Feat/ffmpeg audio migration#21
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…ng and retry logic - Added configurable timeouts and retry mechanisms to improve resilience against network issues. - Refactored stream handling to ensure proper reconnection and error classification. - Introduced jitter-based backoff strategy for retries to reduce server load. - Updated tests to cover new behaviors and ensure reliability under various scenarios. - Removed unnecessary unsafe implementations for thread safety in FftAnalyzer.
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Pull request overview
该 PR 主要完成 native/audio-engine 从 ffmpeg-next 迁移到自研 ffmpeg_audio（静态 FFmpeg、零环境依赖）的改造，并补齐网络 URL 播放所需的 Read + Seek HTTP Range 数据源，以便在 Electron 主进程侧跨平台稳定解码/播放。
Changes:
- 将扫描/解码/元数据提取从
ffmpeg-next全量迁移到ffmpeg_audioAPI。 - 新增
HttpRangeSource：用 ureq(TLS) 实现带 Range 的顺序读 + 失败重连 + cancel 中断传播。 - 调整 FFT 路径：统一使用
TARGET_SAMPLE_RATE，不再单独 44.1k 重采样；同步更新构建/CI 文档与依赖。
Reviewed changes
Copilot reviewed 13 out of 14 changed files in this pull request and generated 4 comments.
Show a summary per file
| File | Description |
|---|---|
| native/audio-engine/src/scanner.rs | 扫库 probe 改用 AudioReader 读取时长/标签/封面缩略图 |
| native/audio-engine/src/player.rs | FFT analyzer 初始化采样率切到 TARGET_SAMPLE_RATE |
| native/audio-engine/src/metadata.rs | 元数据/tag/封面/歌词提取改为适配 ffmpeg_audio 的 HashMap/cover API |
| native/audio-engine/src/lib.rs | 初始化 logger 时改用 ffmpeg_audio 的日志级别控制；注册新模块 |
| native/audio-engine/src/http_source.rs | 新增 HTTP Range Read+Seek 数据源、重连/退避/cancel 机制及单测 |
| native/audio-engine/src/fft.rs | 移除手写 Send/Sync，实现维持 FFT 分析逻辑不变 |
| native/audio-engine/src/decoder.rs | 解码主循环改为 AudioReader::receive_frame 推 chunk，并接入 HttpRangeSource |
| native/audio-engine/Cargo.toml | 替换 ffmpeg-next 为 git 版 ffmpeg_audio；新增 ureq/httpmock |
| CLAUDE.md | 更新原生构建说明：静态 FFmpeg 由 ffmpeg_audio 内置构建 |
| Cargo.lock | 锁文件更新（引入 ffmpeg_audio/ureq/httpmock 等依赖树） |
| .github/workflows/dev.yml | CI 不再下载外部 FFmpeg 包；精简 Linux 依赖 |
| .github/workflows/claude.yml | 删除 Claude Code workflow |
| .github/workflows/claude-code-review.yml | 删除 Claude Code Review workflow |
| .github/copilot-instructions.md | 更新仓库指引：audio-engine 改用 ffmpeg_audio + HttpRangeSource |
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| let status = resp.status(); | ||
| let (total_size, range_supported) = match status { | ||
| 206 => { | ||
| let cr = resp | ||
| .header("Content-Range") | ||
| .ok_or_else(|| anyhow!("206 响应缺少 Content-Range 头: {url}"))?; | ||
| let total = parse_content_range_total(cr) | ||
| .ok_or_else(|| anyhow!("Content-Range 解析失败: {cr}"))?; | ||
| (total, true) | ||
| } | ||
| 200 => { | ||
| let cl = resp | ||
| .header("Content-Length") | ||
| .ok_or_else(|| anyhow!("200 响应缺少 Content-Length，无法估算长度: {url}"))?; | ||
| let total: u64 = cl | ||
| .parse() | ||
| .with_context(|| format!("Content-Length 解析失败: {cl}"))?; | ||
| warn!(url, "服务端不支持 Range（返回 200），seek 将失败"); | ||
| (total, false) | ||
| } | ||
| s => return Err(anyhow!("初始 GET 意外状态: {s}")), |
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| 200 => { | ||
| let cl = resp | ||
| .header("Content-Length") | ||
| .ok_or_else(|| anyhow!("200 响应缺少 Content-Length，无法估算长度: {url}"))?; | ||
| let total: u64 = cl | ||
| .parse() | ||
| .with_context(|| format!("Content-Length 解析失败: {cl}"))?; | ||
| warn!(url, "服务端不支持 Range（返回 200），seek 将失败"); | ||
| (total, false) | ||
| } |
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| let samples = match data.reader.receive_frame() { | ||
| Ok(Some(s)) => s.to_vec(), | ||
| Ok(None) => return, | ||
| Err(AudioError::Eof) => return, | ||
| Err(e) => { | ||
| warn!(error = %e, "解码失败"); | ||
| shared.mark_decode_failed(); | ||
| return; | ||
| } | ||
| Err(ffmpeg::Error::Other { errno }) if errno == ffmpeg::ffi::EAGAIN => { | ||
| if eof_sent { | ||
| // 已发送 EOF，解码器清空完毕，flush resampler 尾巴 | ||
| flush_and_push_tail(data, shared); | ||
| return; | ||
| } | ||
| }; | ||
|
|
||
| // 读取下一个数据包；Packet::read 能区分真正的 EOF 与读取失败 | ||
| let mut packet = ffmpeg::codec::packet::Packet::empty(); | ||
| match packet.read(&mut data.input_ctx) { | ||
| Ok(()) => { | ||
| if packet.stream() == data.audio_stream_index { | ||
| consecutive_read_failures = 0; | ||
| if data.decoder.send_packet(&packet).is_err() { | ||
| return; | ||
| } | ||
| } | ||
| // 非音频包：跳过 | ||
| } | ||
| Err(ffmpeg::Error::Eof) => { | ||
| // 真正的文件结束，刷新解码器 | ||
| let _ = data.decoder.send_eof(); | ||
| eof_sent = true; | ||
| } | ||
| Err(err) => { | ||
| // 读取失败：网络中断 / URL 失效 | ||
| consecutive_read_failures += 1; | ||
| warn!(error = %err, retries = consecutive_read_failures, "音频源读取失败"); | ||
| if consecutive_read_failures <= NETWORK_READ_RETRIES { | ||
| // 网络流可能恢复，等待后重试 | ||
| std::thread::sleep(std::time::Duration::from_millis( | ||
| NETWORK_RETRY_DELAY_MS, | ||
| )); | ||
| continue; | ||
| } | ||
| // 重试耗尽：标记音源失效，仍走 EOF 流程把已缓冲数据放完 | ||
| shared.mark_decode_failed(); | ||
| let _ = data.decoder.send_eof(); | ||
| eof_sent = true; | ||
| } | ||
| } | ||
| } | ||
| Err(err) => { | ||
| // 其他解码错误，重试后放弃 | ||
| consecutive_read_failures += 1; | ||
| warn!(error = %err, retries = consecutive_read_failures, "解码错误"); | ||
| if consecutive_read_failures <= NETWORK_READ_RETRIES { | ||
| std::thread::sleep(std::time::Duration::from_millis(NETWORK_RETRY_DELAY_MS)); | ||
| continue; | ||
| } | ||
| return; | ||
| } | ||
| if samples.is_empty() { | ||
| continue; | ||
| } | ||
| } | ||
| } | ||
|
|
||
| /// 对解码帧进行重采样，分别输出播放数据和 FFT 数据 | ||
| fn resample_frame( | ||
| data: &mut DecoderData, | ||
| decoded: &mut ffmpeg::frame::Audio, | ||
| player_buf: &mut Vec<f32>, | ||
| fft_buf: &mut Vec<f32>, | ||
| ) { | ||
| // 播放重采样 | ||
| if let Some(ref mut resampler) = data.resampler { | ||
| if let Some(frame) = try_resample(resampler, decoded) { | ||
| interleave_planar_frame(player_buf, &frame, frame.samples()); | ||
| } | ||
| } else { | ||
| interleave_planar_frame(player_buf, decoded, decoded.samples()); | ||
| } | ||
|
|
||
| // FFT 重采样（单声道） | ||
| if let Some(ref mut resampler) = data.fft_resampler { | ||
| if let Some(frame) = try_resample(resampler, decoded) { | ||
| let samples = frame.samples(); | ||
| if samples > 0 { | ||
| fft_buf.extend_from_slice(&frame.plane::<f32>(0)[..samples]); | ||
| } | ||
| } | ||
| } else { | ||
| let samples = decoded.samples(); | ||
| if samples > 0 { | ||
| fft_buf.extend_from_slice(&decoded.plane::<f32>(0)[..samples]); | ||
| } | ||
| } | ||
| } | ||
|
|
||
| /// 执行重采样，为每次调用创建正确尺寸的输出帧 | ||
| fn try_resample( | ||
| resampler: &mut ffmpeg::software::resampling::Context, | ||
| decoded: &ffmpeg::frame::Audio, | ||
| ) -> Option<ffmpeg::frame::Audio> { | ||
| let output_samples = (decoded.samples() as f64 * resampler.output().rate as f64 | ||
| / decoded.rate() as f64) | ||
| .ceil() as usize; | ||
|
|
||
| let mut output_frame = ffmpeg::frame::Audio::new( | ||
| resampler.output().format, | ||
| output_samples, | ||
| resampler.output().channel_layout, | ||
| ); | ||
| let mut player_samples = samples; | ||
|
|
||
| if resampler.run(decoded, &mut output_frame).is_ok() { | ||
| Some(output_frame) | ||
| } else { | ||
| None | ||
| } | ||
| } | ||
| // FFT 取左声道（48k mono），fft.rs 直接用 TARGET_SAMPLE_RATE 不再单独重采样 | ||
| let fft_samples: Vec<f32> = player_samples.chunks_exact(2).map(|c| c[0]).collect(); |
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| napi-derive = "3.5" | ||
| ffmpeg-next = { version = "8", default-features = false, features = [ | ||
| "codec", | ||
| "format", | ||
| "software-resampling", | ||
| "static", | ||
| ] } | ||
| ffmpeg_audio = { git = "https://github.com/SPlayer-Dev/ffmpeg-audio", branch = "feat/duration-api", default-features = false } | ||
| ureq = { version = "2", default-features = false, features = ["tls"] } |
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