48 kHz only: drop 96 kHz from the AES67 output
All sources (player, test signals) are 48 kHz; at 96 kHz Spotify and HLS played at the wrong speed. Validation, the page and the docs allow 48000 only, and a 96000 stored before is reset to 48000 at boot (stored values aren't re-validated, and the SDP would still have said 96 kHz). Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
This commit is contained in:
@@ -16,6 +16,7 @@
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#include "lwip/inet.h"
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#include "lwip/sockets.h"
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#define AES67_RATE 48000 // the only sample rate (all sources are 48 kHz)
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#define MAX_FRAMES 192 // 4 ms at 48 kHz
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#define MAX_CH 2
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#define RTP_HDR 12
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@@ -56,7 +57,7 @@ static bool aes67_validate(const cJSON *g, char *err, size_t n)
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{
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static const char *const discovery[] = { "manual", "sap", NULL };
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static const char *const encodings[] = { "L16", "L24", NULL };
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static const double rates[] = { 48000, 96000 };
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static const double rates[] = { AES67_RATE }; // the sources (player, test signals) are 48 kHz
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static const double ptimes[] = { 0.125, 0.25, 0.333, 1, 4 };
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bool ok = cfg_check_str(g, "name", 1, 63, err, n) &&
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@@ -67,7 +68,7 @@ static bool aes67_validate(const cJSON *g, char *err, size_t n)
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cfg_check_int(g, "dscp", 0, 63, err, n) &&
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cfg_check_int(g, "channels", 1, 2, err, n) &&
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cfg_check_enum(g, "encoding", encodings, err, n) &&
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cfg_check_num_in(g, "rate", rates, 2, err, n) &&
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cfg_check_num_in(g, "rate", rates, 1, err, n) &&
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cfg_check_num_in(g, "ptime", ptimes, 5, err, n) &&
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cfg_check_int(g, "pt", 96, 127, err, n) &&
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cfg_check_int(g, "ssrc", 0, 4294967295.0, err, n) &&
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@@ -168,8 +169,6 @@ static int64_t samples_to_ns(int64_t s, int rate)
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return (s / rate) * 1000000000LL + ((s % rate) * 1000000000LL + rate - 1) / rate;
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}
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// 1 kHz tone, phase from the PTP sample index, so every sender's tone lines up.
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// One period is rate / 1000 samples (48 at 48 kHz, 96 at 96 kHz): precomputed table.
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// Pink noise: xorshift32 white noise through Paul Kellet's refined pink filter (within 0.05 dB
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// above 9 Hz). The scale gives -18 dBFS RMS (measured over 10 s; peaks about -6 dBFS).
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#define PINK_SCALE 0.0714f
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@@ -209,9 +208,11 @@ static void to_mono(int32_t *buf, int frames, bool sum)
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}
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}
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// 1 kHz tone, phase from the PTP sample index, so every sender's tone lines up.
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// One period is rate / 1000 samples (48 at 48 kHz): precomputed table.
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static void tone(int32_t *buf, int frames, int channels, int rate, int64_t s0)
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{
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static int32_t table[96000 / TONE_HZ];
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static int32_t table[AES67_RATE / TONE_HZ];
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static int table_rate;
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int period = rate / TONE_HZ;
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if (table_rate != rate) {
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@@ -361,5 +362,13 @@ esp_err_t aes67_tx_init(void)
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if (err != ESP_OK) {
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return err;
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}
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// Stored before 96 kHz was dropped: stored values aren't re-validated, so fix it here (SDP too).
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cJSON *a = cfg_get("aes67");
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double rate = cJSON_GetObjectItemCaseSensitive(a, "rate")->valuedouble;
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cJSON_Delete(a);
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if (rate != AES67_RATE) {
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ESP_LOGW(TAG, "stored sample rate %.0f not supported: using %d", rate, AES67_RATE);
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cfg_set_number("aes67", "rate", AES67_RATE, false);
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}
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return status_register(tx_status);
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}
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@@ -19,7 +19,7 @@ Reference devices for UI and defaults: Riedel Bolero (PTP status), Riedel Direct
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## Config schema (core groups)
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- ptp: {mode: multicast|hybrid, role: slave|auto|master, domain, priority1, priority2, log_sync, log_announce, announce_timeout, log_delay_req, dscp, hw_ts}
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- aes67: {name, enabled, discovery: manual|sap, mcast, port, ttl, dscp, channels, mono_sum, encoding: L16|L24, rate, ptime, pt, ssrc, clk_offset, session_id, session_ver}
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- aes67: {name, enabled, discovery: manual|sap, mcast, port, ttl, dscp, channels, mono_sum, encoding: L16|L24, rate: 48000 only (all sources are 48 kHz), ptime, pt, ssrc, clk_offset, session_id, session_ver}
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- net: {hostname, vlan, web_on: both|inet|aes67, dhcp, ip, mask, gw, dns} (AES67 / untagged interface)
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- inet: {vlan_id, pcp, dhcp, ip, mask, gw, dns} (internet / tagged, only if net.vlan)
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- log: {syslog, host, port, level: error|warn|info|debug, facility, format: rfc5424|rfc3164}
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+1
-2
@@ -105,7 +105,7 @@ The secret is write-only: after saving, the field shows empty. Leave it empty to
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| Channels | 2 | 1 or 2. |
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| Mono sum | off | Sends (left + right) / 2 as a 1-channel stream. Forces *Channels* to 1; it can't clip. |
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| Bit depth | L24 | L24 or L16. |
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| Sample rate | 48000 | **Use 48000 with Spotify and HLS** (see [Limitations](#limitations)). |
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| Sample rate | 48000 | Fixed; all sources are 48 kHz. |
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| Packet time | 1 ms | 0.125, 0.25, 0.333, 1 or 4 ms. 1 ms is the AES67 default and works with nearly all receivers. |
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| Payload type / SSRC / time stamp offset | 96 / 0 / 0 | Only change these if a receiver needs it. |
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@@ -220,7 +220,6 @@ The full device API (status, config, SDP, firmware) is described in `docs/aes67-
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## Limitations
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These are known limitations in the current firmware:
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- **96 kHz:** only the test tone and pink noise work correctly at 96 kHz. Spotify and HLS are 48 kHz sources and must be streamed at **48000**. At 96 kHz they play at the wrong speed.
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- **Autoplay stream on boot:** the checkbox has no effect yet. HLS always starts on boot.
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- **VLAN split** (separate VLANs for AES67 and internet) isn't available yet.
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- **No password** on the web page or API.
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+1
-1
@@ -145,7 +145,7 @@ e.g. curl -X POST http://p4-aes67.local/api/player/next
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<small>(Ch 1 + Ch 2) / 2 → 1-channel stream</small></label>
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<label><span>Bit depth</span><select name="aes67.encoding"><option>L24</option><option>L16</option></select>
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<small>Default L24</small></label>
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<label><span>Sample rate</span><select name="aes67.rate"><option>48000</option><option>96000</option></select></label>
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<label><span>Sample rate</span><select name="aes67.rate"><option>48000</option></select></label>
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<label><span>Packet time</span><select name="aes67.ptime">
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<option value="0.125">0.125 ms</option><option value="0.25">0.250 ms</option><option value="0.333">0.333 ms</option>
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<option value="1">1.000 ms</option><option value="4">4.000 ms</option></select>
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