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audio: Allow channel maps to specify -1 to mute a channel.
Fixes #11373.
This commit is contained in:
@@ -1189,8 +1189,12 @@ extern SDL_DECLSPEC int * SDLCALL SDL_GetAudioStreamOutputChannelMap(SDL_AudioSt
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* channel that it should be remapped to. To reverse a stereo signal's left
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* and right values, you'd have an array of `{ 1, 0 }`. It is legal to remap
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* multiple channels to the same thing, so `{ 1, 1 }` would duplicate the
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* right channel to both channels of a stereo signal. You cannot change the
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* number of channels through a channel map, just reorder them.
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* right channel to both channels of a stereo signal. An element in the
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* channel map set to -1 instead of a valid channel will mute that channel,
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* setting it to a silence value.
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*
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* You cannot change the number of channels through a channel map, just
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* reorder/mute them.
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*
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* Data that was previously queued in the stream will still be operated on in
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* the order that was current when it was added, which is to say you can put
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@@ -1206,7 +1210,7 @@ extern SDL_DECLSPEC int * SDLCALL SDL_GetAudioStreamOutputChannelMap(SDL_AudioSt
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*
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* If `count` is not equal to the current number of channels in the audio
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* stream's format, this will fail. This is a safety measure to make sure a a
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* race condition hasn't changed the format while you this call is setting the
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* race condition hasn't changed the format while this call is setting the
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* channel map.
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*
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* \param stream the SDL_AudioStream to change.
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@@ -1235,12 +1239,16 @@ extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioStreamInputChannelMap(SDL_AudioStre
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* The output channel map reorders data that leaving a stream via
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* SDL_GetAudioStreamData.
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*
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* Each item in the array represents an output channel, and its value is the
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* Each item in the array represents an input channel, and its value is the
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* channel that it should be remapped to. To reverse a stereo signal's left
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* and right values, you'd have an array of `{ 1, 0 }`. It is legal to remap
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* multiple channels to the same thing, so `{ 1, 1 }` would duplicate the
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* right channel to both channels of a stereo signal. You cannot change the
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* number of channels through a channel map, just reorder them.
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* right channel to both channels of a stereo signal. An element in the
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* channel map set to -1 instead of a valid channel will mute that channel,
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* setting it to a silence value.
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*
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* You cannot change the number of channels through a channel map, just
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* reorder/mute them.
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*
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* The output channel map can be changed at any time, as output remapping is
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* applied during SDL_GetAudioStreamData.
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@@ -1253,7 +1261,7 @@ extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioStreamInputChannelMap(SDL_AudioStre
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*
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* If `count` is not equal to the current number of channels in the audio
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* stream's format, this will fail. This is a safety measure to make sure a a
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* race condition hasn't changed the format while you this call is setting the
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* race condition hasn't changed the format while this call is setting the
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* channel map.
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*
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* \param stream the SDL_AudioStream to change.
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@@ -129,7 +129,7 @@ bool SDL_ChannelMapIsBogus(const int *chmap, int channels)
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if (chmap) {
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for (int i = 0; i < channels; i++) {
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const int mapping = chmap[i];
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if ((mapping < 0) || (mapping >= channels)) {
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if ((mapping < -1) || (mapping >= channels)) {
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return true;
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}
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}
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@@ -150,27 +150,60 @@ bool SDL_ChannelMapIsDefault(const int *chmap, int channels)
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}
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// Swizzle audio channels. src and dst can be the same pointer. It does not change the buffer size.
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static void SwizzleAudio(const int num_frames, void *dst, const void *src, int channels, const int *map, int bitsize)
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static void SwizzleAudio(const int num_frames, void *dst, const void *src, int channels, const int *map, SDL_AudioFormat fmt)
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{
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const int bitsize = (int) SDL_AUDIO_BITSIZE(fmt);
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bool has_null_mappings = false;
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for (int i = 0; i < channels; i++) {
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if (map[i] == -1) {
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has_null_mappings = true;
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break;
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}
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}
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#define CHANNEL_SWIZZLE(bits) { \
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Uint##bits *tdst = (Uint##bits *) dst; /* treat as UintX; we only care about moving bits and not the type here. */ \
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const Uint##bits *tsrc = (const Uint##bits *) src; \
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if (src != dst) { /* don't need to copy to a temporary frame first. */ \
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for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
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for (int ch = 0; ch < channels; ch++) { \
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tdst[ch] = tsrc[map[ch]]; \
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if (has_null_mappings) { \
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const Uint##bits silence = (Uint##bits) SDL_GetSilenceValueForFormat(fmt); \
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for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
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for (int ch = 0; ch < channels; ch++) { \
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const int m = map[ch]; \
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tdst[ch] = (m == -1) ? silence : tsrc[m]; \
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} \
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} \
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} else { \
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for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
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for (int ch = 0; ch < channels; ch++) { \
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tdst[ch] = tsrc[map[ch]]; \
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} \
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} \
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} \
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} else { \
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bool isstack; \
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Uint##bits *tmp = (Uint##bits *) SDL_small_alloc(int, channels, &isstack); /* !!! FIXME: allocate this when setting the channel map instead. */ \
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if (tmp) { \
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for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
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for (int ch = 0; ch < channels; ch++) { \
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tmp[ch] = tsrc[map[ch]]; \
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if (has_null_mappings) { \
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const Uint##bits silence = (Uint##bits) SDL_GetSilenceValueForFormat(fmt); \
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for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
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for (int ch = 0; ch < channels; ch++) { \
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const int m = map[ch]; \
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tmp[ch] = (m == -1) ? silence : tsrc[m]; \
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} \
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for (int ch = 0; ch < channels; ch++) { \
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tdst[ch] = tmp[ch]; \
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} \
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} \
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for (int ch = 0; ch < channels; ch++) { \
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tdst[ch] = tmp[ch]; \
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} else { \
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for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \
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for (int ch = 0; ch < channels; ch++) { \
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tmp[ch] = tsrc[map[ch]]; \
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} \
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for (int ch = 0; ch < channels; ch++) { \
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tdst[ch] = tmp[ch]; \
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} \
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} \
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} \
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SDL_small_free(tmp, isstack); \
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@@ -221,9 +254,7 @@ void ConvertAudio(int num_frames,
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SDL_Log("SDL_AUDIO_CONVERT: Convert format %04x->%04x, channels %u->%u", src_format, dst_format, src_channels, dst_channels);
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#endif
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const int src_bitsize = (int) SDL_AUDIO_BITSIZE(src_format);
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const int dst_bitsize = (int) SDL_AUDIO_BITSIZE(dst_format);
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const int dst_sample_frame_size = (dst_bitsize / 8) * dst_channels;
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/* Type conversion goes like this now:
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@@ -245,7 +276,7 @@ void ConvertAudio(int num_frames,
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// swizzle input to "standard" format if necessary.
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if (src_map) {
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void* buf = scratch ? scratch : dst; // use scratch if available, since it has to be big enough to hold src, unless it's NULL, then dst has to be.
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SwizzleAudio(num_frames, buf, src, src_channels, src_map, src_bitsize);
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SwizzleAudio(num_frames, buf, src, src_channels, src_map, src_format);
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src = buf;
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}
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@@ -254,7 +285,7 @@ void ConvertAudio(int num_frames,
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if (src_format == dst_format) {
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// nothing to do, we're already in the right format, just copy it over if necessary.
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if (dst_map) {
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SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_bitsize);
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SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format);
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} else if (src != dst) {
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SDL_memcpy(dst, src, num_frames * dst_sample_frame_size);
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}
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@@ -264,7 +295,7 @@ void ConvertAudio(int num_frames,
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// just a byteswap needed?
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if ((src_format ^ dst_format) == SDL_AUDIO_MASK_BIG_ENDIAN) {
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if (dst_map) { // do this first, in case we duplicate channels, we can avoid an extra copy if src != dst.
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SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_bitsize);
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SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format);
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src = dst;
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}
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ConvertAudioSwapEndian(dst, src, num_frames * dst_channels, dst_bitsize);
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@@ -348,7 +379,7 @@ void ConvertAudio(int num_frames,
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SDL_assert(src == dst); // if we got here, we _had_ to have done _something_. Otherwise, we should have memcpy'd!
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if (dst_map) {
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SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_bitsize);
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SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format);
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}
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}
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