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Dequeue audio only once per frame, improve quality

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Eidolon 2025-09-23 23:37:53 -05:00
parent 7109cd8c13
commit b52e1cb61f
3 changed files with 63 additions and 68 deletions
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118
src/d_clisrv.c
View file

@ -206,8 +206,11 @@ static UINT32 g_player_voice_frames_this_tic[MAXPLAYERS];
static OpusEncoder *g_local_opus_encoder; static OpusEncoder *g_local_opus_encoder;
static ReNameNoiseDenoiseState *g_local_renamenoise_state; static ReNameNoiseDenoiseState *g_local_renamenoise_state;
static UINT64 g_local_opus_frame = 0; static UINT64 g_local_opus_frame = 0;
#define SRB2_VOICE_OPUS_FRAME_SIZE 960 #define SRB2_VOICE_OPUS_FRAME_SIZE (20 * 48)
static float g_local_voice_buffer[SRB2_VOICE_OPUS_FRAME_SIZE]; #define SRB2_VOICE_MAX_FRAMES 8
#define SRB2_VOICE_MAX_DEQUEUE_SAMPLES (SRB2_VOICE_MAX_FRAMES * SRB2_VOICE_OPUS_FRAME_SIZE)
#define SRB2_VOICE_MAX_DEQUEUE_BYTES (SRB2_VOICE_MAX_DEQUEUE_SAMPLES * sizeof(float))
static float g_local_voice_buffer[SRB2_VOICE_MAX_DEQUEUE_SAMPLES];
static INT32 g_local_voice_buffer_len = 0; static INT32 g_local_voice_buffer_len = 0;
static INT32 g_local_voice_threshold_time = 0; static INT32 g_local_voice_threshold_time = 0;
float g_local_voice_last_peak = 0; float g_local_voice_last_peak = 0;
@ -3742,8 +3745,6 @@ static void InitializeLocalVoiceEncoder(void)
CONS_Alert(CONS_WARNING, "Failed to create Opus voice encoder: opus error %d\n", error); CONS_Alert(CONS_WARNING, "Failed to create Opus voice encoder: opus error %d\n", error);
encoder = NULL; encoder = NULL;
} }
opus_encoder_ctl(encoder, OPUS_SET_VBR(0));
opus_encoder_ctl(encoder, OPUS_SET_BITRATE(28000));
g_local_opus_encoder = encoder; g_local_opus_encoder = encoder;
g_local_opus_frame = 0; g_local_opus_frame = 0;
} }
@ -5379,7 +5380,7 @@ static void PT_HandleVoiceClient(SINT8 node, boolean isserver)
{ {
return; return;
} }
float *decoded_out = Z_Malloc(sizeof(float) * SRB2_VOICE_OPUS_FRAME_SIZE, PU_STATIC, NULL); float *decoded_out = Z_Malloc(sizeof(float) * 1920, PU_STATIC, NULL);
INT32 decoded_samples = 0; INT32 decoded_samples = 0;
UINT64 missedframes = 0; UINT64 missedframes = 0;
@ -5390,7 +5391,7 @@ static void PT_HandleVoiceClient(SINT8 node, boolean isserver)
for (UINT64 i = 0; i < missedframes; i++) for (UINT64 i = 0; i < missedframes; i++)
{ {
decoded_samples = opus_decode_float(decoder, NULL, 0, decoded_out, SRB2_VOICE_OPUS_FRAME_SIZE, 0); decoded_samples = opus_decode_float(decoder, NULL, 0, decoded_out, 1920, 0);
if (decoded_samples < 0) if (decoded_samples < 0)
{ {
continue; continue;
@ -5402,7 +5403,7 @@ static void PT_HandleVoiceClient(SINT8 node, boolean isserver)
} }
g_player_opus_lastframe[playernum] = framenum; g_player_opus_lastframe[playernum] = framenum;
decoded_samples = opus_decode_float(decoder, frame, framesize, decoded_out, SRB2_VOICE_OPUS_FRAME_SIZE, 0); decoded_samples = opus_decode_float(decoder, frame, framesize, decoded_out, 1920, 0);
if (decoded_samples < 0) if (decoded_samples < 0)
{ {
Z_Free(decoded_out); Z_Free(decoded_out);
@ -7566,9 +7567,19 @@ void NetUpdate(void)
FileSendTicker(); FileSendTicker();
} }
static INT32 BiggestOpusFrameLength(INT32 samples)
{
if (samples >= 1920) return 1920;
if (samples >= 960) return 960;
if (samples >= 480) return 480;
return 0;
}
void NetVoiceUpdate(void) void NetVoiceUpdate(void)
{ {
UINT8 *encoded = NULL; UINT8 *encoded = NULL;
float *subframe_buffer = NULL;
float *denoise_buffer = NULL;
if (dedicated) if (dedicated)
{ {
@ -7576,42 +7587,25 @@ void NetVoiceUpdate(void)
} }
UINT32 bytes_dequed = 0; UINT32 bytes_dequed = 0;
do
bytes_dequed = S_SoundInputDequeueSamples((void*)(g_local_voice_buffer + g_local_voice_buffer_len), SRB2_VOICE_MAX_DEQUEUE_BYTES - (g_local_voice_buffer_len * sizeof(float)));
g_local_voice_buffer_len += bytes_dequed / 4;
INT32 buffer_offset = 0;
INT32 frame_length = 0;
for (
;
(frame_length = BiggestOpusFrameLength(g_local_voice_buffer_len - buffer_offset)) > 0 && (buffer_offset + frame_length) < g_local_voice_buffer_len;
buffer_offset += frame_length
)
{ {
// We need to drain the input queue completely, so do this in a full loop float *frame_buffer = g_local_voice_buffer + buffer_offset;
UINT32 to_read = (SRB2_VOICE_OPUS_FRAME_SIZE - g_local_voice_buffer_len) * sizeof(float);
if (to_read > 0)
{
// Attempt to fill the voice frame buffer
bytes_dequed = S_SoundInputDequeueSamples((void*)(g_local_voice_buffer + g_local_voice_buffer_len), to_read);
g_local_voice_buffer_len += bytes_dequed / 4;
}
else
{
bytes_dequed = 0;
}
if (g_local_voice_buffer_len < SRB2_VOICE_OPUS_FRAME_SIZE)
{
continue;
}
if (S_SoundInputRemainingSamples() > 5 * SRB2_VOICE_OPUS_FRAME_SIZE)
{
// If there are too many frames worth of samples to dequeue (100ms), skip this frame instead of encoding.
// This is so we drain the queue without sending too many packets that might queue up on the network driver.
g_local_voice_buffer_len = 0;
continue;
}
// Amp of +10 dB is appromiately "twice as loud" // Amp of +10 dB is appromiately "twice as loud"
float ampfactor = powf(10, (float) cv_voice_inputamp.value / 20.f); float ampfactor = powf(10, (float) cv_voice_inputamp.value / 20.f);
for (int i = 0; i < g_local_voice_buffer_len; i++) for (int i = 0; i < frame_length; i++)
{ {
g_local_voice_buffer[i] *= ampfactor; frame_buffer[i] *= ampfactor;
} }
if (cv_voice_denoise.value) if (cv_voice_denoise.value)
@ -7620,30 +7614,34 @@ void NetVoiceUpdate(void)
{ {
InitializeLocalVoiceDenoiser(); InitializeLocalVoiceDenoiser();
} }
int rnnoise_size = renamenoise_get_frame_size(); int rnnoise_size = renamenoise_get_frame_size(); // this is always 480
float *subframe_buffer = (float*) Z_Malloc(rnnoise_size * sizeof(float), PU_STATIC, NULL); if (subframe_buffer == NULL)
float *denoise_buffer = (float*) Z_Malloc(rnnoise_size * sizeof(float), PU_STATIC, NULL); {
subframe_buffer = (float*) Z_Malloc(rnnoise_size * sizeof(float), PU_STATIC, NULL);
}
if (denoise_buffer == NULL)
{
denoise_buffer = (float*) Z_Malloc(rnnoise_size * sizeof(float), PU_STATIC, NULL);
}
// rnnoise frames are smaller than opus, but we should not expect the opus frame to be an exact multiple of rnnoise // rnnoise frames are smaller than opus, but we should not expect the opus frame to be an exact multiple of rnnoise
for (int denoise_position = 0; denoise_position < SRB2_VOICE_OPUS_FRAME_SIZE; denoise_position += rnnoise_size) for (int denoise_position = 0; denoise_position < frame_length; denoise_position += rnnoise_size)
{ {
memset(subframe_buffer, 0, rnnoise_size * sizeof(float)); memset(subframe_buffer, 0, rnnoise_size * sizeof(float));
memcpy(subframe_buffer, g_local_voice_buffer + denoise_position, min(rnnoise_size * sizeof(float), (SRB2_VOICE_OPUS_FRAME_SIZE - denoise_position) * sizeof(float))); memcpy(subframe_buffer, frame_buffer + denoise_position, min(rnnoise_size * sizeof(float), (frame_length - denoise_position) * sizeof(float)));
renamenoise_process_frame(g_local_renamenoise_state, denoise_buffer, subframe_buffer); renamenoise_process_frame(g_local_renamenoise_state, denoise_buffer, subframe_buffer);
memcpy(g_local_voice_buffer + denoise_position, denoise_buffer, min(rnnoise_size * sizeof(float), (SRB2_VOICE_OPUS_FRAME_SIZE - denoise_position) * sizeof(float))); memcpy(frame_buffer + denoise_position, denoise_buffer, min(rnnoise_size * sizeof(float), (frame_length - denoise_position) * sizeof(float)));
} }
Z_Free(denoise_buffer);
Z_Free(subframe_buffer);
} }
float softmem = 0.f; float softmem = 0.f;
opus_pcm_soft_clip(g_local_voice_buffer, SRB2_VOICE_OPUS_FRAME_SIZE, 1, &softmem); opus_pcm_soft_clip(frame_buffer, frame_length, 1, &softmem);
// Voice detection gate open/close // Voice detection gate open/close
float maxamplitude = 0.f; float maxamplitude = 0.f;
for (int i = 0; i < g_local_voice_buffer_len; i++) for (int i = 0; i < frame_length; i++)
{ {
maxamplitude = max(fabsf(g_local_voice_buffer[i]), maxamplitude); maxamplitude = max(fabsf(frame_buffer[i]), maxamplitude);
} }
// 20. * log_10(amplitude) -> decibels (up to 0) // 20. * log_10(amplitude) -> decibels (up to 0)
// lower than -30 dB is usually inaudible // lower than -30 dB is usually inaudible
@ -7660,7 +7658,6 @@ void NetVoiceUpdate(void)
case 0: case 0:
if (I_GetTime() - g_local_voice_threshold_time > 15) if (I_GetTime() - g_local_voice_threshold_time > 15)
{ {
g_local_voice_buffer_len = 0;
g_local_voice_detected = false; g_local_voice_detected = false;
continue; continue;
} }
@ -7668,20 +7665,17 @@ void NetVoiceUpdate(void)
case 1: case 1:
if (!g_voicepushtotalk_on) if (!g_voicepushtotalk_on)
{ {
g_local_voice_buffer_len = 0;
g_local_voice_detected = false; g_local_voice_detected = false;
continue; continue;
} }
g_local_voice_detected = true; g_local_voice_detected = true;
break; break;
default: default:
g_local_voice_buffer_len = 0;
continue; continue;
} }
if (cv_voice_selfdeafen.value == 1) if (cv_voice_selfdeafen.value == 1)
{ {
g_local_voice_buffer_len = 0;
continue; continue;
} }
@ -7696,7 +7690,7 @@ void NetVoiceUpdate(void)
} }
OpusEncoder *encoder = g_local_opus_encoder; OpusEncoder *encoder = g_local_opus_encoder;
INT32 result = opus_encode_float(encoder, g_local_voice_buffer, SRB2_VOICE_OPUS_FRAME_SIZE, encoded, 1400); INT32 result = opus_encode_float(encoder, frame_buffer, frame_length, encoded, 1400);
if (result < 0) if (result < 0)
{ {
continue; continue;
@ -7718,14 +7712,20 @@ void NetVoiceUpdate(void)
{ {
RecreatePlayerOpusDecoder(consoleplayer); RecreatePlayerOpusDecoder(consoleplayer);
} }
result = opus_decode_float(g_player_opus_decoders[consoleplayer], encoded, result, g_local_voice_buffer, SRB2_VOICE_OPUS_FRAME_SIZE, 0); result = opus_decode_float(g_player_opus_decoders[consoleplayer], encoded, result, frame_buffer, frame_length, 0);
S_QueueVoiceFrameFromPlayer(consoleplayer, g_local_voice_buffer, result * sizeof(float), false); S_QueueVoiceFrameFromPlayer(consoleplayer, frame_buffer, result * sizeof(float), false);
} }
g_local_voice_buffer_len = 0;
g_local_opus_frame += 1; g_local_opus_frame += 1;
} while (bytes_dequed > 0); }
if (buffer_offset > 0)
{
memmove(g_local_voice_buffer, g_local_voice_buffer + buffer_offset, (g_local_voice_buffer_len - buffer_offset) * sizeof(float));
g_local_voice_buffer_len -= buffer_offset;
}
if (denoise_buffer) Z_Free(denoise_buffer);
if (subframe_buffer) Z_Free(subframe_buffer);
if (encoded) Z_Free(encoded); if (encoded) Z_Free(encoded);
return; return;
} }

4
src/s_sound.c
View file

@ -970,7 +970,7 @@ void S_UpdateVoicePositionalProperties(void)
for (i = 0; i < MAXPLAYERS; i++) for (i = 0; i < MAXPLAYERS; i++)
{ {
if (!playeringame[i] || !voice_proximity_enabled) if (!playeringame[i])
{ {
I_SetPlayerVoiceProperties(i, speakingplayerattenuation, 0.0f); I_SetPlayerVoiceProperties(i, speakingplayerattenuation, 0.0f);
continue; continue;
@ -1025,7 +1025,7 @@ void S_UpdateVoicePositionalProperties(void)
attenuation_distance = voice_distanceattenuation_teamdistance; attenuation_distance = voice_distanceattenuation_teamdistance;
} }
if (attenuation_distance > 0 && voice_distanceattenuation_factor >= 0 && voice_distanceattenuation_factor <= 1.0f) if (voice_proximity_enabled && attenuation_distance > 0 && voice_distanceattenuation_factor >= 0 && voice_distanceattenuation_factor <= 1.0f)
{ {
float invfactor = 1.0f - voice_distanceattenuation_factor; float invfactor = 1.0f - voice_distanceattenuation_factor;
float distfactor = max(0.f, min(attenuation_distance, pdistance)) / attenuation_distance; float distfactor = max(0.f, min(attenuation_distance, pdistance)) / attenuation_distance;

9
src/sdl/new_sound.cpp
View file

@ -155,7 +155,7 @@ public:
float left_scale = std::cos(sep_pan); float left_scale = std::cos(sep_pan);
float right_scale = std::sin(sep_pan); float right_scale = std::sin(sep_pan);
buffer[i] = {buffer[i].amplitudes[0] * volume_ * left_scale, buffer[i].amplitudes[1] * volume_ * right_scale}; buffer[i] = {std::clamp(buffer[i].amplitudes[0] * volume_ * left_scale, -1.f, 1.f), std::clamp(buffer[i].amplitudes[1] * volume_ * right_scale, -1.f, 1.f)};
} }
return buffer.size(); return buffer.size();
@ -1046,13 +1046,8 @@ UINT32 I_SoundInputDequeueSamples(void *data, UINT32 len)
{ {
return 0; return 0;
} }
UINT32 avail = SDL_GetQueuedAudioSize(g_input_device_id);
if (avail == 0)
{
return 0;
}
UINT32 ret = SDL_DequeueAudio(g_input_device_id, data, std::min(len, avail)); UINT32 ret = SDL_DequeueAudio(g_input_device_id, data, len);
return ret; return ret;
} }