impressionyang
dc4ebab47c
- audio_capture 启动时输出详细设备信息（名称、Host API、采样率） - 录音停止时输出 RMS 电平和峰值，帮助诊断音频质量问题 - 设置页面新增音频输入设备下拉选择，支持从 PortAudio 设备列表中手动选择 - 语音输入服务使用配置的音频设备和采样率参数 - 检测 monitor/output 类型设备时发出警告，避免选错回环设备 Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
277 lines
8.9 KiB
C++
277 lines
8.9 KiB
C++
#include "audio_capture.h"
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#include "utils/logger.h"
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#ifdef HAVE_PORTAUDIO
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#include <portaudio.h>
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#include <cmath>
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#endif
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static const char* const kTag = "AudioCapture";
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namespace impress {
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// 预分配缓冲区，避免在实时回调中分配内存
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static constexpr int kMaxBufferSize = 8192;
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// 全局 PortAudio 初始化状态
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static bool gPaInitialized = false;
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/** 安全初始化 PortAudio（多次调用不报错） */
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static bool ensurePaInitialized() {
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if (gPaInitialized) return true;
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if (Pa_Initialize() == paNoError) {
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gPaInitialized = true;
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return true;
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}
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return false;
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}
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/** 安全终止 PortAudio */
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static void safePaTerminate() {
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if (gPaInitialized) {
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Pa_Terminate();
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gPaInitialized = false;
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}
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}
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// 回调上下文：独立于 Impl 的 POD 结构，供静态回调使用
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struct CallbackContext {
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AudioCapture* owner = nullptr;
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#ifdef HAVE_PORTAUDIO
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PaStream* stream = nullptr;
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float buffer[kMaxBufferSize];
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#endif
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int sampleRate = 16000;
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// 音频电平诊断
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float peakLevel = 0.0f;
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double sumSquares = 0.0;
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int sampleCount = 0;
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};
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struct AudioCapture::Impl {
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CallbackContext ctx;
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};
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static int paCallback(const void* input, void* /*output*/,
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unsigned long frameCount,
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const PaStreamCallbackTimeInfo* /*timeInfo*/,
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PaStreamCallbackFlags /*statusFlags*/,
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void* userData)
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{
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#ifdef HAVE_PORTAUDIO
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auto* ctx = static_cast<CallbackContext*>(userData);
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const float* samples = static_cast<const float*>(input);
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// 使用预分配缓冲区，避免实时线程中分配内存
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unsigned long count = frameCount;
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if (count > kMaxBufferSize) count = kMaxBufferSize;
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// 拷贝到预分配缓冲区 + 计算音频电平
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float peak = 0.0f;
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double ss = 0.0;
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for (unsigned long i = 0; i < count; i++) {
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float s = samples[i];
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ctx->buffer[i] = s;
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float absS = std::fabs(s);
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if (absS > peak) peak = absS;
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ss += s * s;
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}
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// 更新诊断数据
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if (peak > ctx->peakLevel) ctx->peakLevel = peak;
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ctx->sumSquares += ss;
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ctx->sampleCount += count;
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// 发射信号（Qt 使用 QueuedConnection，线程安全）
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std::vector<float> data(ctx->buffer, ctx->buffer + count);
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emit ctx->owner->audioDataReady(data, ctx->sampleRate);
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return paContinue;
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#else
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(void)input; (void)frameCount; (void)userData;
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return 0;
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#endif
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}
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AudioCapture::AudioCapture(QObject* parent)
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: QObject(parent)
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, impl_(std::make_unique<Impl>())
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{
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impl_->ctx.owner = this;
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}
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AudioCapture::~AudioCapture() {
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stop();
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}
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QStringList AudioCapture::getDeviceList() {
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QStringList devices;
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devices << "默认设备";
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#ifdef HAVE_PORTAUDIO
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if (!ensurePaInitialized()) {
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LOG_ERROR(kTag, "PortAudio 初始化失败");
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return devices;
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}
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int count = Pa_GetDeviceCount();
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for (int i = 0; i < count; ++i) {
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const PaDeviceInfo* info = Pa_GetDeviceInfo(i);
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if (info && info->maxInputChannels > 0) {
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const PaHostApiInfo* hostApi = Pa_GetHostApiInfo(info->hostApi);
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QString hostApiName = hostApi ? hostApi->name : "未知";
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devices << QString("[%1] %2 (CH:%3, SR:%4, %5)")
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.arg(i).arg(info->name).arg(info->maxInputChannels)
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.arg(info->defaultSampleRate).arg(hostApiName);
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}
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}
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#endif
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return devices;
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}
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int AudioCapture::getDefaultDeviceIndex() {
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#ifdef HAVE_PORTAUDIO
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if (!ensurePaInitialized()) return -1;
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return Pa_GetDefaultInputDevice();
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#else
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return -1;
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#endif
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}
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bool AudioCapture::start(int deviceIndex, int sampleRate, int bufferSizeMs) {
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if (running_) {
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LOG_WARNING(kTag, "已在运行中");
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return false;
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}
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#ifdef HAVE_PORTAUDIO
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if (!ensurePaInitialized()) {
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LOG_ERROR(kTag, "PortAudio 初始化失败");
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return false;
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}
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// 枚举所有 Host API 用于诊断
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LOG_DEBUG(kTag, QString("Host API 数量: %1").arg(Pa_GetHostApiCount()));
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for (int i = 0; i < Pa_GetHostApiCount(); i++) {
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const PaHostApiInfo* api = Pa_GetHostApiInfo(i);
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if (api) {
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LOG_DEBUG(kTag, QString(" Host API #%1: %2 (设备数: %3)")
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.arg(i).arg(api->name).arg(api->deviceCount));
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}
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}
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// 选择设备
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int devIdx = deviceIndex < 0 ? Pa_GetDefaultInputDevice() : deviceIndex;
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if (devIdx < 0 || devIdx >= Pa_GetDeviceCount()) {
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LOG_ERROR(kTag, QString("无效的音频设备索引: %1 (默认设备: %2)")
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.arg(deviceIndex).arg(Pa_GetDefaultInputDevice()));
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return false;
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}
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const PaDeviceInfo* devInfo = Pa_GetDeviceInfo(devIdx);
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if (!devInfo || devInfo->maxInputChannels <= 0) {
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LOG_ERROR(kTag, "所选设备不是输入设备");
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return false;
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}
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const PaHostApiInfo* hostApi = Pa_GetHostApiInfo(devInfo->hostApi);
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LOG_INFO(kTag, QString("=== 音频设备诊断 ==="));
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LOG_INFO(kTag, QString(" 设备 #%1: %2").arg(devIdx).arg(devInfo->name));
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LOG_INFO(kTag, QString(" Host API: %1").arg(hostApi ? hostApi->name : "未知"));
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LOG_INFO(kTag, QString(" 最大输入通道: %1").arg(devInfo->maxInputChannels));
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LOG_INFO(kTag, QString(" 设备默认采样率: %1 Hz").arg(devInfo->defaultSampleRate, 0, 'f', 0));
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LOG_INFO(kTag, QString(" 请求采样率: %1 Hz").arg(sampleRate));
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LOG_INFO(kTag, QString(" 采样格式: paFloat32 | paNonInterleaved"));
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LOG_INFO(kTag, QString(" 请求通道数: 1 (mono)"));
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LOG_INFO(kTag, QString(" 缓冲区: %1ms (%2 帧)").arg(bufferSizeMs)
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.arg(sampleRate * bufferSizeMs / 1000));
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// 检查是否可能选错设备（名称包含 monitor 的通常是回环设备）
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QString devName = QString(devInfo->name).toLower();
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if (devName.contains("monitor") || devName.contains("output")) {
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LOG_WARNING(kTag, "⚠️ 当前设备名称包含 'monitor' 或 'output'，"
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"这可能是扬声器回环设备而非麦克风！如果录制的是噪音，请在设置中选择正确的麦克风设备。");
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}
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PaStreamParameters inputParams{};
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inputParams.device = devIdx;
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inputParams.channelCount = 1;
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inputParams.sampleFormat = paFloat32 | paNonInterleaved;
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// 使用高延迟以避免回调过快
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inputParams.suggestedLatency = devInfo->defaultHighInputLatency;
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int framesPerBuffer = sampleRate * bufferSizeMs / 1000;
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if (framesPerBuffer < 256) framesPerBuffer = 256;
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PaError err = Pa_OpenStream(
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&impl_->ctx.stream, &inputParams, nullptr, sampleRate,
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static_cast<unsigned long>(framesPerBuffer),
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paClipOff, paCallback, &impl_->ctx);
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if (err != paNoError || !impl_->ctx.stream) {
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LOG_ERROR(kTag, QString("打开音频流失败: %1").arg(Pa_GetErrorText(err)));
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return false;
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}
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err = Pa_StartStream(impl_->ctx.stream);
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if (err != paNoError) {
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LOG_ERROR(kTag, QString("启动音频流失败: %1").arg(Pa_GetErrorText(err)));
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Pa_CloseStream(impl_->ctx.stream);
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impl_->ctx.stream = nullptr;
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return false;
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}
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// 重置诊断计数器
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impl_->ctx.peakLevel = 0.0f;
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impl_->ctx.sumSquares = 0.0;
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impl_->ctx.sampleCount = 0;
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impl_->ctx.sampleRate = sampleRate;
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running_ = true;
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emit runningChanged(true);
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LOG_INFO(kTag, QString("音频采集已启动 (设备: %1, 采样率: %2, 缓冲区: %3ms)")
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.arg(deviceIndex).arg(sampleRate).arg(bufferSizeMs));
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return true;
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#else
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LOG_ERROR(kTag, "PortAudio 未编译启用");
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emit error("PortAudio 未编译启用");
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return false;
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#endif
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}
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void AudioCapture::stop() {
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if (!running_) return;
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#ifdef HAVE_PORTAUDIO
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// 输出音频电平诊断信息
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if (impl_->ctx.sampleCount > 0) {
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double rms = std::sqrt(impl_->ctx.sumSquares / impl_->ctx.sampleCount);
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LOG_INFO(kTag, QString("=== 音频电平诊断 ==="));
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LOG_INFO(kTag, QString(" 总样本数: %1").arg(impl_->ctx.sampleCount));
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LOG_INFO(kTag, QString(" RMS 电平: %1").arg(rms, 0, 'f', 6));
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LOG_INFO(kTag, QString(" 峰值: %1").arg(impl_->ctx.peakLevel, 0, 'f', 4));
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if (rms < 0.001) {
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LOG_WARNING(kTag, "⚠️ 音频信号过弱，可能是静音或设备未正确采集");
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} else if (rms > 0.5) {
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LOG_WARNING(kTag, "⚠️ 音频信号过强，可能存在削波");
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} else if (impl_->ctx.peakLevel > 0.9f) {
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LOG_INFO(kTag, " 信号幅度正常");
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} else {
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LOG_WARNING(kTag, "⚠️ 信号幅度偏低，请检查设备选择");
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}
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}
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if (impl_->ctx.stream) {
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Pa_StopStream(impl_->ctx.stream);
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Pa_CloseStream(impl_->ctx.stream);
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impl_->ctx.stream = nullptr;
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}
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safePaTerminate();
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#endif
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running_ = false;
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emit runningChanged(false);
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LOG_INFO(kTag, "音频采集已停止");
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}
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} // namespace impress
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