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:
2026-09-25 14:52:19 +10:00
parent 8a6cb53e3b
commit 3ebb27e4bb
6 changed files with 122 additions and 41 deletions
+20 -1
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@@ -9,6 +9,24 @@ dependencies:
registry_url: https://components.espressif.com/ registry_url: https://components.espressif.com/
type: service type: service
version: 2.5.0 version: 2.5.0
espressif/esp_audio_effects:
component_hash: 6ad72ee106e25b07c9ae5b8ec77b566ef8d697473095f1c982e13247005b9ba6
dependencies:
- name: espressif/gmf_fft
registry_url: https://components.espressif.com
require: private
version: ~1.0
source:
registry_url: https://components.espressif.com/
type: service
version: 1.3.0~1
espressif/gmf_fft:
component_hash: b59451ef2f96d22b80c27a162515b21f1b85954dda434c0b092172164028779d
dependencies: []
source:
registry_url: https://components.espressif.com
type: service
version: 1.0.0
espressif/mdns: espressif/mdns:
component_hash: b679eafd0acae2066e2645bd91d073e33ca5be515e2545cd3c4e55a3e3bce3cb component_hash: b679eafd0acae2066e2645bd91d073e33ca5be515e2545cd3c4e55a3e3bce3cb
dependencies: dependencies:
@@ -25,7 +43,8 @@ dependencies:
version: 5.5.5 version: 5.5.5
direct_dependencies: direct_dependencies:
- espressif/esp_audio_codec - espressif/esp_audio_codec
- espressif/esp_audio_effects
- espressif/mdns - espressif/mdns
manifest_hash: e85f2bff1cc890def1a35e7da5210473c09ec845364a2bc2e54f6d2a7de54cbc manifest_hash: 95404bd3492778da35a8aeee53935ee8ad13b09c56ae5fcbe77a4e921dafbb23
target: esp32p4 target: esp32p4
version: 2.0.0 version: 2.0.0
+67 -5
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@@ -6,11 +6,21 @@
#include "esp_audio_dec_default.h" #include "esp_audio_dec_default.h"
#include "esp_audio_simple_dec.h" #include "esp_audio_simple_dec.h"
#include "esp_audio_simple_dec_default.h" #include "esp_audio_simple_dec_default.h"
#include "esp_ae_rate_cvt.h"
#include "player.h"
#include "esp_heap_caps.h" #include "esp_heap_caps.h"
#include "esp_log.h" #include "esp_log.h"
static const char *TAG = "decoder"; static const char *TAG = "decoder";
#define OUT_RATE 48000
#define CONV_FRAMES 4096 // per rate converter call
static esp_ae_rate_cvt_handle_t s_cvt; // NULL: no conversion (source is 48 kHz)
static uint32_t s_cvt_rate; // input rate the converter was opened for
static int32_t *s_in32, *s_out32; // stereo int32 work buffers (PSRAM)
static uint64_t s_seg_out; // 48 kHz frames written this segment
static esp_audio_simple_dec_handle_t s_dec; static esp_audio_simple_dec_handle_t s_dec;
static uint8_t *s_pcm; static uint8_t *s_pcm;
static uint32_t s_pcm_size = 8192; static uint32_t s_pcm_size = 8192;
@@ -25,15 +35,61 @@ static int s_pmt_pid = -1, s_audio_pid = -1;
static void report_segment(void) static void report_segment(void)
{ {
if (s_seg_count && s_info.sample_rate) { if (s_seg_count && s_info.sample_rate) {
ESP_LOGI(TAG, "segment: %llu ES bytes -> %llu PCM frames = %.3f s (%lu Hz, %u ch, %u bit)", ESP_LOGI(TAG, "segment: %llu ES bytes -> %llu frames at %lu Hz = %.3f s -> %llu frames at 48 kHz = %.3f s",
s_seg_es, s_seg_frames, (double)s_seg_frames / s_info.sample_rate, s_seg_es, s_seg_frames, (unsigned long)s_info.sample_rate,
(unsigned long)s_info.sample_rate, s_info.channel, s_info.bits_per_sample); (double)s_seg_frames / s_info.sample_rate, s_seg_out, (double)s_seg_out / OUT_RATE);
} }
s_seg_frames = 0; s_seg_frames = 0;
s_seg_es = 0; s_seg_es = 0;
s_seg_out = 0;
s_seg_count++; s_seg_count++;
} }
// Decoded PCM (s16, any channel count) -> stereo int32 -> 48 kHz -> ring.
static void output_pcm(const int16_t *pcm, uint32_t frames)
{
int ch = s_info.channel;
if (s_info.sample_rate != s_cvt_rate) {
if (s_cvt) {
esp_ae_rate_cvt_close(s_cvt);
s_cvt = NULL;
}
s_cvt_rate = s_info.sample_rate;
if (s_cvt_rate != OUT_RATE) {
esp_ae_rate_cvt_cfg_t cfg = {
.src_rate = s_cvt_rate, .dest_rate = OUT_RATE, .channel = 2, .bits_per_sample = 32,
.complexity = 3, .perf_type = ESP_AE_RATE_CVT_PERF_TYPE_SPEED,
};
if (esp_ae_rate_cvt_open(&cfg, &s_cvt) != ESP_AE_ERR_OK) {
ESP_LOGE(TAG, "rate converter %lu -> %d Hz: open failed", (unsigned long)s_cvt_rate, OUT_RATE);
} else {
ESP_LOGI(TAG, "rate converter %lu -> %d Hz", (unsigned long)s_cvt_rate, OUT_RATE);
}
}
}
while (frames) {
uint32_t n = frames < CONV_FRAMES ? frames : CONV_FRAMES;
for (uint32_t i = 0; i < n; i++) { // to stereo int32 (full scale = INT32_MAX)
int32_t l = (int32_t)pcm[i * ch] << 16;
s_in32[i * 2] = l;
s_in32[i * 2 + 1] = ch > 1 ? (int32_t)pcm[i * ch + 1] << 16 : l;
}
const int32_t *out = s_in32;
uint32_t out_n = n;
if (s_cvt) {
out_n = CONV_FRAMES * 2;
if (esp_ae_rate_cvt_process(s_cvt, s_in32, n, s_out32, &out_n) != ESP_AE_ERR_OK) {
ESP_LOGW(TAG, "rate conversion failed");
return;
}
out = s_out32;
}
s_seg_out += player_write(out, out_n);
pcm += n * ch;
frames -= n;
}
}
// Feed ADTS-AAC elementary stream bytes to the decoder. // Feed ADTS-AAC elementary stream bytes to the decoder.
static void decode_es(const uint8_t *data, size_t len) static void decode_es(const uint8_t *data, size_t len)
{ {
@@ -62,7 +118,11 @@ static void decode_es(const uint8_t *data, size_t len)
ESP_LOGI(TAG, "stream: %lu Hz, %u ch, %u bit, %lu bit/s", (unsigned long)s_info.sample_rate, ESP_LOGI(TAG, "stream: %lu Hz, %u ch, %u bit, %lu bit/s", (unsigned long)s_info.sample_rate,
s_info.channel, s_info.bits_per_sample, (unsigned long)s_info.bitrate); s_info.channel, s_info.bits_per_sample, (unsigned long)s_info.bitrate);
} }
s_seg_frames += out.decoded_size / (s_info.channel * s_info.bits_per_sample / 8); uint32_t n = out.decoded_size / (s_info.channel * s_info.bits_per_sample / 8);
s_seg_frames += n;
if (s_info.bits_per_sample == 16) {
output_pcm((const int16_t *)out.buffer, n);
}
} }
raw.buffer += raw.consumed; raw.buffer += raw.consumed;
raw.len -= raw.consumed; raw.len -= raw.consumed;
@@ -168,5 +228,7 @@ esp_err_t decoder_init(void)
return ESP_FAIL; return ESP_FAIL;
} }
s_pcm = heap_caps_malloc(s_pcm_size, MALLOC_CAP_SPIRAM); s_pcm = heap_caps_malloc(s_pcm_size, MALLOC_CAP_SPIRAM);
return s_pcm ? ESP_OK : ESP_ERR_NO_MEM; s_in32 = heap_caps_malloc(CONV_FRAMES * 2 * sizeof(int32_t), MALLOC_CAP_SPIRAM);
s_out32 = heap_caps_malloc(CONV_FRAMES * 2 * 2 * sizeof(int32_t), MALLOC_CAP_SPIRAM);
return s_pcm && s_in32 && s_out32 ? ESP_OK : ESP_ERR_NO_MEM;
} }
+15 -17
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@@ -19,6 +19,7 @@
#define LIVE_BACK 3 // start this many segments behind the live edge #define LIVE_BACK 3 // start this many segments behind the live edge
#define CHUNK 4096 #define CHUNK 4096
#define MAX_REDIRECTS 5 #define MAX_REDIRECTS 5
#define SEG_MAX (4 * 1024 * 1024) // largest segment we accept (PSRAM)
static const char *TAG = "hls"; static const char *TAG = "hls";
@@ -34,6 +35,7 @@ static char *s_text; // playlist buffer (PSRAM)
static media_pl_t *s_pl; // parsed media playlist (PSRAM) static media_pl_t *s_pl; // parsed media playlist (PSRAM)
static char s_url[URL_MAX]; // configured URL static char s_url[URL_MAX]; // configured URL
static char s_media_url[URL_MAX]; static char s_media_url[URL_MAX];
static uint8_t *s_seg; // current segment (PSRAM)
// hls_url when source.mode needs HLS, else "". // hls_url when source.mode needs HLS, else "".
static void wanted_url(char *out, size_t n) static void wanted_url(char *out, size_t n)
@@ -209,30 +211,25 @@ static bool fetch_segment(long long seq, const char *url)
if (!c) { if (!c) {
return false; return false;
} }
static uint8_t buf[CHUNK]; // Download completely first, then decode: the connection is not held open while the
// decoder waits for space in the ring (a stalled TCP stream for ~10 s may get cut by the CDN).
size_t total = 0; size_t total = 0;
int r; int r = 0;
bool ok = true, first = true; bool ok = true;
while ((r = esp_http_client_read(c, (char *)buf, sizeof(buf))) > 0) { while (total < SEG_MAX && (r = esp_http_client_read(c, (char *)s_seg + total, SEG_MAX - total)) > 0) {
if (s_sink && !s_sink(buf, r, first)) {
ok = false;
break;
}
first = false;
total += r; total += r;
if (!still_wanted()) {
ok = false;
break;
} }
} if (r < 0 || total == SEG_MAX) {
if (r < 0) {
ok = false; ok = false;
} }
esp_http_client_close(c); esp_http_client_close(c);
esp_http_client_cleanup(c); esp_http_client_cleanup(c);
int64_t us = esp_timer_get_time() - t0; int64_t us = esp_timer_get_time() - t0;
ESP_LOGI(TAG, "segment %lld: %u bytes in %.2f s (%.1f Mbit/s)%s", seq, (unsigned)total, us / 1e6, ESP_LOGD(TAG, "segment %lld: %u bytes in %.2f s (%.1f Mbit/s)%s", seq, (unsigned)total, us / 1e6,
us ? total * 8.0 / us : 0.0, ok ? "" : ", aborted"); us ? total * 8.0 / us : 0.0, ok ? "" : ", failed");
if (ok && s_sink) {
ok = s_sink(s_seg, total, true); // blocks at playback speed while the ring is full
}
return ok; return ok;
} }
@@ -300,7 +297,8 @@ esp_err_t hls_start(hls_sink_t sink)
s_sink = sink; s_sink = sink;
s_text = heap_caps_malloc(PLAYLIST_MAX, MALLOC_CAP_SPIRAM); s_text = heap_caps_malloc(PLAYLIST_MAX, MALLOC_CAP_SPIRAM);
s_pl = heap_caps_malloc(sizeof(media_pl_t), MALLOC_CAP_SPIRAM); s_pl = heap_caps_malloc(sizeof(media_pl_t), MALLOC_CAP_SPIRAM);
if (!s_text || !s_pl) { s_seg = heap_caps_malloc(SEG_MAX, MALLOC_CAP_SPIRAM);
if (!s_text || !s_pl || !s_seg) {
return ESP_ERR_NO_MEM; return ESP_ERR_NO_MEM;
} }
return xTaskCreate(hls_task, "hls", 10240, NULL, 4, NULL) == pdPASS ? ESP_OK : ESP_ERR_NO_MEM; return xTaskCreate(hls_task, "hls", 10240, NULL, 4, NULL) == pdPASS ? ESP_OK : ESP_ERR_NO_MEM;
+2 -1
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@@ -1,3 +1,4 @@
dependencies: dependencies:
# 2.6+ uses P4 assembly that needs chip rev >= 3.0; this board is rev 1.3 (see CLAUDE.md). # Newer versions use P4 assembly that needs chip rev >= 3.0; this board is rev 1.3 (see CLAUDE.md).
espressif/esp_audio_codec: "~2.5.0" espressif/esp_audio_codec: "~2.5.0"
espressif/esp_audio_effects: "~1.3.0"
+13 -17
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@@ -1,6 +1,5 @@
#include "player.h" #include "player.h"
#include <math.h>
#include <string.h> #include <string.h>
#include "aes67_cfg.h" #include "aes67_cfg.h"
@@ -72,24 +71,22 @@ void player_apply(const cJSON *source)
ESP_LOGI(TAG, "source: %s", SRC_NAME[src]); ESP_LOGI(TAG, "source: %s", SRC_NAME[src]);
} }
/* ----- Step 7.1: temporary test producer (440 Hz into the ring), replaced by the HLS player ----- */ // Source side (HLS task): write converted 48 kHz frames, waiting for space at playback speed.
// Gives up when the source is no longer HLS.
static void test_producer(void *arg) size_t player_write(const int32_t *frames, size_t n)
{ {
static int32_t chunk[480 * CHANNELS]; size_t done = 0;
uint32_t phase = 0; while (done < n) {
const double amp = pow(10.0, -18.0 / 20.0) * 2147483647.0; if (s_src != SRC_HLS) {
while (1) { return done;
if (s_src != SRC_HLS || audio_ring_space() < 480) {
vTaskDelay(pdMS_TO_TICKS(10));
continue;
} }
for (int i = 0; i < 480; i++, phase++) { size_t w = audio_ring_write(frames + done * CHANNELS, n - done);
int32_t v = (int32_t)(amp * sin(2.0 * M_PI * 440.0 * (phase % RATE) / RATE)); if (!w) {
chunk[i * 2] = chunk[i * 2 + 1] = v; vTaskDelay(pdMS_TO_TICKS(20));
} }
audio_ring_write(chunk, 480); done += w;
} }
return done;
} }
static void player_status(cJSON *st) static void player_status(cJSON *st)
@@ -113,12 +110,11 @@ esp_err_t player_init(void)
s_src = (src_t)-1; s_src = (src_t)-1;
player_apply(src); player_apply(src);
cJSON_Delete(src); cJSON_Delete(src);
xTaskCreate(test_producer, "test_prod", 3072, NULL, 5, NULL);
err = decoder_init(); err = decoder_init();
if (err != ESP_OK) { if (err != ESP_OK) {
return err; return err;
} }
err = hls_start(decoder_feed); // 7.3: decode and log; resampling + ring in 7.4 err = hls_start(decoder_feed);
if (err != ESP_OK) { if (err != ESP_OK) {
return err; return err;
} }
+5
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@@ -1,9 +1,14 @@
// Project player: picks the audio source for AES67 TX (test tone, silence, HLS, Spotify). // Project player: picks the audio source for AES67 TX (test tone, silence, HLS, Spotify).
#pragma once #pragma once
#include <stddef.h>
#include <stdint.h>
#include "cJSON.h" #include "cJSON.h"
#include "esp_err.h" #include "esp_err.h"
esp_err_t player_init(void); esp_err_t player_init(void);
// Apply the "source" config group (mode etc.). // Apply the "source" config group (mode etc.).
void player_apply(const cJSON *source); void player_apply(const cJSON *source);
// Source side: write 48 kHz stereo int32 frames, blocking while the ring is full.
size_t player_write(const int32_t *frames, size_t n);