++ b/source/blender/blenkernel/intern/writeffmpeg.c
@@ -549,7 +549,7 @@ static AVStream* alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
 	c->sample_rate = rd->ffcodecdata.audio_mixrate;
 	c->bit_rate = ffmpeg_audio_bitrate*1000;
 	c->sample_fmt = SAMPLE_FMT_S16;
-	c->channels = 2;
+	c->channels = rd->ffcodecdata.audio_channels;
 	codec = avcodec_find_encoder(c->codec_id);
 	if (!codec) {
 		//XXX error("Couldn't find a valid audio codec");
@@ -574,12 +574,11 @@ static AVStream* alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
 			audio_outbuf_size = c->frame_size * c->channels * sizeof(int16_t) * 4;
 	}
 
-	audio_output_buffer = (uint8_t*)MEM_mallocN(
-		audio_outbuf_size, "FFMPEG audio encoder input buffer");
+	audio_output_buffer = (uint8_t*)av_malloc(
+		audio_outbuf_size);
 
-	audio_input_buffer = (uint8_t*)MEM_mallocN(
-		audio_input_samples * c->channels * sizeof(int16_t),
-		"FFMPEG audio encoder output buffer");
+	audio_input_buffer = (uint8_t*)av_malloc(
+		audio_input_samples * c->channels * sizeof(int16_t));
 
 	audio_time = 0.0f;
 
@@ -701,7 +700,7 @@ static int start_ffmpeg_impl(struct RenderData *rd, int rectx, int recty, Report
 	
 	if (ffmpeg_type == FFMPEG_DV) {
 		fmt->audio_codec = CODEC_ID_PCM_S16LE;
-		if (ffmpeg_audio_codec != CODEC_ID_NONE && rd->ffcodecdata.audio_mixrate != 48000) {
+		if (ffmpeg_audio_codec != CODEC_ID_NONE && rd->ffcodecdata.audio_mixrate != 48000 && rd->ffcodecdata.audio_channels != 2) {
 			BKE_report(reports, RPT_ERROR, "FFMPEG only supports 48khz / stereo audio for DV!");
 			return 0;
 		}
@@ -971,11 +970,11 @@ void end_ffmpeg(void)
 		video_buffer = 0;
 	}
 	if (audio_output_buffer) {
-		MEM_freeN(audio_output_buffer);
+		av_free(audio_output_buffer);
 		audio_output_buffer = 0;
 	}
 	if (audio_input_buffer) {
-		MEM_freeN(audio_input_buffer);
+		av_free(audio_input_buffer);
 		audio_input_buffer = 0;
 	}