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Merge pull request #55846 from ellenhp/fix_ogg_edge_cases

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Fix ogg edge cases
This commit is contained in:
Rémi Verschelde 2022-06-22 23:33:51 +02:00 committed by GitHub
commit d1dac8427a
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GPG Key ID: 4AEE18F83AFDEB23
3 changed files with 23 additions and 27 deletions
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4
modules/minimp3/audio_stream_mp3.cpp
View File

@ -38,7 +38,9 @@
#include "core/io/file_access.h"
int AudioStreamPlaybackMP3::_mix_internal(AudioFrame *p_buffer, int p_frames) {
ERR_FAIL_COND_V(!active, 0);
if (!active) {
return 0;
}
int todo = p_frames;

17
modules/vorbis/audio_stream_ogg_vorbis.cpp
View File

@ -36,23 +36,22 @@
int AudioStreamPlaybackOGGVorbis::_mix_internal(AudioFrame *p_buffer, int p_frames) {
ERR_FAIL_COND_V(!ready, 0);
ERR_FAIL_COND_V(!active, 0);
if (!active) {
return 0;
}
int todo = p_frames;
int start_buffer = 0;
int frames_mixed_this_step = p_frames;
while (todo && active) {
while (todo > 0 && active) {
AudioFrame *buffer = p_buffer;
if (start_buffer > 0) {
buffer = buffer + start_buffer;
}
int mixed = _mix_frames_vorbis(buffer, todo);
if (mixed < 0) {
return 0;
}
ERR_FAIL_COND_V(mixed < 0, 0);
todo -= mixed;
frames_mixed += mixed;
start_buffer += mixed;
@ -67,16 +66,14 @@ int AudioStreamPlaybackOGGVorbis::_mix_internal(AudioFrame *p_buffer, int p_fram
// we still have buffer to fill, start from this element in the next iteration.
start_buffer = p_frames - todo;
} else {
frames_mixed_this_step = p_frames - todo;
for (int i = p_frames - todo; i < p_frames; i++) {
p_buffer[i] = AudioFrame(0, 0);
}
active = false;
todo = 0;
}
}
}
return frames_mixed_this_step;
return p_frames - todo;
}
int AudioStreamPlaybackOGGVorbis::_mix_frames_vorbis(AudioFrame *p_buffer, int p_frames) {

17
servers/audio/audio_stream.cpp
View File

@ -135,9 +135,10 @@ int AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale,
uint64_t mix_increment = uint64_t(((get_stream_sampling_rate() * p_rate_scale * playback_speed_scale) / double(target_rate)) * double(FP_LEN));
int mixed_frames_total = p_frames;
int mixed_frames_total = -1;
for (int i = 0; i < p_frames; i++) {
int i;
for (i = 0; i < p_frames; i++) {
uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS);
//standard cubic interpolation (great quality/performance ratio)
//this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory.
@ -147,7 +148,7 @@ int AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale,
AudioFrame y2 = internal_buffer[idx - 1];
AudioFrame y3 = internal_buffer[idx - 0];
if (idx <= internal_buffer_end && idx >= internal_buffer_end && mixed_frames_total == p_frames) {
if (idx >= internal_buffer_end && mixed_frames_total == -1) {
// The internal buffer ends somewhere in this range, and we haven't yet recorded the number of good frames we have.
mixed_frames_total = i;
}
@ -167,7 +168,6 @@ int AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale,
internal_buffer[1] = internal_buffer[INTERNAL_BUFFER_LEN + 1];
internal_buffer[2] = internal_buffer[INTERNAL_BUFFER_LEN + 2];
internal_buffer[3] = internal_buffer[INTERNAL_BUFFER_LEN + 3];
if (is_playing()) {
int mixed_frames = _mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
if (mixed_frames != INTERNAL_BUFFER_LEN) {
// internal_buffer[mixed_frames] is the first frame of silence.
@ -176,15 +176,12 @@ int AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale,
// The internal buffer does not contain the first frame of silence.
internal_buffer_end = -1;
}
} else {
//fill with silence, not playing
for (int j = 0; j < INTERNAL_BUFFER_LEN; ++j) {
internal_buffer[j + 4] = AudioFrame(0, 0);
}
}
mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS);
}
}
if (mixed_frames_total == -1 && i == p_frames) {
mixed_frames_total = p_frames;
}
return mixed_frames_total;
}