Step 7.4: HLS audio on AES67 (44.1 -> 48 kHz into the ring)
- decoder: s16 PCM -> stereo int32 -> esp_ae_rate_cvt 44.1 -> 48 kHz (32-bit, complexity 3; bypassed at 48 kHz; mono duplicated) -> ring. esp_audio_effects pinned to ~1.3.0 (1.4+ needs P4 rev >= 3). - hls: each segment is downloaded completely into PSRAM (max 4 MB), then decoded; the connection is not held open while the decoder waits for ring space at playback speed. - player: player_write() blocks while the ring is full and gives up when the source changes; the temporary 440 Hz producer is removed. - Verified with Triple J Hottest: 441344 -> 480375 frames (10.008 s) and 440320 -> 479260 (9.985 s) per segment; RTP 15000 packets, 0 gaps, peak -10 dBFS / RMS -22 dBFS; ring ~4.0 s, 0 underruns over ~50 s; heap 422 KB, PSRAM 27.5 MB free. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
This commit is contained in:
+67
-5
@@ -6,11 +6,21 @@
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#include "esp_audio_dec_default.h"
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#include "esp_audio_simple_dec.h"
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#include "esp_audio_simple_dec_default.h"
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#include "esp_ae_rate_cvt.h"
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#include "player.h"
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#include "esp_heap_caps.h"
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#include "esp_log.h"
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static const char *TAG = "decoder";
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#define OUT_RATE 48000
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#define CONV_FRAMES 4096 // per rate converter call
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static esp_ae_rate_cvt_handle_t s_cvt; // NULL: no conversion (source is 48 kHz)
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static uint32_t s_cvt_rate; // input rate the converter was opened for
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static int32_t *s_in32, *s_out32; // stereo int32 work buffers (PSRAM)
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static uint64_t s_seg_out; // 48 kHz frames written this segment
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static esp_audio_simple_dec_handle_t s_dec;
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static uint8_t *s_pcm;
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static uint32_t s_pcm_size = 8192;
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@@ -25,15 +35,61 @@ static int s_pmt_pid = -1, s_audio_pid = -1;
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static void report_segment(void)
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{
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if (s_seg_count && s_info.sample_rate) {
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ESP_LOGI(TAG, "segment: %llu ES bytes -> %llu PCM frames = %.3f s (%lu Hz, %u ch, %u bit)",
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s_seg_es, s_seg_frames, (double)s_seg_frames / s_info.sample_rate,
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(unsigned long)s_info.sample_rate, s_info.channel, s_info.bits_per_sample);
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ESP_LOGI(TAG, "segment: %llu ES bytes -> %llu frames at %lu Hz = %.3f s -> %llu frames at 48 kHz = %.3f s",
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s_seg_es, s_seg_frames, (unsigned long)s_info.sample_rate,
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(double)s_seg_frames / s_info.sample_rate, s_seg_out, (double)s_seg_out / OUT_RATE);
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}
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s_seg_frames = 0;
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s_seg_es = 0;
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s_seg_out = 0;
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s_seg_count++;
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}
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// Decoded PCM (s16, any channel count) -> stereo int32 -> 48 kHz -> ring.
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static void output_pcm(const int16_t *pcm, uint32_t frames)
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{
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int ch = s_info.channel;
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if (s_info.sample_rate != s_cvt_rate) {
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if (s_cvt) {
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esp_ae_rate_cvt_close(s_cvt);
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s_cvt = NULL;
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}
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s_cvt_rate = s_info.sample_rate;
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if (s_cvt_rate != OUT_RATE) {
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esp_ae_rate_cvt_cfg_t cfg = {
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.src_rate = s_cvt_rate, .dest_rate = OUT_RATE, .channel = 2, .bits_per_sample = 32,
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.complexity = 3, .perf_type = ESP_AE_RATE_CVT_PERF_TYPE_SPEED,
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};
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if (esp_ae_rate_cvt_open(&cfg, &s_cvt) != ESP_AE_ERR_OK) {
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ESP_LOGE(TAG, "rate converter %lu -> %d Hz: open failed", (unsigned long)s_cvt_rate, OUT_RATE);
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} else {
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ESP_LOGI(TAG, "rate converter %lu -> %d Hz", (unsigned long)s_cvt_rate, OUT_RATE);
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}
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}
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}
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while (frames) {
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uint32_t n = frames < CONV_FRAMES ? frames : CONV_FRAMES;
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for (uint32_t i = 0; i < n; i++) { // to stereo int32 (full scale = INT32_MAX)
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int32_t l = (int32_t)pcm[i * ch] << 16;
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s_in32[i * 2] = l;
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s_in32[i * 2 + 1] = ch > 1 ? (int32_t)pcm[i * ch + 1] << 16 : l;
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}
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const int32_t *out = s_in32;
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uint32_t out_n = n;
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if (s_cvt) {
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out_n = CONV_FRAMES * 2;
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if (esp_ae_rate_cvt_process(s_cvt, s_in32, n, s_out32, &out_n) != ESP_AE_ERR_OK) {
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ESP_LOGW(TAG, "rate conversion failed");
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return;
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}
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out = s_out32;
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}
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s_seg_out += player_write(out, out_n);
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pcm += n * ch;
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frames -= n;
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}
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}
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// Feed ADTS-AAC elementary stream bytes to the decoder.
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static void decode_es(const uint8_t *data, size_t len)
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{
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@@ -62,7 +118,11 @@ static void decode_es(const uint8_t *data, size_t len)
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ESP_LOGI(TAG, "stream: %lu Hz, %u ch, %u bit, %lu bit/s", (unsigned long)s_info.sample_rate,
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s_info.channel, s_info.bits_per_sample, (unsigned long)s_info.bitrate);
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}
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s_seg_frames += out.decoded_size / (s_info.channel * s_info.bits_per_sample / 8);
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uint32_t n = out.decoded_size / (s_info.channel * s_info.bits_per_sample / 8);
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s_seg_frames += n;
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if (s_info.bits_per_sample == 16) {
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output_pcm((const int16_t *)out.buffer, n);
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}
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}
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raw.buffer += raw.consumed;
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raw.len -= raw.consumed;
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@@ -168,5 +228,7 @@ esp_err_t decoder_init(void)
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return ESP_FAIL;
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}
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s_pcm = heap_caps_malloc(s_pcm_size, MALLOC_CAP_SPIRAM);
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return s_pcm ? ESP_OK : ESP_ERR_NO_MEM;
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s_in32 = heap_caps_malloc(CONV_FRAMES * 2 * sizeof(int32_t), MALLOC_CAP_SPIRAM);
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s_out32 = heap_caps_malloc(CONV_FRAMES * 2 * 2 * sizeof(int32_t), MALLOC_CAP_SPIRAM);
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return s_pcm && s_in32 && s_out32 ? ESP_OK : ESP_ERR_NO_MEM;
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}
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+15
-17
@@ -19,6 +19,7 @@
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#define LIVE_BACK 3 // start this many segments behind the live edge
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#define CHUNK 4096
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#define MAX_REDIRECTS 5
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#define SEG_MAX (4 * 1024 * 1024) // largest segment we accept (PSRAM)
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static const char *TAG = "hls";
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@@ -34,6 +35,7 @@ static char *s_text; // playlist buffer (PSRAM)
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static media_pl_t *s_pl; // parsed media playlist (PSRAM)
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static char s_url[URL_MAX]; // configured URL
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static char s_media_url[URL_MAX];
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static uint8_t *s_seg; // current segment (PSRAM)
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// hls_url when source.mode needs HLS, else "".
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static void wanted_url(char *out, size_t n)
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@@ -209,30 +211,25 @@ static bool fetch_segment(long long seq, const char *url)
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if (!c) {
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return false;
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}
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static uint8_t buf[CHUNK];
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// Download completely first, then decode: the connection is not held open while the
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// decoder waits for space in the ring (a stalled TCP stream for ~10 s may get cut by the CDN).
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size_t total = 0;
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int r;
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bool ok = true, first = true;
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while ((r = esp_http_client_read(c, (char *)buf, sizeof(buf))) > 0) {
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if (s_sink && !s_sink(buf, r, first)) {
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ok = false;
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break;
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}
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first = false;
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int r = 0;
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bool ok = true;
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while (total < SEG_MAX && (r = esp_http_client_read(c, (char *)s_seg + total, SEG_MAX - total)) > 0) {
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total += r;
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if (!still_wanted()) {
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ok = false;
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break;
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}
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}
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if (r < 0) {
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if (r < 0 || total == SEG_MAX) {
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ok = false;
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}
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esp_http_client_close(c);
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esp_http_client_cleanup(c);
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int64_t us = esp_timer_get_time() - t0;
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ESP_LOGI(TAG, "segment %lld: %u bytes in %.2f s (%.1f Mbit/s)%s", seq, (unsigned)total, us / 1e6,
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us ? total * 8.0 / us : 0.0, ok ? "" : ", aborted");
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ESP_LOGD(TAG, "segment %lld: %u bytes in %.2f s (%.1f Mbit/s)%s", seq, (unsigned)total, us / 1e6,
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us ? total * 8.0 / us : 0.0, ok ? "" : ", failed");
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if (ok && s_sink) {
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ok = s_sink(s_seg, total, true); // blocks at playback speed while the ring is full
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}
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return ok;
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}
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@@ -300,7 +297,8 @@ esp_err_t hls_start(hls_sink_t sink)
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s_sink = sink;
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s_text = heap_caps_malloc(PLAYLIST_MAX, MALLOC_CAP_SPIRAM);
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s_pl = heap_caps_malloc(sizeof(media_pl_t), MALLOC_CAP_SPIRAM);
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if (!s_text || !s_pl) {
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s_seg = heap_caps_malloc(SEG_MAX, MALLOC_CAP_SPIRAM);
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if (!s_text || !s_pl || !s_seg) {
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return ESP_ERR_NO_MEM;
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}
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return xTaskCreate(hls_task, "hls", 10240, NULL, 4, NULL) == pdPASS ? ESP_OK : ESP_ERR_NO_MEM;
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@@ -1,3 +1,4 @@
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dependencies:
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# 2.6+ uses P4 assembly that needs chip rev >= 3.0; this board is rev 1.3 (see CLAUDE.md).
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# Newer versions use P4 assembly that needs chip rev >= 3.0; this board is rev 1.3 (see CLAUDE.md).
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espressif/esp_audio_codec: "~2.5.0"
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espressif/esp_audio_effects: "~1.3.0"
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+13
-17
@@ -1,6 +1,5 @@
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#include "player.h"
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#include <math.h>
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#include <string.h>
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#include "aes67_cfg.h"
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@@ -72,24 +71,22 @@ void player_apply(const cJSON *source)
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ESP_LOGI(TAG, "source: %s", SRC_NAME[src]);
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}
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/* ----- Step 7.1: temporary test producer (440 Hz into the ring), replaced by the HLS player ----- */
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static void test_producer(void *arg)
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// Source side (HLS task): write converted 48 kHz frames, waiting for space at playback speed.
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// Gives up when the source is no longer HLS.
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size_t player_write(const int32_t *frames, size_t n)
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{
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static int32_t chunk[480 * CHANNELS];
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uint32_t phase = 0;
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const double amp = pow(10.0, -18.0 / 20.0) * 2147483647.0;
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while (1) {
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if (s_src != SRC_HLS || audio_ring_space() < 480) {
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vTaskDelay(pdMS_TO_TICKS(10));
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continue;
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size_t done = 0;
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while (done < n) {
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if (s_src != SRC_HLS) {
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return done;
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}
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for (int i = 0; i < 480; i++, phase++) {
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int32_t v = (int32_t)(amp * sin(2.0 * M_PI * 440.0 * (phase % RATE) / RATE));
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chunk[i * 2] = chunk[i * 2 + 1] = v;
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size_t w = audio_ring_write(frames + done * CHANNELS, n - done);
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if (!w) {
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vTaskDelay(pdMS_TO_TICKS(20));
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}
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audio_ring_write(chunk, 480);
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done += w;
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}
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return done;
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}
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static void player_status(cJSON *st)
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@@ -113,12 +110,11 @@ esp_err_t player_init(void)
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s_src = (src_t)-1;
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player_apply(src);
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cJSON_Delete(src);
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xTaskCreate(test_producer, "test_prod", 3072, NULL, 5, NULL);
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err = decoder_init();
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if (err != ESP_OK) {
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return err;
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}
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err = hls_start(decoder_feed); // 7.3: decode and log; resampling + ring in 7.4
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err = hls_start(decoder_feed);
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if (err != ESP_OK) {
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return err;
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}
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@@ -1,9 +1,14 @@
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// Project player: picks the audio source for AES67 TX (test tone, silence, HLS, Spotify).
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#pragma once
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#include <stddef.h>
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#include <stdint.h>
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#include "cJSON.h"
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#include "esp_err.h"
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esp_err_t player_init(void);
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// Apply the "source" config group (mode etc.).
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void player_apply(const cJSON *source);
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// Source side: write 48 kHz stereo int32 frames, blocking while the ring is full.
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size_t player_write(const int32_t *frames, size_t n);
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