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.
This commit is contained in:
2011-06-21 20:39:41 +00:00
parent 8e6b5598e0
commit 2d3d025e8c
21 changed files with 117 additions and 30 deletions

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@@ -882,16 +882,11 @@ AUD_Handle* AUD_pauseAfter(AUD_Handle* handle, float seconds)
AUD_Sound* AUD_createSequencer(int muted, void* data, AUD_volumeFunction volume) AUD_Sound* AUD_createSequencer(int muted, void* data, AUD_volumeFunction volume)
{ {
/* AUD_XXX should be this: but AUD_createSequencer is called before the device // specs are changed at a later point!
* is initialized.
return new AUD_SequencerFactory(AUD_device->getSpecs().specs, data, volume);
*/
AUD_Specs specs; AUD_Specs specs;
specs.channels = AUD_CHANNELS_STEREO; specs.channels = AUD_CHANNELS_STEREO;
specs.rate = AUD_RATE_44100; specs.rate = AUD_RATE_44100;
AUD_Reference<AUD_SequencerFactory>* sequencer = new AUD_Reference<AUD_SequencerFactory>(new AUD_SequencerFactory(specs, muted, data, volume)); AUD_Reference<AUD_SequencerFactory>* sequencer = new AUD_Reference<AUD_SequencerFactory>(new AUD_SequencerFactory(specs, muted, data, volume));
(*sequencer)->setThis(sequencer);
return reinterpret_cast<AUD_Sound*>(sequencer); return reinterpret_cast<AUD_Sound*>(sequencer);
} }
@@ -928,6 +923,16 @@ void AUD_muteSequencer(AUD_Sound* sequencer, AUD_Reference<AUD_SequencerEntry>*
((AUD_SequencerFactory*)sequencer->get())->mute(*entry, mute); ((AUD_SequencerFactory*)sequencer->get())->mute(*entry, mute);
} }
void AUD_setSequencerDeviceSpecs(AUD_Sound* sequencer)
{
((AUD_SequencerFactory*)sequencer->get())->setSpecs(AUD_device->getSpecs().specs);
}
void AUD_setSequencerSpecs(AUD_Sound* sequencer, AUD_Specs specs)
{
((AUD_SequencerFactory*)sequencer->get())->setSpecs(specs);
}
int AUD_readSound(AUD_Sound* sound, sample_t* buffer, int length) int AUD_readSound(AUD_Sound* sound, sample_t* buffer, int length)
{ {
AUD_DeviceSpecs specs; AUD_DeviceSpecs specs;

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@@ -469,6 +469,10 @@ extern void AUD_moveSequencer(AUD_Sound* sequencer, AUD_SEntry* entry,
extern void AUD_muteSequencer(AUD_Sound* sequencer, AUD_SEntry* entry, extern void AUD_muteSequencer(AUD_Sound* sequencer, AUD_SEntry* entry,
char mute); char mute);
extern void AUD_setSequencerDeviceSpecs(AUD_Sound* sequencer);
extern void AUD_setSequencerSpecs(AUD_Sound* sequencer, AUD_Specs specs);
extern int AUD_readSound(AUD_Sound* sound, sample_t* buffer, int length); extern int AUD_readSound(AUD_Sound* sound, sample_t* buffer, int length);
extern void AUD_startPlayback(void); extern void AUD_startPlayback(void);

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@@ -44,6 +44,12 @@ AUD_ChannelMapperReader::~AUD_ChannelMapperReader()
delete[] m_mapping; delete[] m_mapping;
} }
void AUD_ChannelMapperReader::setChannels(AUD_Channels channels)
{
m_target_channels = channels;
calculateMapping();
}
float AUD_ChannelMapperReader::angleDistance(float alpha, float beta) float AUD_ChannelMapperReader::angleDistance(float alpha, float beta)
{ {
alpha = fabs(alpha - beta); alpha = fabs(alpha - beta);

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@@ -109,6 +109,8 @@ public:
*/ */
~AUD_ChannelMapperReader(); ~AUD_ChannelMapperReader();
void setChannels(AUD_Channels channels);
virtual AUD_Specs getSpecs() const; virtual AUD_Specs getSpecs() const;
virtual void read(int& length, bool& eos, sample_t* buffer); virtual void read(int& length, bool& eos, sample_t* buffer);
}; };

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@@ -73,6 +73,11 @@ AUD_DeviceSpecs AUD_Mixer::getSpecs() const
return m_specs; return m_specs;
} }
void AUD_Mixer::setSpecs(AUD_Specs specs)
{
m_specs.specs = specs;
}
void AUD_Mixer::clear(int length) void AUD_Mixer::clear(int length)
{ {
m_buffer.assureSize(length * m_specs.channels * AUD_SAMPLE_SIZE(m_specs)); m_buffer.assureSize(length * m_specs.channels * AUD_SAMPLE_SIZE(m_specs));

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@@ -47,7 +47,7 @@ protected:
/** /**
* The output specification. * The output specification.
*/ */
const AUD_DeviceSpecs m_specs; AUD_DeviceSpecs m_specs;
/** /**
* The length of the mixing buffer. * The length of the mixing buffer.
@@ -81,6 +81,12 @@ public:
*/ */
AUD_DeviceSpecs getSpecs() const; AUD_DeviceSpecs getSpecs() const;
/**
* Sets the target specification for superposing.
* \param specs The target specification.
*/
void setSpecs(AUD_Specs specs);
/** /**
* Mixes a buffer. * Mixes a buffer.
* \param buffer The buffer to superpose. * \param buffer The buffer to superpose.

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@@ -48,9 +48,12 @@ AUD_SequencerFactory::~AUD_SequencerFactory()
{ {
} }
void AUD_SequencerFactory::setThis(AUD_Reference<AUD_SequencerFactory>* self) void AUD_SequencerFactory::setSpecs(AUD_Specs specs)
{ {
m_this = self; m_specs = specs;
for(AUD_ReaderIterator i = m_readers.begin(); i != m_readers.end(); i++)
(*i)->setSpecs(m_specs);
} }
void AUD_SequencerFactory::mute(bool muted) void AUD_SequencerFactory::mute(bool muted)
@@ -103,7 +106,7 @@ void AUD_SequencerFactory::mute(AUD_Reference<AUD_SequencerEntry> entry, bool mu
AUD_Reference<AUD_IReader> AUD_SequencerFactory::createReader() AUD_Reference<AUD_IReader> AUD_SequencerFactory::createReader()
{ {
AUD_Reference<AUD_SequencerReader> reader = new AUD_SequencerReader(*m_this, m_entries, AUD_Reference<AUD_SequencerReader> reader = new AUD_SequencerReader(this, m_entries,
m_specs, m_data, m_specs, m_data,
m_volume); m_volume);
m_readers.push_front(reader); m_readers.push_front(reader);

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@@ -66,7 +66,6 @@ private:
bool m_muted; bool m_muted;
void* m_data; void* m_data;
AUD_volumeFunction m_volume; AUD_volumeFunction m_volume;
AUD_Reference<AUD_SequencerFactory>* m_this;
// hide copy constructor and operator= // hide copy constructor and operator=
AUD_SequencerFactory(const AUD_SequencerFactory&); AUD_SequencerFactory(const AUD_SequencerFactory&);
@@ -76,7 +75,7 @@ public:
AUD_SequencerFactory(AUD_Specs specs, bool muted, void* data, AUD_volumeFunction volume); AUD_SequencerFactory(AUD_Specs specs, bool muted, void* data, AUD_volumeFunction volume);
~AUD_SequencerFactory(); ~AUD_SequencerFactory();
void setThis(AUD_Reference<AUD_SequencerFactory>* self); void setSpecs(AUD_Specs specs);
void mute(bool muted); void mute(bool muted);
bool getMute() const; bool getMute() const;

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@@ -95,6 +95,22 @@ void AUD_SequencerReader::remove(AUD_Reference<AUD_SequencerEntry> entry)
} }
} }
void AUD_SequencerReader::setSpecs(AUD_Specs specs)
{
m_mixer->setSpecs(specs);
AUD_Reference<AUD_SequencerStrip> strip;
for(AUD_StripIterator i = m_strips.begin(); i != m_strips.end(); i++)
{
strip = *i;
if(!strip->mapper.isNull())
{
strip->mapper->setChannels(specs.channels);
strip->resampler->setRate(specs.rate);
}
}
}
bool AUD_SequencerReader::isSeekable() const bool AUD_SequencerReader::isSeekable() const
{ {
return true; return true;
@@ -149,24 +165,30 @@ void AUD_SequencerReader::read(int& length, bool& eos, sample_t* buffer)
strip->reader = (*strip->old_sound)->createReader(); strip->reader = (*strip->old_sound)->createReader();
// resample // resample
#ifdef WITH_SAMPLERATE #ifdef WITH_SAMPLERATE
strip->reader = new AUD_SRCResampleReader(strip->reader, m_mixer->getSpecs().specs); strip->resampler = new AUD_SRCResampleReader(strip->reader, m_mixer->getSpecs().specs);
#else #else
strip->reader = new AUD_LinearResampleReader(strip->reader, m_mixer->getSpecs().specs); strip->resampler = new AUD_LinearResampleReader(strip->reader, m_mixer->getSpecs().specs);
#endif #endif
// rechannel // rechannel
strip->reader = new AUD_ChannelMapperReader(strip->reader, m_mixer->getSpecs().channels); strip->mapper = new AUD_ChannelMapperReader(AUD_Reference<AUD_IReader>(strip->resampler), m_mixer->getSpecs().channels);
} }
catch(AUD_Exception) catch(AUD_Exception)
{ {
strip->reader = NULL; strip->reader = NULL;
strip->resampler = NULL;
strip->mapper = NULL;
} }
} }
else else
{
strip->reader = NULL; strip->reader = NULL;
strip->resampler = NULL;
strip->mapper = NULL;
}
} }
if(!strip->reader.isNull()) if(!strip->mapper.isNull())
{ {
end = floor(strip->entry->end * rate); end = floor(strip->entry->end * rate);
if(m_position < end) if(m_position < end)
@@ -185,9 +207,9 @@ void AUD_SequencerReader::read(int& length, bool& eos, sample_t* buffer)
current += strip->entry->skip * rate; current += strip->entry->skip * rate;
len = length > end - m_position ? end - m_position : length; len = length > end - m_position ? end - m_position : length;
len -= skip; len -= skip;
if(strip->reader->getPosition() != current) if(strip->mapper->getPosition() != current)
strip->reader->seek(current); strip->mapper->seek(current);
strip->reader->read(len, eos, m_buffer.getBuffer()); strip->mapper->read(len, eos, m_buffer.getBuffer());
m_mixer->mix(m_buffer.getBuffer(), skip, len, m_volume(m_data, strip->entry->data, (float)m_position / (float)rate)); m_mixer->mix(m_buffer.getBuffer(), skip, len, m_volume(m_data, strip->entry->data, (float)m_position / (float)rate));
} }
} }

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@@ -35,11 +35,15 @@
#include "AUD_IReader.h" #include "AUD_IReader.h"
#include "AUD_SequencerFactory.h" #include "AUD_SequencerFactory.h"
#include "AUD_Buffer.h" #include "AUD_Buffer.h"
class AUD_Mixer; #include "AUD_Mixer.h"
#include "AUD_ResampleReader.h"
#include "AUD_ChannelMapperReader.h"
struct AUD_SequencerStrip struct AUD_SequencerStrip
{ {
AUD_Reference<AUD_IReader> reader; AUD_Reference<AUD_IReader> reader;
AUD_Reference<AUD_ResampleReader> resampler;
AUD_Reference<AUD_ChannelMapperReader> mapper;
AUD_Reference<AUD_SequencerEntry> entry; AUD_Reference<AUD_SequencerEntry> entry;
AUD_Reference<AUD_IFactory>* old_sound; AUD_Reference<AUD_IFactory>* old_sound;
}; };
@@ -94,6 +98,7 @@ public:
void add(AUD_Reference<AUD_SequencerEntry> entry); void add(AUD_Reference<AUD_SequencerEntry> entry);
void remove(AUD_Reference<AUD_SequencerEntry> entry); void remove(AUD_Reference<AUD_SequencerEntry> entry);
void setSpecs(AUD_Specs specs);
virtual bool isSeekable() const; virtual bool isSeekable() const;
virtual void seek(int position); virtual void seek(int position);

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@@ -11,3 +11,4 @@ else:
bpy.context.scene.render.ffmpeg_audio_mixrate = 48000 bpy.context.scene.render.ffmpeg_audio_mixrate = 48000
bpy.context.scene.render.ffmpeg_audio_codec = "PCM" bpy.context.scene.render.ffmpeg_audio_codec = "PCM"
bpy.context.scene.render.ffmpeg_audio_channels = 2

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@@ -21,3 +21,4 @@ bpy.context.scene.render.ffmpeg_muxrate = 10080000
bpy.context.scene.render.ffmpeg_audio_codec = "AC3" bpy.context.scene.render.ffmpeg_audio_codec = "AC3"
bpy.context.scene.render.ffmpeg_audio_bitrate = 448 bpy.context.scene.render.ffmpeg_audio_bitrate = 448
bpy.context.scene.render.ffmpeg_audio_mixrate = 48000 bpy.context.scene.render.ffmpeg_audio_mixrate = 48000
bpy.context.scene.render.ffmpeg_audio_channels = 6

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@@ -21,3 +21,4 @@ bpy.context.scene.render.ffmpeg_muxrate = 0
bpy.context.scene.render.ffmpeg_audio_bitrate = 224 bpy.context.scene.render.ffmpeg_audio_bitrate = 224
bpy.context.scene.render.ffmpeg_audio_mixrate = 44100 bpy.context.scene.render.ffmpeg_audio_mixrate = 44100
bpy.context.scene.render.ffmpeg_audio_codec = "MP2" bpy.context.scene.render.ffmpeg_audio_codec = "MP2"
bpy.context.scene.render.ffmpeg_audio_channels = 2

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@@ -21,3 +21,4 @@ bpy.context.scene.render.ffmpeg_muxrate = 2352 * 75 * 8
bpy.context.scene.render.ffmpeg_audio_bitrate = 224 bpy.context.scene.render.ffmpeg_audio_bitrate = 224
bpy.context.scene.render.ffmpeg_audio_mixrate = 44100 bpy.context.scene.render.ffmpeg_audio_mixrate = 44100
bpy.context.scene.render.ffmpeg_audio_codec = "MP2" bpy.context.scene.render.ffmpeg_audio_codec = "MP2"
bpy.context.scene.render.ffmpeg_audio_channels = 2

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@@ -596,7 +596,9 @@ class RENDER_PT_encoding(RenderButtonsPanel, bpy.types.Panel):
col.prop(rd, "ffmpeg_audio_bitrate") col.prop(rd, "ffmpeg_audio_bitrate")
col.prop(rd, "ffmpeg_audio_mixrate") col.prop(rd, "ffmpeg_audio_mixrate")
split.prop(rd, "ffmpeg_audio_volume", slider=True) col = split.column()
col.prop(rd, "ffmpeg_audio_volume", slider=True)
col.prop(rd, "ffmpeg_audio_channels")
class RENDER_PT_bake(RenderButtonsPanel, bpy.types.Panel): class RENDER_PT_bake(RenderButtonsPanel, bpy.types.Panel):

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@@ -469,6 +469,7 @@ Scene *add_scene(const char *name)
sce->r.ffcodecdata.audio_mixrate = 44100; sce->r.ffcodecdata.audio_mixrate = 44100;
sce->r.ffcodecdata.audio_volume = 1.0f; sce->r.ffcodecdata.audio_volume = 1.0f;
sce->r.ffcodecdata.audio_bitrate = 192; sce->r.ffcodecdata.audio_bitrate = 192;
sce->r.ffcodecdata.audio_channels = 2;
BLI_strncpy(sce->r.engine, "BLENDER_RENDER", sizeof(sce->r.engine)); BLI_strncpy(sce->r.engine, "BLENDER_RENDER", sizeof(sce->r.engine));

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@@ -346,6 +346,7 @@ AUD_Device* sound_mixdown(struct Scene *scene, AUD_DeviceSpecs specs, int start,
AUD_setDeviceVolume(mixdown, volume); AUD_setDeviceVolume(mixdown, volume);
AUD_setSequencerSpecs(scene->sound_scene, specs.specs);
AUD_freeHandle(AUD_playDevice(mixdown, scene->sound_scene, start / FPS)); AUD_freeHandle(AUD_playDevice(mixdown, scene->sound_scene, start / FPS));
return mixdown; return mixdown;
@@ -405,6 +406,9 @@ static void sound_start_play_scene(struct Scene *scene)
{ {
if(scene->sound_scene_handle) if(scene->sound_scene_handle)
AUD_stop(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))) if((scene->sound_scene_handle = AUD_play(scene->sound_scene, 1)))
AUD_setLoop(scene->sound_scene_handle, -1); AUD_setLoop(scene->sound_scene_handle, -1);
} }

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@@ -549,7 +549,7 @@ static AVStream* alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
c->sample_rate = rd->ffcodecdata.audio_mixrate; c->sample_rate = rd->ffcodecdata.audio_mixrate;
c->bit_rate = ffmpeg_audio_bitrate*1000; c->bit_rate = ffmpeg_audio_bitrate*1000;
c->sample_fmt = SAMPLE_FMT_S16; c->sample_fmt = SAMPLE_FMT_S16;
c->channels = 2; c->channels = rd->ffcodecdata.audio_channels;
codec = avcodec_find_encoder(c->codec_id); codec = avcodec_find_encoder(c->codec_id);
if (!codec) { if (!codec) {
//XXX error("Couldn't find a valid audio 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_outbuf_size = c->frame_size * c->channels * sizeof(int16_t) * 4;
} }
audio_output_buffer = (uint8_t*)MEM_mallocN( audio_output_buffer = (uint8_t*)av_malloc(
audio_outbuf_size, "FFMPEG audio encoder input buffer"); audio_outbuf_size);
audio_input_buffer = (uint8_t*)MEM_mallocN( audio_input_buffer = (uint8_t*)av_malloc(
audio_input_samples * c->channels * sizeof(int16_t), audio_input_samples * c->channels * sizeof(int16_t));
"FFMPEG audio encoder output buffer");
audio_time = 0.0f; 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) { if (ffmpeg_type == FFMPEG_DV) {
fmt->audio_codec = CODEC_ID_PCM_S16LE; 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!"); BKE_report(reports, RPT_ERROR, "FFMPEG only supports 48khz / stereo audio for DV!");
return 0; return 0;
} }
@@ -971,11 +970,11 @@ void end_ffmpeg(void)
video_buffer = 0; video_buffer = 0;
} }
if (audio_output_buffer) { if (audio_output_buffer) {
MEM_freeN(audio_output_buffer); av_free(audio_output_buffer);
audio_output_buffer = 0; audio_output_buffer = 0;
} }
if (audio_input_buffer) { if (audio_input_buffer) {
MEM_freeN(audio_input_buffer); av_free(audio_input_buffer);
audio_input_buffer = 0; audio_input_buffer = 0;
} }

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@@ -11474,6 +11474,12 @@ static void do_versions(FileData *fd, Library *lib, Main *main)
kb->slidermax = kb->slidermin + 1.0f; kb->slidermax = kb->slidermin + 1.0f;
} }
} }
{
Scene *scene;
for (scene=main->scene.first; scene; scene=scene->id.next)
scene->r.ffcodecdata.audio_channels = 2;
}
} }
if (main->versionfile < 256 || (main->versionfile == 256 && main->subversionfile < 1)) { if (main->versionfile < 256 || (main->versionfile == 256 && main->subversionfile < 1)) {

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@@ -126,6 +126,8 @@ typedef struct FFMpegCodecData {
int video_bitrate; int video_bitrate;
int audio_bitrate; int audio_bitrate;
int audio_mixrate; int audio_mixrate;
int audio_channels;
int audio_pad;
float audio_volume; float audio_volume;
int gop_size; int gop_size;
int flags; int flags;

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@@ -98,6 +98,14 @@ EnumPropertyItem snap_element_items[] = {
{SCE_SNAP_MODE_VOLUME, "VOLUME", ICON_SNAP_VOLUME, "Volume", "Snap to volume"}, {SCE_SNAP_MODE_VOLUME, "VOLUME", ICON_SNAP_VOLUME, "Volume", "Snap to volume"},
{0, NULL, 0, NULL, NULL}}; {0, NULL, 0, NULL, NULL}};
static EnumPropertyItem audio_channel_items[] = {
{1, "MONO", 0, "Mono", "Set audio channels to mono"},
{2, "STEREO", 0, "Stereo", "Set audio channels to stereo"},
{4, "SURROUND4", 0, "4 Channels", "Set audio channels to 4 channels"},
{6, "SURROUND51", 0, "5.1 Surround", "Set audio channels to 5.1 surround sound"},
{8, "SURROUND71", 0, "7.1 Surround", "Set audio channels to 7.1 surround sound"},
{0, NULL, 0, NULL, NULL}};
EnumPropertyItem image_type_items[] = { EnumPropertyItem image_type_items[] = {
{0, "", 0, "Image", NULL}, {0, "", 0, "Image", NULL},
{R_BMP, "BMP", ICON_FILE_IMAGE, "BMP", "Output image in bitmap format"}, {R_BMP, "BMP", ICON_FILE_IMAGE, "BMP", "Output image in bitmap format"},
@@ -2468,6 +2476,10 @@ static void rna_def_scene_render_data(BlenderRNA *brna)
RNA_def_property_ui_text(prop, "Volume", "Audio volume"); RNA_def_property_ui_text(prop, "Volume", "Audio volume");
RNA_def_property_update(prop, NC_SCENE|ND_RENDER_OPTIONS, NULL); RNA_def_property_update(prop, NC_SCENE|ND_RENDER_OPTIONS, NULL);
prop= RNA_def_property(srna, "ffmpeg_audio_channels", PROP_ENUM, PROP_NONE);
RNA_def_property_enum_sdna(prop, NULL, "ffcodecdata.audio_channels");
RNA_def_property_enum_items(prop, audio_channel_items);
RNA_def_property_ui_text(prop, "Audio Channels", "Sets the audio channel count");
#endif #endif
prop= RNA_def_property(srna, "fps", PROP_INT, PROP_NONE); prop= RNA_def_property(srna, "fps", PROP_INT, PROP_NONE);