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audio: fix wrong benchmark times

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The decoder benchmark was formatting the wrong variable and produced
meaningless results in the logs.
This commit is contained in:
Marcin Kurczewski 2025-04-26 16:00:16 +02:00
parent 92ff6e12c7
commit e439371c66
1 changed files with 64 additions and 41 deletions
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103
src/libtrx/engine/audio_sample.c
View file

@ -1,5 +1,6 @@
#include "audio_internal.h"
#include "benchmark.h"
#include "debug.h"
#include "log.h"
#include "memory.h"
@ -21,7 +22,6 @@
#include <stdint.h>
#include <stdio.h>
#include <string.h>
#include <time.h>
typedef struct {
char *original_data;
@ -50,8 +50,8 @@ typedef struct {
} AUDIO_SAMPLE_SOUND;
typedef struct {
const char *data;
const char *ptr;
const uint8_t *data;
const uint8_t *ptr;
int32_t size;
int32_t remaining;
} AUDIO_AV_BUFFER;
@ -64,7 +64,7 @@ static double M_DecibelToMultiplier(double db_gain);
static bool M_RecalculateChannelVolumes(int32_t sound_id);
static int32_t M_ReadAVBuffer(void *opaque, uint8_t *dst, int32_t dst_size);
static int64_t M_SeekAVBuffer(void *opaque, int64_t offset, int32_t whence);
static bool M_Convert(const int32_t sample_id);
static bool M_ConvertSample(const int32_t sample_id);
static double M_DecibelToMultiplier(double db_gain)
{
@ -135,18 +135,13 @@ static int64_t M_SeekAVBuffer(void *opaque, int64_t offset, int32_t whence)
return src->ptr - src->data;
}
static bool M_Convert(const int32_t sample_id)
static bool M_ConvertRawData(
const uint8_t *const original_data, const int32_t original_size,
const int32_t dst_sample_rate, const int32_t dst_format,
const int32_t dst_channel_count, uint8_t **const out_sample_data,
size_t *const out_size, size_t *const out_sample_count)
{
ASSERT(sample_id >= 0 && sample_id < m_LoadedSamplesCount);
bool result = false;
AUDIO_SAMPLE *const sample = &m_LoadedSamples[sample_id];
if (sample->sample_data != nullptr) {
return true;
}
const clock_t time_start = clock();
size_t working_buffer_size = 0;
float *working_buffer = nullptr;
@ -188,10 +183,10 @@ static bool M_Convert(const int32_t sample_id)
}
AUDIO_AV_BUFFER av_buf = {
.data = sample->original_data,
.ptr = sample->original_data,
.size = sample->original_size,
.remaining = sample->original_size,
.data = original_data,
.ptr = original_data,
.size = original_size,
.remaining = original_size,
};
av.avio_context = avio_alloc_context(
@ -248,7 +243,7 @@ static bool M_Convert(const int32_t sample_id)
}
av.packet = av_packet_alloc();
if (!av.packet) {
if (av.packet == nullptr) {
error_code = AVERROR(ENOMEM);
goto cleanup;
}
@ -283,7 +278,7 @@ static bool M_Convert(const int32_t sample_id)
swr.src.ch_layout = av.codec_ctx->ch_layout;
swr.src.format = av.codec_ctx->sample_fmt;
swr.dst.sample_rate = AUDIO_WORKING_RATE;
av_channel_layout_default(&swr.dst.ch_layout, 1);
av_channel_layout_default(&swr.dst.ch_layout, dst_channel_count);
swr.dst.format = Audio_GetAVAudioFormat(AUDIO_WORKING_FORMAT);
swr_alloc_set_opts2(
&swr.ctx, &swr.dst.ch_layout, swr.dst.format,
@ -351,25 +346,24 @@ static bool M_Convert(const int32_t sample_id)
av_packet_unref(av.packet);
}
int32_t sample_format_bytes = av_get_bytes_per_sample(swr.dst.format);
sample->num_samples = working_buffer_size / sample_format_bytes
if (out_size != nullptr) {
*out_size = working_buffer_size;
}
if (out_sample_count != nullptr) {
*out_sample_count = (int32_t)(working_buffer_size
/ av_get_bytes_per_sample(swr.dst.format))
/ swr.dst.ch_layout.nb_channels;
sample->channels = swr.src.ch_layout.nb_channels;
sample->sample_data = working_buffer;
}
if (out_sample_data != nullptr) {
*out_sample_data = (uint8_t *)working_buffer;
} else {
Memory_FreePointer(&working_buffer);
}
result = true;
const clock_t time_end = clock();
const double time_delta =
(((double)(time_end - time_start)) / CLOCKS_PER_SEC) * 1000.0f;
LOG_DEBUG(
"Sample %d decoded (%.0f ms)", sample_id, sample->original_size,
time_delta);
cleanup:
if (error_code != 0) {
LOG_ERROR(
"Error while opening sample ID %d: %s", sample_id,
av_err2str(error_code));
LOG_ERROR("Error while decoding sample: %s", av_err2str(error_code));
}
if (swr.ctx) {
@ -387,11 +381,15 @@ cleanup:
av.codec = nullptr;
if (!result) {
sample->sample_data = nullptr;
sample->original_data = nullptr;
sample->original_size = 0;
sample->num_samples = 0;
sample->channels = 0;
if (out_size != nullptr) {
*out_size = 0;
}
if (out_sample_count != nullptr) {
*out_sample_count = 0;
}
if (out_sample_data != nullptr) {
*out_sample_data = nullptr;
}
Memory_FreePointer(&working_buffer);
}
@ -411,6 +409,31 @@ cleanup:
return result;
}
static bool M_ConvertSample(const int32_t sample_id)
{
ASSERT(sample_id >= 0 && sample_id < m_LoadedSamplesCount);
AUDIO_SAMPLE *const sample = &m_LoadedSamples[sample_id];
if (sample->sample_data != nullptr) {
return true;
}
size_t num_samples;
BENCHMARK benchmark = Benchmark_Start();
const bool result = M_ConvertRawData(
(uint8_t *)sample->original_data, sample->original_size,
AUDIO_WORKING_RATE, Audio_GetAVAudioFormat(AUDIO_WORKING_FORMAT), 1,
(uint8_t **)&sample->sample_data, nullptr, &num_samples);
char buffer[80];
sprintf(buffer, "sample %d decoded", sample_id);
Benchmark_End(&benchmark, buffer);
sample->channels = 1;
sample->num_samples = num_samples;
return result;
}
void Audio_Sample_Init(void)
{
for (int32_t sound_id = 0; sound_id < AUDIO_MAX_ACTIVE_SAMPLES;
@ -553,7 +576,7 @@ int32_t Audio_Sample_Play(
continue;
}
M_Convert(sample_id);
M_ConvertSample(sample_id);
sound->is_used = true;
sound->is_playing = true;