5b0d7c9ba4
- main/audio_out: one converter instance per source (own rate converter state): s16 any rate/channels -> 48 kHz stereo int32 -> ring. - player_write(src, ...) only writes for the active source, drops the rest; player_src_t is public in player.h. - HLS decoder uses audio_out (no behaviour change). - Verified: HLS still sample exact (441344 -> 480375 frames per 10.008 s segment), buffer ~4 s, 0 underruns. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
186 lines
6.4 KiB
C
186 lines
6.4 KiB
C
#include "decoder.h"
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#include <stdlib.h>
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#include <string.h>
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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 "audio_out.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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static audio_conv_t *s_conv; // HLS: s16 -> 48 kHz -> ring
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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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static esp_audio_simple_dec_info_t s_info;
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static uint64_t s_seg_frames, s_seg_count, s_seg_es;
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// MPEG-TS demux state: packets reassembled across HTTP chunks, PAT -> PMT -> ADTS-AAC PID.
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static uint8_t s_pkt[188];
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static size_t s_pkt_len;
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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 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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// 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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s_seg_es += len;
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esp_audio_simple_dec_raw_t raw = { .buffer = (uint8_t *)data, .len = len };
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while (raw.len) {
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esp_audio_simple_dec_out_t out = { .buffer = s_pcm, .len = s_pcm_size };
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esp_audio_err_t err = esp_audio_simple_dec_process(s_dec, &raw, &out);
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if (err == ESP_AUDIO_ERR_BUFF_NOT_ENOUGH) {
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uint8_t *p = heap_caps_realloc(s_pcm, out.needed_size, MALLOC_CAP_SPIRAM);
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if (!p) {
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return;
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}
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s_pcm = p;
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s_pcm_size = out.needed_size;
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continue;
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}
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if (err != ESP_AUDIO_ERR_OK) {
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ESP_LOGW(TAG, "decode error %d, resetting decoder", err);
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esp_audio_simple_dec_reset(s_dec);
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return;
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}
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if (out.decoded_size) {
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if (!s_info.sample_rate) {
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esp_audio_simple_dec_get_info(s_dec, &s_info);
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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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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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s_seg_out += audio_conv_write(s_conv, (const int16_t *)out.buffer, n, s_info.channel,
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s_info.sample_rate);
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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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}
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}
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static void ts_packet(const uint8_t *p)
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{
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if (p[0] != 0x47) {
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return;
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}
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bool pusi = p[1] & 0x40;
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int pid = ((p[1] & 0x1f) << 8) | p[2];
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int afc = (p[3] >> 4) & 3;
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if (!(afc & 1)) {
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return; // no payload
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}
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size_t off = 4 + ((afc & 2) ? 1 + p[4] : 0);
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if (off >= 188) {
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return;
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}
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const uint8_t *pl = p + off;
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size_t len = 188 - off;
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if (pid == 0 && pusi) { // PAT: first program's PMT PID
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const uint8_t *sec = pl + 1 + pl[0];
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s_pmt_pid = ((sec[10] & 0x1f) << 8) | sec[11];
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} else if (pid == s_pmt_pid && pusi && s_audio_pid < 0) { // PMT: first ADTS-AAC stream
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const uint8_t *sec = pl + 1 + pl[0];
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int slen = ((sec[1] & 0x0f) << 8) | sec[2];
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int i = 12 + (((sec[10] & 0x0f) << 8) | sec[11]);
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while (i + 5 <= 3 + slen - 4 && sec + i + 5 <= p + 188) {
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int type = sec[i], epid = ((sec[i + 1] & 0x1f) << 8) | sec[i + 2];
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if (type == 0x0F) {
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s_audio_pid = epid;
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ESP_LOGI(TAG, "TS: PMT PID 0x%x, ADTS-AAC on PID 0x%x", s_pmt_pid, epid);
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break;
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}
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i += 5 + (((sec[i + 3] & 0x0f) << 8) | sec[i + 4]);
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}
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} else if (pid == s_audio_pid) {
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if (pusi) { // strip the PES header
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if (len < 9 || pl[0] || pl[1] || pl[2] != 1 || 9u + pl[8] > len) {
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return;
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}
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size_t h = 9 + pl[8];
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pl += h;
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len -= h;
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}
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decode_es(pl, len);
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}
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}
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// hls_sink_t: MPEG-TS bytes in any chunking.
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bool decoder_feed(const uint8_t *data, size_t len, bool segment_start)
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{
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if (segment_start) {
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report_segment();
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s_pkt_len = 0; // segments start on a packet boundary
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}
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while (len) {
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if (!s_pkt_len) {
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// resync on 0x47 if needed, then take whole packets straight from the input
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while (len && data[0] != 0x47) {
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data++;
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len--;
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}
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while (len >= 188 && data[0] == 0x47) {
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ts_packet(data);
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data += 188;
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len -= 188;
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}
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if (len && data[0] != 0x47) {
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continue;
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}
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}
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size_t n = 188 - s_pkt_len < len ? 188 - s_pkt_len : len;
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memcpy(s_pkt + s_pkt_len, data, n);
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s_pkt_len += n;
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data += n;
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len -= n;
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if (s_pkt_len == 188) {
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ts_packet(s_pkt);
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s_pkt_len = 0;
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}
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}
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return true;
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}
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esp_err_t decoder_init(void)
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{
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esp_audio_dec_register_default();
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esp_audio_simple_dec_register_default();
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// Own TS demux (the library's TS decoder lost ~8% of the frames); AAC with ADTS headers,
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// AAC-Plus on so HE-AAC v1/v2 variants work too.
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esp_aac_dec_cfg_t aac_cfg = ESP_AAC_DEC_CONFIG_DEFAULT();
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aac_cfg.aac_plus_enable = true;
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esp_audio_simple_dec_cfg_t cfg = {
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.dec_type = ESP_AUDIO_SIMPLE_DEC_TYPE_AAC, .dec_cfg = &aac_cfg, .cfg_size = sizeof(aac_cfg),
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};
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if (esp_audio_simple_dec_open(&cfg, &s_dec) != ESP_AUDIO_ERR_OK) {
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ESP_LOGE(TAG, "AAC decoder open failed");
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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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s_conv = audio_conv_create(PLAYER_SRC_HLS, "hls");
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return s_pcm && s_conv ? ESP_OK : ESP_ERR_NO_MEM;
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}
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