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Diffstat (limited to 'servers/audio/audio_rb_resampler.cpp')
-rw-r--r--servers/audio/audio_rb_resampler.cpp210
1 files changed, 79 insertions, 131 deletions
diff --git a/servers/audio/audio_rb_resampler.cpp b/servers/audio/audio_rb_resampler.cpp
index 113e356612..b0b94a1f49 100644
--- a/servers/audio/audio_rb_resampler.cpp
+++ b/servers/audio/audio_rb_resampler.cpp
@@ -28,6 +28,9 @@
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#include "audio_rb_resampler.h"
+#include "core/math/math_funcs.h"
+#include "os/os.h"
+#include "servers/audio_server.h"
int AudioRBResampler::get_channel_count() const {
@@ -37,8 +40,11 @@ int AudioRBResampler::get_channel_count() const {
return channels;
}
+// Linear interpolation based sample rate convertion (low quality)
+// Note that AudioStreamPlaybackResampled::mix has better algorithm,
+// but it wasn't obvious to integrate that with VideoPlayer
template <int C>
-uint32_t AudioRBResampler::_resample(int32_t *p_dest, int p_todo, int32_t p_increment) {
+uint32_t AudioRBResampler::_resample(AudioFrame *p_dest, int p_todo, int32_t p_increment) {
uint32_t read = offset & MIX_FRAC_MASK;
@@ -47,186 +53,128 @@ uint32_t AudioRBResampler::_resample(int32_t *p_dest, int p_todo, int32_t p_incr
offset = (offset + p_increment) & (((1 << (rb_bits + MIX_FRAC_BITS)) - 1));
read += p_increment;
uint32_t pos = offset >> MIX_FRAC_BITS;
- uint32_t frac = offset & MIX_FRAC_MASK;
-#ifndef FAST_AUDIO
+ float frac = float(offset & MIX_FRAC_MASK) / float(MIX_FRAC_LEN);
ERR_FAIL_COND_V(pos >= rb_len, 0);
-#endif
uint32_t pos_next = (pos + 1) & rb_mask;
- //printf("rb pos %i\n",pos);
// since this is a template with a known compile time value (C), conditionals go away when compiling.
if (C == 1) {
- int32_t v0 = rb[pos];
- int32_t v0n = rb[pos_next];
-#ifndef FAST_AUDIO
- v0 += (v0n - v0) * (int32_t)frac >> MIX_FRAC_BITS;
-#endif
- v0 <<= 16;
- p_dest[i] = v0;
+ float v0 = rb[pos];
+ float v0n = rb[pos_next];
+ v0 += (v0n - v0) * frac;
+ p_dest[i] = AudioFrame(v0, v0);
}
+
if (C == 2) {
- int32_t v0 = rb[(pos << 1) + 0];
- int32_t v1 = rb[(pos << 1) + 1];
- int32_t v0n = rb[(pos_next << 1) + 0];
- int32_t v1n = rb[(pos_next << 1) + 1];
-
-#ifndef FAST_AUDIO
- v0 += (v0n - v0) * (int32_t)frac >> MIX_FRAC_BITS;
- v1 += (v1n - v1) * (int32_t)frac >> MIX_FRAC_BITS;
-#endif
- v0 <<= 16;
- v1 <<= 16;
- p_dest[(i << 1) + 0] = v0;
- p_dest[(i << 1) + 1] = v1;
+ float v0 = rb[(pos << 1) + 0];
+ float v1 = rb[(pos << 1) + 1];
+ float v0n = rb[(pos_next << 1) + 0];
+ float v1n = rb[(pos_next << 1) + 1];
+
+ v0 += (v0n - v0) * frac;
+ v1 += (v1n - v1) * frac;
+ p_dest[i] = AudioFrame(v0, v1);
}
+ // For now, channels higher than stereo are almost ignored
if (C == 4) {
- int32_t v0 = rb[(pos << 2) + 0];
- int32_t v1 = rb[(pos << 2) + 1];
- int32_t v2 = rb[(pos << 2) + 2];
- int32_t v3 = rb[(pos << 2) + 3];
- int32_t v0n = rb[(pos_next << 2) + 0];
- int32_t v1n = rb[(pos_next << 2) + 1];
- int32_t v2n = rb[(pos_next << 2) + 2];
- int32_t v3n = rb[(pos_next << 2) + 3];
-
-#ifndef FAST_AUDIO
- v0 += (v0n - v0) * (int32_t)frac >> MIX_FRAC_BITS;
- v1 += (v1n - v1) * (int32_t)frac >> MIX_FRAC_BITS;
- v2 += (v2n - v2) * (int32_t)frac >> MIX_FRAC_BITS;
- v3 += (v3n - v3) * (int32_t)frac >> MIX_FRAC_BITS;
-#endif
- v0 <<= 16;
- v1 <<= 16;
- v2 <<= 16;
- v3 <<= 16;
- p_dest[(i << 2) + 0] = v0;
- p_dest[(i << 2) + 1] = v1;
- p_dest[(i << 2) + 2] = v2;
- p_dest[(i << 2) + 3] = v3;
+ float v0 = rb[(pos << 2) + 0];
+ float v1 = rb[(pos << 2) + 1];
+ float v2 = rb[(pos << 2) + 2];
+ float v3 = rb[(pos << 2) + 3];
+ float v0n = rb[(pos_next << 2) + 0];
+ float v1n = rb[(pos_next << 2) + 1];
+ float v2n = rb[(pos_next << 2) + 2];
+ float v3n = rb[(pos_next << 2) + 3];
+
+ v0 += (v0n - v0) * frac;
+ v1 += (v1n - v1) * frac;
+ v2 += (v2n - v2) * frac;
+ v3 += (v3n - v3) * frac;
+ p_dest[i] = AudioFrame(v0, v1);
}
if (C == 6) {
- int32_t v0 = rb[(pos * 6) + 0];
- int32_t v1 = rb[(pos * 6) + 1];
- int32_t v2 = rb[(pos * 6) + 2];
- int32_t v3 = rb[(pos * 6) + 3];
- int32_t v4 = rb[(pos * 6) + 4];
- int32_t v5 = rb[(pos * 6) + 5];
- int32_t v0n = rb[(pos_next * 6) + 0];
- int32_t v1n = rb[(pos_next * 6) + 1];
- int32_t v2n = rb[(pos_next * 6) + 2];
- int32_t v3n = rb[(pos_next * 6) + 3];
- int32_t v4n = rb[(pos_next * 6) + 4];
- int32_t v5n = rb[(pos_next * 6) + 5];
-
-#ifndef FAST_AUDIO
- v0 += (v0n - v0) * (int32_t)frac >> MIX_FRAC_BITS;
- v1 += (v1n - v1) * (int32_t)frac >> MIX_FRAC_BITS;
- v2 += (v2n - v2) * (int32_t)frac >> MIX_FRAC_BITS;
- v3 += (v3n - v3) * (int32_t)frac >> MIX_FRAC_BITS;
- v4 += (v4n - v4) * (int32_t)frac >> MIX_FRAC_BITS;
- v5 += (v5n - v5) * (int32_t)frac >> MIX_FRAC_BITS;
-#endif
- v0 <<= 16;
- v1 <<= 16;
- v2 <<= 16;
- v3 <<= 16;
- v4 <<= 16;
- v5 <<= 16;
- p_dest[(i * 6) + 0] = v0;
- p_dest[(i * 6) + 1] = v1;
- p_dest[(i * 6) + 2] = v2;
- p_dest[(i * 6) + 3] = v3;
- p_dest[(i * 6) + 4] = v4;
- p_dest[(i * 6) + 5] = v5;
+ float v0 = rb[(pos * 6) + 0];
+ float v1 = rb[(pos * 6) + 1];
+ float v2 = rb[(pos * 6) + 2];
+ float v3 = rb[(pos * 6) + 3];
+ float v4 = rb[(pos * 6) + 4];
+ float v5 = rb[(pos * 6) + 5];
+ float v0n = rb[(pos_next * 6) + 0];
+ float v1n = rb[(pos_next * 6) + 1];
+ float v2n = rb[(pos_next * 6) + 2];
+ float v3n = rb[(pos_next * 6) + 3];
+ float v4n = rb[(pos_next * 6) + 4];
+ float v5n = rb[(pos_next * 6) + 5];
+
+ p_dest[i] = AudioFrame(v0, v1);
}
}
- return read >> MIX_FRAC_BITS; //rb_read_pos=offset>>MIX_FRAC_BITS;
+ return read >> MIX_FRAC_BITS; //rb_read_pos = offset >> MIX_FRAC_BITS;
}
-bool AudioRBResampler::mix(int32_t *p_dest, int p_frames) {
+bool AudioRBResampler::mix(AudioFrame *p_dest, int p_frames) {
if (!rb)
return false;
- int write_pos_cache = rb_write_pos;
-
int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
-
- int rb_todo;
-
- if (write_pos_cache == rb_read_pos) {
- return false; //out of buffer
-
- } else if (rb_read_pos < write_pos_cache) {
-
- rb_todo = write_pos_cache - rb_read_pos; //-1?
- } else {
-
- rb_todo = (rb_len - rb_read_pos) + write_pos_cache; //-1?
- }
-
- int todo = MIN(((int64_t(rb_todo) << MIX_FRAC_BITS) / increment) + 1, p_frames);
+ int read_space = get_reader_space();
+ int target_todo = MIN(get_num_of_ready_frames(), p_frames);
{
-
- int read = 0;
+ int src_read = 0;
switch (channels) {
- case 1: read = _resample<1>(p_dest, todo, increment); break;
- case 2: read = _resample<2>(p_dest, todo, increment); break;
- case 4: read = _resample<4>(p_dest, todo, increment); break;
- case 6: read = _resample<6>(p_dest, todo, increment); break;
+ case 1: src_read = _resample<1>(p_dest, target_todo, increment); break;
+ case 2: src_read = _resample<2>(p_dest, target_todo, increment); break;
+ case 4: src_read = _resample<4>(p_dest, target_todo, increment); break;
+ case 6: src_read = _resample<6>(p_dest, target_todo, increment); break;
}
- //end of stream, fadeout
- int remaining = p_frames - todo;
- if (remaining && todo > 0) {
-
- //print_line("fadeout");
- for (uint32_t c = 0; c < channels; c++) {
+ if (src_read > read_space)
+ src_read = read_space;
- for (int i = 0; i < todo; i++) {
+ rb_read_pos = (rb_read_pos + src_read) & rb_mask;
- int32_t samp = p_dest[i * channels + c] >> 8;
- uint32_t mul = (todo - i) * 256 / todo;
- //print_line("mul: "+itos(i)+" "+itos(mul));
- p_dest[i * channels + c] = samp * mul;
- }
+ // Create fadeout effect for the end of stream (note that it can be because of slow writer)
+ if (p_frames - target_todo > 0) {
+ for (int i = 0; i < target_todo; i++) {
+ p_dest[i] = p_dest[i] * float(target_todo - i) / float(target_todo);
}
}
- //zero out what remains there to avoid glitches
- for (uint32_t i = todo * channels; i < int(p_frames) * channels; i++) {
-
- p_dest[i] = 0;
+ // Fill zeros (silence) for the rest of frames
+ for (uint32_t i = target_todo; i < p_frames; i++) {
+ p_dest[i] = AudioFrame(0, 0);
}
-
- if (read > rb_todo)
- read = rb_todo;
-
- rb_read_pos = (rb_read_pos + read) & rb_mask;
}
return true;
}
+int AudioRBResampler::get_num_of_ready_frames() {
+ if (!is_ready())
+ return 0;
+ int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
+ int read_space = get_reader_space();
+ return (int64_t(read_space) << MIX_FRAC_BITS) / increment;
+}
+
Error AudioRBResampler::setup(int p_channels, int p_src_mix_rate, int p_target_mix_rate, int p_buffer_msec, int p_minbuff_needed) {
ERR_FAIL_COND_V(p_channels != 1 && p_channels != 2 && p_channels != 4 && p_channels != 6, ERR_INVALID_PARAMETER);
- //float buffering_sec = int(GLOBAL_DEF("audio/stream_buffering_ms",500))/1000.0;
int desired_rb_bits = nearest_shift(MAX((p_buffer_msec / 1000.0) * p_src_mix_rate, p_minbuff_needed));
bool recreate = !rb;
if (rb && (uint32_t(desired_rb_bits) != rb_bits || channels != uint32_t(p_channels))) {
- //recreate
memdelete_arr(rb);
memdelete_arr(read_buf);
@@ -239,8 +187,8 @@ Error AudioRBResampler::setup(int p_channels, int p_src_mix_rate, int p_target_m
rb_bits = desired_rb_bits;
rb_len = (1 << rb_bits);
rb_mask = rb_len - 1;
- rb = memnew_arr(int16_t, rb_len * p_channels);
- read_buf = memnew_arr(int16_t, rb_len * p_channels);
+ rb = memnew_arr(float, rb_len *p_channels);
+ read_buf = memnew_arr(float, rb_len *p_channels);
}
src_mix_rate = p_src_mix_rate;
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-26 12:08:35 +0000