mirror of
https://github.com/sle118/squeezelite-esp32.git
synced 2025-12-07 20:17:04 +03:00
full i2s fix & loudness limited log
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@@ -170,7 +170,7 @@ static void raop_sink_start(nm_state_t state_id, int sub_state) {
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esp_netif_get_mac(netif, mac);
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cmd_handler_chain = raop_cbs.cmd;
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LOG_INFO( "Starting Airplay for ip %s with servicename %s", inet_ntoa(ipInfo.ip.addr), sink_name);
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LOG_INFO( "starting Airplay for ip %s with servicename %s", inet_ntoa(ipInfo.ip.addr), sink_name);
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raop = raop_create(ipInfo.ip.addr, sink_name, mac, 0, cmd_handler, raop_cbs.data);
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free(sink_name);
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}
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@@ -307,7 +307,8 @@ void cspotPlayer::runTask() {
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bell::MDNSService::registerService( blob->getDeviceName(), "_spotify-connect", "_tcp", "", serverPort,
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{ {"VERSION", "1.0"}, {"CPath", "/spotify_info"}, {"Stack", "SP"} });
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static int count = 0;
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CSPOT_LOG(info, "CSpot instance service name %s (id %s)", blob->getDeviceName().c_str(), blob->getDeviceId().c_str());
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// gone with the wind...
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while (1) {
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clientConnected.wait();
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@@ -163,15 +163,11 @@ static bool cmd_handler(cspot_event_t event, ...) {
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*/
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static void cspot_sink_start(nm_state_t state_id, int sub_state) {
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const char *hostname;
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uint8_t mac[6];
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cmd_handler_chain = cspot_cbs.cmd;
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network_get_hostname(&hostname);
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esp_netif_get_mac(network_get_active_interface(), mac);
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for (int i = 0; i < 6; i++) sprintf(deviceId + 2*i, "%02x", mac[i]);
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ESP_LOGI(TAG, "Starting Spotify (CSpot) servicename %s with id %s", hostname, deviceId);
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ESP_LOGI(TAG, "starting Spotify on host %s", hostname);
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int port;
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httpd_handle_t server = http_get_server(&port);
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@@ -64,13 +64,17 @@ static const float loudness_envelope_coefficients[EQ_BANDS][POLYNOME_COUNT] = {
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* calculate loudness gains
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*/
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static void calculate_loudness(void) {
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char trace[EQ_BANDS * 5 + 1];
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size_t n = 0;
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for (int i = 0; i < EQ_BANDS; i++) {
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for (int j = 0; j < POLYNOME_COUNT && equalizer.loudness != 0; j++) {
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equalizer.loudness_gain[i] +=
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loudness_envelope_coefficients[i][j] * pow(equalizer.volume, j);
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}
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equalizer.loudness_gain[i] *= equalizer.loudness / 2;
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n += sprintf(trace + n, "%.2g%c", equalizer.loudness_gain[i], i < EQ_BANDS ? ',' : '\0');
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}
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LOG_INFO("loudness %s", trace);
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}
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/****************************************************************************************
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@@ -211,7 +215,6 @@ void equalizer_process(uint8_t *buf, uint32_t bytes) {
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esp_equalizer_set_band_value(equalizer.handle, gain, i, 0);
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esp_equalizer_set_band_value(equalizer.handle, gain, i, 1);
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active |= gain != 0;
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LOG_INFO("EQUALIZER INDEX %u => gain:%.2f, loudness:%.2f,", i, equalizer.gain[i], equalizer.loudness_gain[i]);
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}
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// at the end do not activate equalizer if all gain are 0
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@@ -54,11 +54,22 @@ sure that using rate_delay would fix that
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#define FRAME_BLOCK MAX_SILENCE_FRAMES
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#define SPDIF_BLOCK 256
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// must have an integer ratio with FRAME_BLOCK (see spdif comment)
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#define DMA_SIZE 6144
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// FRAME_BLOCK must be a multiple of DMA_BUF_LEN no matter what
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#define DMA_BUF_LEN FRAME_BLOCK
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#define DMA_BUF_COUNT (DMA_SIZE / DMA_BUF_LEN)
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/* we produce FRAME_BLOCK (2048) per loop of the i2s thread so it's better if they fit
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* inside a set of DMA buffer nicely, i.e. DMA_BUF_FRAMES * DMA_BUF_COUNT is a multiple
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* of FRAME_BLOCK so that each DMA buffer is filled and we fully empty a FRAME_BLOCK at
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* each loop. Because one DMA buffer in esp32 is 4092 or below, when using 16 bits
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* samples and 2 channels, the best multiple is 512 (512*2*2=2048) and we use 6 of these.
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* In SPDIF, as we virtually use 32 bits per sample, the next proper multiple would
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* be 256 but such DMA buffers are too small and this causes stuttering. So we will use
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* non-multiples which means that at every loop one DMA buffer will be not fully filled.
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* At least, let's make sure it's not a too small amount of samples so 450*4*2=3600 fits
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* nicely in one DMA buffer and 2048/450 = 4 buffers + ~1/2 buffer which is acceptable.
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*/
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#define DMA_BUF_FRAMES 512
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#define DMA_BUF_COUNT 12
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#define DMA_BUF_FRAMES_SPDIF 450
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#define DMA_BUF_COUNT_SPDIF 7
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#define DECLARE_ALL_MIN_MAX \
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DECLARE_MIN_MAX(o); \
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@@ -262,6 +273,8 @@ void output_init_i2s(log_level level, char *device, unsigned output_buf_size, ch
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i2s_config.use_apll = true;
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#endif
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i2s_config.intr_alloc_flags = ESP_INTR_FLAG_LEVEL1; //Interrupt level 1
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i2s_config.dma_buf_len = DMA_BUF_FRAMES;
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i2s_config.dma_buf_count = DMA_BUF_COUNT;
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if (strcasestr(device, "spdif")) {
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spdif.enabled = true;
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@@ -278,14 +291,14 @@ void output_init_i2s(log_level level, char *device, unsigned output_buf_size, ch
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i2s_config.sample_rate = output.current_sample_rate * 2;
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i2s_config.bits_per_sample = 32;
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// Normally counted in frames, but 16 sample are transformed into 32 bits in spdif
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i2s_config.dma_buf_len = DMA_BUF_LEN / 2;
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i2s_config.dma_buf_count = DMA_BUF_COUNT * 2;
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i2s_config.dma_buf_len = DMA_BUF_FRAMES_SPDIF;
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i2s_config.dma_buf_count = DMA_BUF_COUNT_SPDIF;
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/*
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In DMA, we have room for (LEN * COUNT) frames of 32 bits samples that
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we push at sample_rate * 2. Each of these pseudo-frames is a single true
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audio frame. So the real depth in true frames is (LEN * COUNT / 2)
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*/
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dma_buf_frames = DMA_BUF_COUNT * DMA_BUF_LEN / 2;
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dma_buf_frames = i2s_config.dma_buf_len * i2s_config.dma_buf_count / 2;
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// silence DAC output if sharing the same ws/bck
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if (i2s_dac_pin.ws_io_num == i2s_spdif_pin.ws_io_num && i2s_dac_pin.bck_io_num == i2s_spdif_pin.bck_io_num) silent_do = i2s_dac_pin.data_out_num;
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@@ -297,9 +310,9 @@ void output_init_i2s(log_level level, char *device, unsigned output_buf_size, ch
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i2s_config.sample_rate = output.current_sample_rate;
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i2s_config.bits_per_sample = BYTES_PER_FRAME * 8 / 2;
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// Counted in frames (but i2s allocates a buffer <= 4092 bytes)
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i2s_config.dma_buf_len = DMA_BUF_LEN;
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i2s_config.dma_buf_len = DMA_BUF_FRAMES;
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i2s_config.dma_buf_count = DMA_BUF_COUNT;
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dma_buf_frames = DMA_BUF_COUNT * DMA_BUF_LEN;
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dma_buf_frames = i2s_config.dma_buf_len * i2s_config.dma_buf_count;
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// silence SPDIF output
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silent_do = i2s_spdif_pin.data_out_num;
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