add flac support and files management app
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francis.wang committed
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12 files changed
+1526
-139
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+266
-15
@@ -34,6 +34,11 @@
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#define WAV_MAX_BLOCKALIGN 8 /* 支持的 WAV 帧对齐上限（16/24/32bit × 最多 2 声道） */
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#define WAV_MAX_FRAMES 1024 /* WAV 原始 PCM 中转缓冲容纳的最大帧数 */
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#define AUDIO_SRC_MAX_FRAMES 1024 /* 单次 audio_source_read 最多输出的帧数（与播放器缓冲一致） */
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#define DECIM_CHUNK_FRAMES 256 /* 抽取时每次处理多少输出帧（控制临时缓冲大小） */
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#define AUDIO_MAX_OUT_RATE 48000 /* 输出采样率上限 */
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#define AUDIO_MAX_DECIM 8 /* 最大抽取倍数 */
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/* ---------------- 内部结构 ---------------- */
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struct audio_source {
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audio_codec_t codec;
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@@ -54,6 +59,10 @@ struct audio_source {
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/* ---- FLAC ---- */
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flac_dec_t *flac;
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/* ---- 高采样率抽取（192k/96k → ≤48k） ---- */
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uint8_t decim; /* 抽取倍数，1 表示不抽取 */
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uint8_t *decim_buf; /* 抽取用的原生采样中转缓冲 */
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};
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/* ---------------- 位宽映射 ---------------- */
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@@ -76,6 +85,10 @@ static inline uint16_t pick_out_bits(uint16_t bits)
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return 32;
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}
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/* 采样率限制 / 抽取（实现见文件末尾，供 audio_source_open 使用） */
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static bool decim_setup(audio_source_t *src);
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static void decim_mix(const uint8_t *src, uint8_t *dst, int frames, int decim, uint8_t ob);
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/* ---------------- WAV 解析 ---------------- */
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static inline uint16_t rd_u16(const uint8_t *p)
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@@ -256,8 +269,7 @@ static int wav_read(audio_source_t *src, void *out, int max_frames)
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}
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}
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src->frames_read += frames;
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/* 注意：frames_read 统一由 audio_source_read() 累加，这里不要重复计数 */
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return (int)frames;
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}
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@@ -323,14 +335,25 @@ audio_source_t *audio_source_open(const char *path, audio_info_t *info)
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src->info.total_sec = (fi.total_samples > 0) ?
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(uint32_t)(fi.total_samples / fi.sample_rate) : 0;
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/* 高采样率（96k/192k）抽取到 ES8388 / MCLK 能承受的范围 */
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if (!decim_setup(src))
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{
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ESP_LOGE(TAG, "unsupported sample rate %u Hz: %s",
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(unsigned)fi.sample_rate, path);
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audio_source_close(src);
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return NULL;
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}
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if (info != NULL)
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{
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*info = src->info;
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}
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ESP_LOGI(TAG, "%s: FLAC %u Hz, %u bit -> I2S %u bit, %u ch, %u s",
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path, (unsigned)src->info.sample_rate, src->info.bits,
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src->info.out_bits, src->info.channels,
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ESP_LOGI(TAG, "%s: FLAC %u Hz%s, %u bit -> I2S %u bit @ %u Hz, %u ch, %u s",
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path, (unsigned)fi.sample_rate,
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(src->decim > 1) ? " (downsampled)" : "",
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src->info.bits, src->info.out_bits,
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(unsigned)src->info.sample_rate, src->info.channels,
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(unsigned)src->info.total_sec);
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return src;
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@@ -359,6 +382,15 @@ audio_source_t *audio_source_open(const char *path, audio_info_t *info)
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src->info.out_bytes = (uint8_t)(bits / 8); /* WAV 位深必为 16/24/32 */
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src->info.total_sec = src->data_size / src->byte_rate;
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/* 高采样率（96k/192k）抽取到 ES8388 / MCLK 能承受的范围 */
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if (!decim_setup(src))
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{
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ESP_LOGW(TAG, "unsupported sample rate %u Hz: %s",
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(unsigned)src->info.sample_rate, path);
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audio_source_close(src);
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return NULL;
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}
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/* 原始 PCM 中转缓冲（16/24/32bit × 最多 2 声道 → 每帧最多 8 字节） */
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src->scratch = (uint8_t *)malloc((size_t)WAV_MAX_FRAMES * src->block_align);
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@@ -385,6 +417,87 @@ audio_source_t *audio_source_open(const char *path, audio_info_t *info)
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return src;
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}
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/**
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* @brief 带抽取的读取：把 decim 个原生帧合成 1 个输出帧
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*/
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static int audio_source_read_decim(audio_source_t *src, void *out, int max_frames)
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{
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uint8_t *dst = (uint8_t *)out;
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const uint8_t ob = src->info.out_bytes;
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const int decim = src->decim;
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int produced = 0;
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while (produced < max_frames)
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{
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int room = max_frames - produced;
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if (room > DECIM_CHUNK_FRAMES)
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{
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room = DECIM_CHUNK_FRAMES;
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}
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if ((room * decim) > WAV_MAX_FRAMES)
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{
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room = WAV_MAX_FRAMES / decim; /* 受中转缓冲帧数限制 */
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}
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if (room <= 0)
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{
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break;
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}
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/* 先凑满 room * decim 个原生帧，避免每次少读几帧造成丢样本 */
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const int need = room * decim;
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int have = 0;
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bool eof = false;
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while (have < need)
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{
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int got = (src->codec == AUDIO_CODEC_FLAC)
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? flac_dec_read(src->flac,
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src->decim_buf + (size_t)have * 2u * ob,
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need - have)
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: wav_read(src,
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src->decim_buf + (size_t)have * 2u * ob,
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need - have);
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if (got < 0)
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{
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return -1;
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}
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if (got == 0)
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{
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eof = true;
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break;
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}
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have += got;
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}
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int frames = have / decim;
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if (frames > 0)
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{
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decim_mix(src->decim_buf, dst + (size_t)produced * 2u * ob,
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frames, decim, ob);
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produced += frames;
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}
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if (eof)
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{
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break; /* 文件末尾 */
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}
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}
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if (produced > 0)
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{
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src->frames_read += (uint64_t)produced;
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}
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return produced;
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}
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int audio_source_read(audio_source_t *src, void *out, int max_frames)
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{
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if ((src == NULL) || (out == NULL) || (max_frames <= 0))
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@@ -392,19 +505,26 @@ int audio_source_read(audio_source_t *src, void *out, int max_frames)
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return -1;
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}
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if (src->codec == AUDIO_CODEC_FLAC)
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if (max_frames > AUDIO_SRC_MAX_FRAMES)
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{
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int n = flac_dec_read(src->flac, out, max_frames);
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if (n > 0)
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{
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src->frames_read += (uint64_t)n;
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}
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return n;
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max_frames = AUDIO_SRC_MAX_FRAMES; /* 保护调用者缓冲 */
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}
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return wav_read(src, out, max_frames);
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if (src->decim > 1)
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{
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return audio_source_read_decim(src, out, max_frames);
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}
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int n = (src->codec == AUDIO_CODEC_FLAC)
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? flac_dec_read(src->flac, out, max_frames)
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: wav_read(src, out, max_frames);
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if (n > 0)
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{
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src->frames_read += (uint64_t)n;
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}
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return n;
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}
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uint32_t audio_source_elapsed_sec(const audio_source_t *src)
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@@ -446,5 +566,136 @@ void audio_source_close(audio_source_t *src)
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free(src->scratch);
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src->scratch = NULL;
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free(src->decim_buf);
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src->decim_buf = NULL;
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free(src);
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}
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/* ---------------- 采样率限制 ---------------- */
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/**
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* @brief 计算抽取倍数（把采样率压到 AUDIO_MAX_OUT_RATE 以内）
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* @note 板载 ES8388 的 MCLK 由 I2S 产生，MCLK = mclk_multiple × fs.
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* 24bit 时 mclk_multiple 必须是 384（3 的倍数），若直接播放 192kHz,
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* MCLK 会高达 73.7MHz，远超 ES8388 的输入范围，结果是无声音或杂音.
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* 因此这里对 88.2k/96k/192k 等做整数抽取，统一落到 ≤48kHz.
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*/
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static uint8_t pick_decim(uint32_t rate)
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{
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uint8_t d = 1;
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while (((rate / d) > AUDIO_MAX_OUT_RATE) && (d < AUDIO_MAX_DECIM))
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{
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d++;
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}
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return d;
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}
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/**
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* @brief 按输出采样率为音源分配抽取缓冲
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* @retval true 成功；false 采样率过高或内存不足
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*/
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static bool decim_setup(audio_source_t *src)
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{
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uint8_t d = pick_decim(src->info.sample_rate);
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if ((src->info.sample_rate / d) > AUDIO_MAX_OUT_RATE)
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{
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return false; /* 采样率过高，无法用整数抽取覆盖 */
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}
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src->decim = d;
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if (d <= 1)
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{
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return true; /* 常见 44.1k/48k：完全不额外占用内存 */
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}
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size_t bytes = (size_t)DECIM_CHUNK_FRAMES * d * 2u * src->info.out_bytes;
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src->decim_buf = (uint8_t *)malloc(bytes);
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if (src->decim_buf == NULL)
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{
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return false;
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}
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/* 对外统一暴露输出采样率，I2S / 界面都按这个值工作 */
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src->info.sample_rate /= d;
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return true;
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}
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/* ---------------- 抽取用的采样读写 ---------------- */
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static inline int32_t pcm_sample_get(const uint8_t *p, uint8_t bytes)
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{
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switch (bytes)
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{
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case 2:
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return (int16_t)(p[0] | (p[1] << 8));
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case 3:
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{
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int32_t v = (int32_t)(p[0] | (p[1] << 8) | (p[2] << 16));
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if (v & 0x800000)
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{
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v -= 0x1000000;
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}
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return v;
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}
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default:
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return (int32_t)((uint32_t)p[0] | ((uint32_t)p[1] << 8) |
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((uint32_t)p[2] << 16) | ((uint32_t)p[3] << 24));
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}
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}
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static inline void pcm_sample_put(uint8_t *p, uint8_t bytes, int32_t v)
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{
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switch (bytes)
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{
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case 2:
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p[0] = (uint8_t)(v & 0xFF);
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p[1] = (uint8_t)((v >> 8) & 0xFF);
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break;
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case 3:
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p[0] = (uint8_t)(v & 0xFF);
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p[1] = (uint8_t)((v >> 8) & 0xFF);
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p[2] = (uint8_t)((v >> 16) & 0xFF);
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break;
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default:
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p[0] = (uint8_t)(v & 0xFF);
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p[1] = (uint8_t)((v >> 8) & 0xFF);
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p[2] = (uint8_t)((v >> 16) & 0xFF);
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p[3] = (uint8_t)((v >> 24) & 0xFF);
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break;
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}
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}
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/**
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* @brief 整数抽取：把 decim 帧平均成 1 帧（简易低通，抑制抽取混叠）
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*/
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static void decim_mix(const uint8_t *src, uint8_t *dst, int frames, int decim, uint8_t ob)
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{
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for (int i = 0; i < frames; i++)
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{
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for (int c = 0; c < 2; c++)
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{
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int64_t sum = 0;
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for (int k = 0; k < decim; k++)
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{
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const size_t idx = ((size_t)i * (size_t)decim + (size_t)k) * 2u + (size_t)c;
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sum += pcm_sample_get(src + idx * ob, ob);
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}
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pcm_sample_put(dst + ((size_t)i * 2u + (size_t)c) * ob, ob, (int32_t)(sum / decim));
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}
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}
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}
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@@ -21,6 +21,12 @@
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* 17~24bit → 24bit（每采样 3 字节）
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* 25~32bit → 32bit（每采样 4 字节）
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* 非整字节位深（如 20bit）按左对齐放入更宽的槽位，音量不受影响。
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*
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* 采样率限制：
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* ES8388 的 MCLK 由 I2S 产生（MCLK = mclk_multiple × fs），24bit 时
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* mclk_multiple 必须是 384。若直接播放 192kHz，MCLK 会到 73.7MHz，
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* 远超 codec 能力。因此 88.2k/96k/192k 等会做整数抽取降到 ≤48kHz，
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* audio_info_t.sample_rate 返回的始终是「实际输出采样率」。
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****************************************************************************************************
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*/
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@@ -43,7 +49,7 @@ typedef enum {
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/** 音源信息 */
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typedef struct {
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audio_codec_t codec; /* 容器类型 */
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uint32_t sample_rate; /* 采样率 → 用于配置 I2S 频率 */
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uint32_t sample_rate; /* 采样率 → 用于配置 I2S 频率（已按输出上限抽取） */
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uint16_t bits; /* 源位深（界面显示用） */
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uint16_t channels; /* 源声道数 */
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uint16_t out_bits; /* I2S 输出位宽：16 / 24 / 32 */
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+15
-20
@@ -292,6 +292,8 @@ static void audio_set_format(uint32_t sample_rate, uint16_t bits)
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/* ---------------- 曲目扫描 ---------------- */
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/**
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* @brief 判断是否为可播放的音频文件（.wav / .flac）
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* @note 两种扩展名都要检查，不能因为不是 .wav 就提前 return false,
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* 否则 .flac 永远被过滤掉。
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*/
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static bool name_is_audio(const char *name)
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{
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@@ -302,34 +304,27 @@ static bool name_is_audio(const char *name)
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return false;
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}
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const char *ext = name + len - 4; /* 指向 ".xxx" */
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const char *ext = name + len - 4; /* 指向最后 4 个字符 */
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if (ext[0] != '.')
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{
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return false;
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}
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/* .wav */
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if (((ext[1] == 'w') || (ext[1] == 'W')) &&
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/* .wav —— 最后 4 个字符为 ".wav" */
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if ((ext[0] == '.') &&
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((ext[1] == 'w') || (ext[1] == 'W')) &&
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((ext[2] == 'a') || (ext[2] == 'A')) &&
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((ext[3] == 'v') || (ext[3] == 'V')))
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{
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return true;
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}
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/* .flac（长度至少 5："a.flac"） */
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if (len >= 5)
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{
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const char *f = name + len - 5;
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/* .flac —— 最后 5 个字符为 ".flac" */
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const char *ext5 = name + len - 5;
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if ((f[0] == '.') &&
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((f[1] == 'f') || (f[1] == 'F')) &&
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((f[2] == 'l') || (f[2] == 'L')) &&
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((f[3] == 'a') || (f[3] == 'A')) &&
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((f[4] == 'c') || (f[4] == 'C')))
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{
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return true;
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}
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if ((ext5[0] == '.') &&
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((ext5[1] == 'f') || (ext5[1] == 'F')) &&
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((ext5[2] == 'l') || (ext5[2] == 'L')) &&
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((ext5[3] == 'a') || (ext5[3] == 'A')) &&
|
||||
((ext5[4] == 'c') || (ext5[4] == 'C')))
|
||||
{
|
||||
return true;
|
||||
}
|
||||
|
||||
return false;
|
||||
|
||||
Reference in new issue
Block a user