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audio: Add channel remapping to SDL_AudioSpec and SDL_AudioStream.
Fixes #8367.
This commit is contained in:
Ryan C. Gordon
@@ -242,108 +242,22 @@ static const char *get_audio_device(void *handle, const int channels)
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return dev->name;
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}
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// !!! FIXME: is there a channel swizzler in alsalib instead?
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// Swizzle channels to match SDL defaults.
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// These are swizzles _from_ SDL's layouts to what ALSA wants.
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// 5.1 swizzle:
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// https://bugzilla.libsdl.org/show_bug.cgi?id=110
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// "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
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// and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
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#define SWIZ6(T) \
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static void swizzle_alsa_channels_6_##T(void *buffer, const Uint32 bufferlen) \
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{ \
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T *ptr = (T *)buffer; \
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Uint32 i; \
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for (i = 0; i < bufferlen; i++, ptr += 6) { \
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T tmp; \
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tmp = ptr[2]; \
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ptr[2] = ptr[4]; \
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ptr[4] = tmp; \
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tmp = ptr[3]; \
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ptr[3] = ptr[5]; \
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ptr[5] = tmp; \
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} \
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}
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// !!! FIXME: is there a channel swizzler in alsalib instead?
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// !!! FIXME: this screams for a SIMD shuffle operation.
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static const Uint8 swizzle_alsa_channels_6[6] = { 0, 1, 4, 5, 2, 3 };
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// 7.1 swizzle:
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// https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/mapping-stream-formats-to-speaker-configurations
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// For Linux ALSA, this appears to be FL-FR-RL-RR-C-LFE-SL-SR
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// and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-SL-SR-RL-RR"
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#define SWIZ8(T) \
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static void swizzle_alsa_channels_8_##T(void *buffer, const Uint32 bufferlen) \
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{ \
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T *ptr = (T *)buffer; \
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Uint32 i; \
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for (i = 0; i < bufferlen; i++, ptr += 6) { \
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const T center = ptr[2]; \
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const T subwoofer = ptr[3]; \
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const T side_left = ptr[4]; \
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const T side_right = ptr[5]; \
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const T rear_left = ptr[6]; \
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const T rear_right = ptr[7]; \
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ptr[2] = rear_left; \
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ptr[3] = rear_right; \
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ptr[4] = center; \
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ptr[5] = subwoofer; \
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ptr[6] = side_left; \
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ptr[7] = side_right; \
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} \
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}
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static const Uint8 swizzle_alsa_channels_8[8] = { 0, 1, 6, 7, 2, 3, 4, 5 };
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#define CHANNEL_SWIZZLE(x) \
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x(Uint64) \
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x(Uint32) \
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x(Uint16) \
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x(Uint8)
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CHANNEL_SWIZZLE(SWIZ6)
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CHANNEL_SWIZZLE(SWIZ8)
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#undef CHANNEL_SWIZZLE
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#undef SWIZ6
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#undef SWIZ8
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// Called right before feeding device->hidden->mixbuf to the hardware. Swizzle
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// channels from Windows/Mac order to the format alsalib will want.
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static void swizzle_alsa_channels(SDL_AudioDevice *device, void *buffer, Uint32 bufferlen)
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{
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switch (device->spec.channels) {
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#define CHANSWIZ(chans) \
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case chans: \
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switch ((device->spec.format & (0xFF))) { \
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case 8: \
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swizzle_alsa_channels_##chans##_Uint8(buffer, bufferlen); \
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break; \
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case 16: \
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swizzle_alsa_channels_##chans##_Uint16(buffer, bufferlen); \
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break; \
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case 32: \
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swizzle_alsa_channels_##chans##_Uint32(buffer, bufferlen); \
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break; \
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case 64: \
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swizzle_alsa_channels_##chans##_Uint64(buffer, bufferlen); \
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break; \
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default: \
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SDL_assert(!"unhandled bitsize"); \
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break; \
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} \
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return;
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CHANSWIZ(6);
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CHANSWIZ(8);
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#undef CHANSWIZ
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default:
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break;
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}
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}
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#ifdef SND_CHMAP_API_VERSION
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// Some devices have the right channel map, no swizzling necessary
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static void no_swizzle(SDL_AudioDevice *device, void *buffer, Uint32 bufferlen)
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{
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}
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#endif // SND_CHMAP_API_VERSION
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// This function waits until it is possible to write a full sound buffer
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static int ALSA_WaitDevice(SDL_AudioDevice *device)
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@@ -380,8 +294,6 @@ static int ALSA_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buf
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const int frame_size = SDL_AUDIO_FRAMESIZE(device->spec);
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snd_pcm_uframes_t frames_left = (snd_pcm_uframes_t) (buflen / frame_size);
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device->hidden->swizzle_func(device, sample_buf, frames_left);
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while ((frames_left > 0) && !SDL_AtomicGet(&device->shutdown)) {
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const int rc = ALSA_snd_pcm_writei(device->hidden->pcm_handle, sample_buf, frames_left);
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//SDL_LogInfo(SDL_LOG_CATEGORY_AUDIO, "ALSA PLAYDEVICE: WROTE %d of %d bytes", (rc >= 0) ? ((int) (rc * frame_size)) : rc, (int) (frames_left * frame_size));
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@@ -447,8 +359,6 @@ static int ALSA_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
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return -1;
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}
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return 0; // go back to WaitDevice and try again.
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} else if (rc > 0) {
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device->hidden->swizzle_func(device, buffer, total_frames - rc);
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}
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//SDL_LogInfo(SDL_LOG_CATEGORY_AUDIO, "ALSA: recorded %d bytes", rc * frame_size);
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@@ -611,23 +521,6 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
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}
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device->spec.format = test_format;
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// Validate number of channels and determine if swizzling is necessary.
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// Assume original swizzling, until proven otherwise.
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device->hidden->swizzle_func = swizzle_alsa_channels;
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#ifdef SND_CHMAP_API_VERSION
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snd_pcm_chmap_t *chmap = ALSA_snd_pcm_get_chmap(pcm_handle);
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if (chmap) {
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char chmap_str[64];
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if (ALSA_snd_pcm_chmap_print(chmap, sizeof(chmap_str), chmap_str) > 0) {
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if (SDL_strcmp("FL FR FC LFE RL RR", chmap_str) == 0 ||
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SDL_strcmp("FL FR FC LFE SL SR", chmap_str) == 0) {
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device->hidden->swizzle_func = no_swizzle;
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}
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}
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free(chmap); // This should NOT be SDL_free()
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}
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#endif // SND_CHMAP_API_VERSION
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// Set the number of channels
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status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
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device->spec.channels);
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@@ -640,6 +533,33 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device)
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device->spec.channels = channels;
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}
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// Validate number of channels and determine if swizzling is necessary.
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// Assume original swizzling, until proven otherwise.
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if (channels == 6) {
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device->spec.use_channel_map = SDL_TRUE;
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SDL_memcpy(device->spec.channel_map, swizzle_alsa_channels_6, sizeof (device->spec.channel_map[0]) * channels);
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} else if (channels == 8) {
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device->spec.use_channel_map = SDL_TRUE;
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SDL_memcpy(device->spec.channel_map, swizzle_alsa_channels_8, sizeof (device->spec.channel_map[0]) * channels);
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}
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#ifdef SND_CHMAP_API_VERSION
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snd_pcm_chmap_t *chmap = ALSA_snd_pcm_get_chmap(pcm_handle);
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if (chmap) {
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char chmap_str[64];
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if (ALSA_snd_pcm_chmap_print(chmap, sizeof(chmap_str), chmap_str) > 0) {
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if ( (channels == 6) &&
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((SDL_strcmp("FL FR FC LFE RL RR", chmap_str) == 0) ||
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(SDL_strcmp("FL FR FC LFE SL SR", chmap_str) == 0)) ) {
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device->spec.use_channel_map = SDL_FALSE;
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} else if ((channels == 8) && (SDL_strcmp("FL FR FC LFE SL SR RL RR", chmap_str) == 0)) {
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device->spec.use_channel_map = SDL_FALSE;
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}
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}
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free(chmap); // This should NOT be SDL_free()
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}
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#endif // SND_CHMAP_API_VERSION
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// Set the audio rate
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unsigned int rate = device->spec.freq;
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status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
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