Merge branch 'main' into opus

2 years ago · e1181fbe59
parent c3f40a8cc3 592f231627
commit e1181fbe59
43 changed files with 1576 additions and 6125 deletions
--- a/.gitignore
+++ b/.gitignore
@ -4,6 +4,7 @@
 .cache/
 .clangd
 .helix/
 build/
 build.clang/
 sdkconfig
--- a/lib/stb_vorbis/CMakeLists.txt
+++ b/lib/stb_vorbis/CMakeLists.txt
@ -1,8 +0,0 @@
 # Copyright 2023 jacqueline <me@jacqueline.id.au>
 #
 # SPDX-License-Identifier: GPL-3.0-only
 idf_component_register(
  SRCS "stb_vorbis.c"
  INCLUDE_DIRS "include"
 )
--- a/lib/stb_vorbis/include/stb_vorbis.h
+++ b/lib/stb_vorbis/include/stb_vorbis.h
@ -1,418 +0,0 @@
 // Ogg Vorbis audio decoder - v1.22 - public domain
 // http://nothings.org/stb_vorbis/
 //
 // Original version written by Sean Barrett in 2007.
 //
 // Originally sponsored by RAD Game Tools. Seeking implementation
 // sponsored by Phillip Bennefall, Marc Andersen, Aaron Baker,
 // Elias Software, Aras Pranckevicius, and Sean Barrett.
 //
 // LICENSE
 //
 //   See end of file for license information.
 //
 // Limitations:
 //
 //   - floor 0 not supported (used in old ogg vorbis files pre-2004)
 //   - lossless sample-truncation at beginning ignored
 //   - cannot concatenate multiple vorbis streams
 //   - sample positions are 32-bit, limiting seekable 192Khz
 //       files to around 6 hours (Ogg supports 64-bit)
 //
 // Feature contributors:
 //    Dougall Johnson (sample-exact seeking)
 //
 // Bugfix/warning contributors:
 //    Terje Mathisen     Niklas Frykholm     Andy Hill
 //    Casey Muratori     John Bolton         Gargaj
 //    Laurent Gomila     Marc LeBlanc        Ronny Chevalier
 //    Bernhard Wodo      Evan Balster        github:alxprd
 //    Tom Beaumont       Ingo Leitgeb        Nicolas Guillemot
 //    Phillip Bennefall  Rohit               Thiago Goulart
 //    github:manxorist   Saga Musix          github:infatum
 //    Timur Gagiev       Maxwell Koo         Peter Waller
 //    github:audinowho   Dougall Johnson     David Reid
 //    github:Clownacy    Pedro J. Estebanez  Remi Verschelde
 //    AnthoFoxo          github:morlat       Gabriel Ravier
 //
 // Partial history:
 //    1.22    - 2021-07-11 - various small fixes
 //    1.21    - 2021-07-02 - fix bug for files with no comments
 //    1.20    - 2020-07-11 - several small fixes
 //    1.19    - 2020-02-05 - warnings
 //    1.18    - 2020-02-02 - fix seek bugs; parse header comments; misc warnings etc.
 //    1.17    - 2019-07-08 - fix CVE-2019-13217..CVE-2019-13223 (by ForAllSecure)
 //    1.16    - 2019-03-04 - fix warnings
 //    1.15    - 2019-02-07 - explicit failure if Ogg Skeleton data is found
 //    1.14    - 2018-02-11 - delete bogus dealloca usage
 //    1.13    - 2018-01-29 - fix truncation of last frame (hopefully)
 //    1.12    - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
 //    1.11    - 2017-07-23 - fix MinGW compilation
 //    1.10    - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
 //    1.09    - 2016-04-04 - back out 'truncation of last frame' fix from previous version
 //    1.08    - 2016-04-02 - warnings; setup memory leaks; truncation of last frame
 //    1.07    - 2015-01-16 - fixes for crashes on invalid files; warning fixes; const
 //    1.06    - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
 //                           some crash fixes when out of memory or with corrupt files
 //                           fix some inappropriately signed shifts
 //    1.05    - 2015-04-19 - don't define __forceinline if it's redundant
 //    1.04    - 2014-08-27 - fix missing const-correct case in API
 //    1.03    - 2014-08-07 - warning fixes
 //    1.02    - 2014-07-09 - declare qsort comparison as explicitly _cdecl in Windows
 //    1.01    - 2014-06-18 - fix stb_vorbis_get_samples_float (interleaved was correct)
 //    1.0     - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in >2-channel;
 //                           (API change) report sample rate for decode-full-file funcs
 //
 // See end of file for full version history.
 //////////////////////////////////////////////////////////////////////////////
 //
 //  HEADER BEGINS HERE
 //
 #ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
 #define STB_VORBIS_INCLUDE_STB_VORBIS_H
 #if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
 #define STB_VORBIS_NO_STDIO 1
 #endif
 #ifndef STB_VORBIS_NO_STDIO
 #include <stdio.h>
 #endif
 #ifdef __cplusplus
 extern "C" {
 #endif
 ///////////   THREAD SAFETY
 // Individual stb_vorbis* handles are not thread-safe; you cannot decode from
 // them from multiple threads at the same time. However, you can have multiple
 // stb_vorbis* handles and decode from them independently in multiple thrads.
 ///////////   MEMORY ALLOCATION
 // normally stb_vorbis uses malloc() to allocate memory at startup,
 // and alloca() to allocate temporary memory during a frame on the
 // stack. (Memory consumption will depend on the amount of setup
 // data in the file and how you set the compile flags for speed
 // vs. size. In my test files the maximal-size usage is ~150KB.)
 //
 // You can modify the wrapper functions in the source (setup_malloc,
 // setup_temp_malloc, temp_malloc) to change this behavior, or you
 // can use a simpler allocation model: you pass in a buffer from
 // which stb_vorbis will allocate _all_ its memory (including the
 // temp memory). "open" may fail with a VORBIS_outofmem if you
 // do not pass in enough data; there is no way to determine how
 // much you do need except to succeed (at which point you can
 // query get_info to find the exact amount required. yes I know
 // this is lame).
 //
 // If you pass in a non-NULL buffer of the type below, allocation
 // will occur from it as described above. Otherwise just pass NULL
 // to use malloc()/alloca()
 typedef struct
 {
   char *alloc_buffer;
   int   alloc_buffer_length_in_bytes;
 } stb_vorbis_alloc;
 ///////////   FUNCTIONS USEABLE WITH ALL INPUT MODES
 typedef struct stb_vorbis stb_vorbis;
 typedef struct
 {
   unsigned int sample_rate;
   int channels;
   unsigned int setup_memory_required;
   unsigned int setup_temp_memory_required;
   unsigned int temp_memory_required;
   int max_frame_size;
 } stb_vorbis_info;
 typedef struct
 {
   char *vendor;
   int comment_list_length;
   char **comment_list;
 } stb_vorbis_comment;
 // get general information about the file
 extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f);
 // get ogg comments
 extern stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f);
 // get the last error detected (clears it, too)
 extern int stb_vorbis_get_error(stb_vorbis *f);
 // close an ogg vorbis file and free all memory in use
 extern void stb_vorbis_close(stb_vorbis *f);
 // this function returns the offset (in samples) from the beginning of the
 // file that will be returned by the next decode, if it is known, or -1
 // otherwise. after a flush_pushdata() call, this may take a while before
 // it becomes valid again.
 // NOT WORKING YET after a seek with PULLDATA API
 extern int stb_vorbis_get_sample_offset(stb_vorbis *f);
 // returns the current seek point within the file, or offset from the beginning
 // of the memory buffer. In pushdata mode it returns 0.
 extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f);
 ///////////   PUSHDATA API
 #ifndef STB_VORBIS_NO_PUSHDATA_API
 // this API allows you to get blocks of data from any source and hand
 // them to stb_vorbis. you have to buffer them; stb_vorbis will tell
 // you how much it used, and you have to give it the rest next time;
 // and stb_vorbis may not have enough data to work with and you will
 // need to give it the same data again PLUS more. Note that the Vorbis
 // specification does not bound the size of an individual frame.
 extern stb_vorbis *stb_vorbis_open_pushdata(
         const unsigned char * datablock, int datablock_length_in_bytes,
         int *datablock_memory_consumed_in_bytes,
         int *error,
         const stb_vorbis_alloc *alloc_buffer);
 // create a vorbis decoder by passing in the initial data block containing
 //    the ogg&vorbis headers (you don't need to do parse them, just provide
 //    the first N bytes of the file--you're told if it's not enough, see below)
 // on success, returns an stb_vorbis *, does not set error, returns the amount of
 //    data parsed/consumed on this call in *datablock_memory_consumed_in_bytes;
 // on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed
 // if returns NULL and *error is VORBIS_need_more_data, then the input block was
 //       incomplete and you need to pass in a larger block from the start of the file
 extern int stb_vorbis_decode_frame_pushdata(
         stb_vorbis *f,
         const unsigned char *datablock, int datablock_length_in_bytes,
         int *channels,             // place to write number of float * buffers
         float ***output,           // place to write float ** array of float * buffers
         int *samples               // place to write number of output samples
     );
 // decode a frame of audio sample data if possible from the passed-in data block
 //
 // return value: number of bytes we used from datablock
 //
 // possible cases:
 //     0 bytes used, 0 samples output (need more data)
 //     N bytes used, 0 samples output (resynching the stream, keep going)
 //     N bytes used, M samples output (one frame of data)
 // note that after opening a file, you will ALWAYS get one N-bytes,0-sample
 // frame, because Vorbis always "discards" the first frame.
 //
 // Note that on resynch, stb_vorbis will rarely consume all of the buffer,
 // instead only datablock_length_in_bytes-3 or less. This is because it wants
 // to avoid missing parts of a page header if they cross a datablock boundary,
 // without writing state-machiney code to record a partial detection.
 //
 // The number of channels returned are stored in *channels (which can be
 // NULL--it is always the same as the number of channels reported by
 // get_info). *output will contain an array of float* buffers, one per
 // channel. In other words, (*output)[0][0] contains the first sample from
 // the first channel, and (*output)[1][0] contains the first sample from
 // the second channel.
 //
 // *output points into stb_vorbis's internal output buffer storage; these
 // buffers are owned by stb_vorbis and application code should not free
 // them or modify their contents. They are transient and will be overwritten
 // once you ask for more data to get decoded, so be sure to grab any data
 // you need before then.
 extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
 // inform stb_vorbis that your next datablock will not be contiguous with
 // previous ones (e.g. you've seeked in the data); future attempts to decode
 // frames will cause stb_vorbis to resynchronize (as noted above), and
 // once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it
 // will begin decoding the _next_ frame.
 //
 // if you want to seek using pushdata, you need to seek in your file, then
 // call stb_vorbis_flush_pushdata(), then start calling decoding, then once
 // decoding is returning you data, call stb_vorbis_get_sample_offset, and
 // if you don't like the result, seek your file again and repeat.
 #endif
 //////////   PULLING INPUT API
 #ifndef STB_VORBIS_NO_PULLDATA_API
 // This API assumes stb_vorbis is allowed to pull data from a source--
 // either a block of memory containing the _entire_ vorbis stream, or a
 // FILE * that you or it create, or possibly some other reading mechanism
 // if you go modify the source to replace the FILE * case with some kind
 // of callback to your code. (But if you don't support seeking, you may
 // just want to go ahead and use pushdata.)
 #if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
 extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output);
 #endif
 #if !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
 extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output);
 #endif
 // decode an entire file and output the data interleaved into a malloc()ed
 // buffer stored in *output. The return value is the number of samples
 // decoded, or -1 if the file could not be opened or was not an ogg vorbis file.
 // When you're done with it, just free() the pointer returned in *output.
 extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len,
                                  int *error, const stb_vorbis_alloc *alloc_buffer);
 // create an ogg vorbis decoder from an ogg vorbis stream in memory (note
 // this must be the entire stream!). on failure, returns NULL and sets *error
 #ifndef STB_VORBIS_NO_STDIO
 extern stb_vorbis * stb_vorbis_open_filename(const char *filename,
                                  int *error, const stb_vorbis_alloc *alloc_buffer);
 // create an ogg vorbis decoder from a filename via fopen(). on failure,
 // returns NULL and sets *error (possibly to VORBIS_file_open_failure).
 extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close,
                                  int *error, const stb_vorbis_alloc *alloc_buffer);
 // create an ogg vorbis decoder from an open FILE *, looking for a stream at
 // the _current_ seek point (ftell). on failure, returns NULL and sets *error.
 // note that stb_vorbis must "own" this stream; if you seek it in between
 // calls to stb_vorbis, it will become confused. Moreover, if you attempt to
 // perform stb_vorbis_seek_*() operations on this file, it will assume it
 // owns the _entire_ rest of the file after the start point. Use the next
 // function, stb_vorbis_open_file_section(), to limit it.
 extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close,
                int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len);
 // create an ogg vorbis decoder from an open FILE *, looking for a stream at
 // the _current_ seek point (ftell); the stream will be of length 'len' bytes.
 // on failure, returns NULL and sets *error. note that stb_vorbis must "own"
 // this stream; if you seek it in between calls to stb_vorbis, it will become
 // confused.
 #endif
 extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
 extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
 // these functions seek in the Vorbis file to (approximately) 'sample_number'.
 // after calling seek_frame(), the next call to get_frame_*() will include
 // the specified sample. after calling stb_vorbis_seek(), the next call to
 // stb_vorbis_get_samples_* will start with the specified sample. If you
 // do not need to seek to EXACTLY the target sample when using get_samples_*,
 // you can also use seek_frame().
 extern int stb_vorbis_seek_start(stb_vorbis *f);
 // this function is equivalent to stb_vorbis_seek(f,0)
 extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f);
 extern float        stb_vorbis_stream_length_in_seconds(stb_vorbis *f);
 // these functions return the total length of the vorbis stream
 extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output);
 // decode the next frame and return the number of samples. the number of
 // channels returned are stored in *channels (which can be NULL--it is always
 // the same as the number of channels reported by get_info). *output will
 // contain an array of float* buffers, one per channel. These outputs will
 // be overwritten on the next call to stb_vorbis_get_frame_*.
 //
 // You generally should not intermix calls to stb_vorbis_get_frame_*()
 // and stb_vorbis_get_samples_*(), since the latter calls the former.
 #ifndef STB_VORBIS_NO_INTEGER_CONVERSION
 extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts);
 extern int stb_vorbis_get_frame_short            (stb_vorbis *f, int num_c, short **buffer, int num_samples);
 #endif
 // decode the next frame and return the number of *samples* per channel.
 // Note that for interleaved data, you pass in the number of shorts (the
 // size of your array), but the return value is the number of samples per
 // channel, not the total number of samples.
 //
 // The data is coerced to the number of channels you request according to the
 // channel coercion rules (see below). You must pass in the size of your
 // buffer(s) so that stb_vorbis will not overwrite the end of the buffer.
 // The maximum buffer size needed can be gotten from get_info(); however,
 // the Vorbis I specification implies an absolute maximum of 4096 samples
 // per channel.
 // Channel coercion rules:
 //    Let M be the number of channels requested, and N the number of channels present,
 //    and Cn be the nth channel; let stereo L be the sum of all L and center channels,
 //    and stereo R be the sum of all R and center channels (channel assignment from the
 //    vorbis spec).
 //        M    N       output
 //        1    k      sum(Ck) for all k
 //        2    *      stereo L, stereo R
 //        k    l      k > l, the first l channels, then 0s
 //        k    l      k <= l, the first k channels
 //    Note that this is not _good_ surround etc. mixing at all! It's just so
 //    you get something useful.
 extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats);
 extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples);
 // gets num_samples samples, not necessarily on a frame boundary--this requires
 // buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES.
 // Returns the number of samples stored per channel; it may be less than requested
 // at the end of the file. If there are no more samples in the file, returns 0.
 #ifndef STB_VORBIS_NO_INTEGER_CONVERSION
 extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts);
 extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples);
 #endif
 // gets num_samples samples, not necessarily on a frame boundary--this requires
 // buffering so you have to supply the buffers. Applies the coercion rules above
 // to produce 'channels' channels. Returns the number of samples stored per channel;
 // it may be less than requested at the end of the file. If there are no more
 // samples in the file, returns 0.
 #endif
 ////////   ERROR CODES
 enum STBVorbisError
 {
   VORBIS__no_error,
   VORBIS_need_more_data=1,             // not a real error
   VORBIS_invalid_api_mixing,           // can't mix API modes
   VORBIS_outofmem,                     // not enough memory
   VORBIS_feature_not_supported,        // uses floor 0
   VORBIS_too_many_channels,            // STB_VORBIS_MAX_CHANNELS is too small
   VORBIS_file_open_failure,            // fopen() failed
   VORBIS_seek_without_length,          // can't seek in unknown-length file
   VORBIS_unexpected_eof=10,            // file is truncated?
   VORBIS_seek_invalid,                 // seek past EOF
   // decoding errors (corrupt/invalid stream) -- you probably
   // don't care about the exact details of these
   // vorbis errors:
   VORBIS_invalid_setup=20,
   VORBIS_invalid_stream,
   // ogg errors:
   VORBIS_missing_capture_pattern=30,
   VORBIS_invalid_stream_structure_version,
   VORBIS_continued_packet_flag_invalid,
   VORBIS_incorrect_stream_serial_number,
   VORBIS_invalid_first_page,
   VORBIS_bad_packet_type,
   VORBIS_cant_find_last_page,
   VORBIS_seek_failed,
   VORBIS_ogg_skeleton_not_supported
 };
 #ifdef __cplusplus
 }
 #endif
 #endif // STB_VORBIS_INCLUDE_STB_VORBIS_H
 //
 //  HEADER ENDS HERE
 //
 //////////////////////////////////////////////////////////////////////////////
--- a/lib/stb_vorbis/stb_vorbis.c
+++ b/lib/stb_vorbis/stb_vorbis.c
--- a/sdkconfig.common
+++ b/sdkconfig.common
@ -1,16 +1,25 @@
-CONFIG_BOOTLOADER_LOG_LEVEL_WARN=y
+CONFIG_BOOTLOADER_LOG_LEVEL_NONE=y
 CONFIG_ESPTOOLPY_FLASHSIZE_16MB=y
 CONFIG_ESPTOOLPY_HEADER_FLASHSIZE_UPDATE=y
 CONFIG_PARTITION_TABLE_CUSTOM=y
 CONFIG_COMPILER_OPTIMIZATION_ASSERTIONS_DISABLE=y
 CONFIG_COMPILER_STACK_CHECK_MODE_NORM=y
 CONFIG_BT_ENABLED=y
 CONFIG_BT_CLASSIC_ENABLED=y
 CONFIG_BT_A2DP_ENABLE=y
 # CONFIG_BT_BLE_ENABLED is not set
 # CONFIG_BT_MULTI_CONNECTION_ENBALE is not set
 CONFIG_BT_ALLOCATION_FROM_SPIRAM_FIRST=y
 CONFIG_BTDM_CTRL_MODE_BR_EDR_ONLY=y
 # CONFIG_TWAI_ERRATA_FIX_BUS_OFF_REC is not set
 # CONFIG_TWAI_ERRATA_FIX_TX_INTR_LOST is not set
 # CONFIG_TWAI_ERRATA_FIX_RX_FRAME_INVALID is not set
 # CONFIG_TWAI_ERRATA_FIX_RX_FIFO_CORRUPT is not set
 CONFIG_I2S_ISR_IRAM_SAFE=y
 # CONFIG_ETH_USE_ESP32_EMAC is not set
 # CONFIG_ETH_USE_SPI_ETHERNET is not set
 CONFIG_ESP_HTTP_CLIENT_ENABLE_DIGEST_AUTH=y
 CONFIG_ESP32_REV_MIN_3=y
 CONFIG_ESP_PHY_REDUCE_TX_POWER=y
 CONFIG_SPIRAM=y
 CONFIG_SPIRAM_TRY_ALLOCATE_WIFI_LWIP=y
 CONFIG_SPIRAM_ALLOW_BSS_SEG_EXTERNAL_MEMORY=y
@ -22,14 +31,20 @@ CONFIG_ESP_MAIN_TASK_STACK_SIZE=12000
 CONFIG_ESP_INT_WDT_TIMEOUT_MS=1000
 CONFIG_ESP_TASK_WDT_TIMEOUT_S=10
 CONFIG_ESP_IPC_TASK_STACK_SIZE=1536
-CONFIG_ESP32_WIFI_STATIC_RX_BUFFER_NUM=10
+CONFIG_ESP_WIFI_STATIC_RX_BUFFER_NUM=10
-CONFIG_ESP32_WIFI_RX_BA_WIN=6
+CONFIG_ESP_WIFI_RX_BA_WIN=6
 CONFIG_ESP_WIFI_IRAM_OPT=y
 CONFIG_ESP_WIFI_RX_IRAM_OPT=y
 CONFIG_FATFS_VOLUME_COUNT=1
 CONFIG_FATFS_LFN_HEAP=y
 CONFIG_FATFS_API_ENCODING_UTF_8=y
 CONFIG_FATFS_USE_FASTSEEK=y
 CONFIG_FREERTOS_VTASKLIST_INCLUDE_COREID=y
 CONFIG_FREERTOS_GENERATE_RUN_TIME_STATS=y
 CONFIG_FREERTOS_WATCHPOINT_END_OF_STACK=y
 CONFIG_HEAP_POISONING_COMPREHENSIVE=y
 CONFIG_HEAP_ABORT_WHEN_ALLOCATION_FAILS=y
 CONFIG_WIFI_PROV_BLE_FORCE_ENCRYPTION=y
 CONFIG_LV_COLOR_16_SWAP=y
 CONFIG_LV_COLOR_MIX_ROUND_OFS=0
 CONFIG_LV_MEM_CUSTOM=y
--- a/src/app_console/app_console.cpp
+++ b/src/app_console/app_console.cpp
@ -7,16 +7,19 @@
 #include "app_console.hpp"
 #include <dirent.h>
 #include <stdint.h>
 #include <algorithm>
 #include <cstdint>
 #include <cstdio>
 #include <cstdlib>
 #include <iomanip>
 #include <iostream>
 #include <ostream>
 #include <sstream>
 #include <string>
 #include "FreeRTOSConfig.h"
 #include "audio_events.hpp"
 #include "audio_fsm.hpp"
 #include "database.hpp"
@ -25,6 +28,8 @@
 #include "esp_log.h"
 #include "event_queue.hpp"
 #include "ff.h"
 #include "freertos/FreeRTOSConfig_arch.h"
 #include "freertos/projdefs.h"
 #include "index.hpp"
 #include "track.hpp"
@ -324,6 +329,116 @@ void RegisterDbDump() {
  esp_console_cmd_register(&cmd);
 }
 int CmdTaskStats(int argc, char** argv) {
  if (!configUSE_TRACE_FACILITY) {
    std::cout << "configUSE_TRACE_FACILITY must be enabled" << std::endl;
    std::cout << "also consider configTASKLIST_USE_COREID" << std::endl;
    return 1;
  }
  static const std::string usage = "usage: task_stats";
  if (argc != 1) {
    std::cout << usage << std::endl;
    return 1;
  }
  // Pad the number of tasks so that uxTaskGetSystemState still returns info if
  // new tasks are started during measurement.
  size_t num_tasks = uxTaskGetNumberOfTasks() + 4;
  TaskStatus_t* start_status = new TaskStatus_t[num_tasks];
  TaskStatus_t* end_status = new TaskStatus_t[num_tasks];
  uint32_t start_elapsed_ticks = 0;
  uint32_t end_elapsed_ticks = 0;
  size_t start_num_tasks =
      uxTaskGetSystemState(start_status, num_tasks, &start_elapsed_ticks);
  vTaskDelay(pdMS_TO_TICKS(2500));
  size_t end_num_tasks =
      uxTaskGetSystemState(end_status, num_tasks, &end_elapsed_ticks);
  std::vector<std::pair<uint32_t, std::string>> info_strings;
  for (int i = 0; i < start_num_tasks; i++) {
    int k = -1;
    for (int j = 0; j < end_num_tasks; j++) {
      if (start_status[i].xHandle == end_status[j].xHandle) {
        k = j;
        break;
      }
    }
    if (k >= 0) {
      uint32_t run_time =
          end_status[k].ulRunTimeCounter - start_status[i].ulRunTimeCounter;
      float time_percent =
          static_cast<float>(run_time) /
          static_cast<float>(end_elapsed_ticks - start_elapsed_ticks);
      auto depth = uxTaskGetStackHighWaterMark2(start_status[i].xHandle);
      float depth_kib = static_cast<float>(depth) / 1024.0f;
      std::ostringstream str;
      str << start_status[i].pcTaskName;
      if (str.str().size() < 8) {
        str << "\t\t";
      } else {
        str << "\t";
      }
      if (configTASKLIST_INCLUDE_COREID) {
        if (start_status[i].xCoreID == tskNO_AFFINITY) {
          str << "any\t";
        } else {
          str << start_status[i].xCoreID << "\t";
        }
      }
      str << std::fixed << std::setprecision(1) << depth_kib;
      str << " KiB";
      if (depth_kib >= 10) {
        str << "\t";
      } else {
        str << "\t\t";
      }
      str << std::fixed << std::setprecision(1) << time_percent * 100;
      str << "%";
      info_strings.push_back({run_time, str.str()});
    }
  }
  std::sort(info_strings.begin(), info_strings.end(),
            [](const auto& first, const auto& second) {
              return first.first >= second.first;
            });
  std::cout << "name\t\t";
  if (configTASKLIST_INCLUDE_COREID) {
    std::cout << "core\t";
  }
  std::cout << "free stack\trun time" << std::endl;
  for (const auto& i : info_strings) {
    std::cout << i.second << std::endl;
  }
  delete[] start_status;
  delete[] end_status;
  return 0;
 }
 void RegisterTaskStates() {
  esp_console_cmd_t cmd{.command = "task_stats",
                        .help = "prints performance info for all tasks",
                        .hint = NULL,
                        .func = &CmdTaskStats,
                        .argtable = NULL};
  esp_console_cmd_register(&cmd);
 }
 auto AppConsole::RegisterExtraComponents() -> void {
  RegisterListDir();
  RegisterPlayFile();
@ -336,6 +451,7 @@ auto AppConsole::RegisterExtraComponents() -> void {
  RegisterDbTracks();
  RegisterDbIndex();
  RegisterDbDump();
  RegisterTaskStates();
 }
 }  // namespace console
--- a/src/audio/CMakeLists.txt
+++ b/src/audio/CMakeLists.txt
@ -5,7 +5,7 @@
 idf_component_register(
  SRCS "audio_task.cpp" "chunk.cpp" "fatfs_audio_input.cpp"
  "stream_message.cpp" "i2s_audio_output.cpp" "stream_buffer.cpp" "track_queue.cpp"
-  "stream_event.cpp" "stream_info.cpp" "audio_fsm.cpp"
+  "stream_event.cpp" "stream_info.cpp" "audio_fsm.cpp" "sink_mixer.cpp" "resample.cpp"
  INCLUDE_DIRS "include"
  REQUIRES "codecs" "drivers" "cbor" "result" "tasks" "span" "memory" "tinyfsm" "database" "system_fsm" "playlist")
--- a/src/audio/audio_task.cpp
+++ b/src/audio/audio_task.cpp
@ -34,6 +34,8 @@
 #include "freertos/queue.h"
 #include "freertos/ringbuf.h"
 #include "pipeline.hpp"
 #include "sample.hpp"
 #include "sink_mixer.hpp"
 #include "span.hpp"
 #include "arena.hpp"
@ -107,7 +109,10 @@ auto Timer::bytes_to_samples(uint32_t bytes) -> uint32_t {
 auto AudioTask::Start(IAudioSource* source, IAudioSink* sink) -> AudioTask* {
  AudioTask* task = new AudioTask(source, sink);
-  tasks::StartPersistent<tasks::Type::kAudio>([=]() { task->Main(); });
+  // Pin to CORE1 because codecs should be fixed point anyway, and being on
  // the opposite core to the mixer maximises throughput in the worst case
  // (some heavy codec like opus + resampling for bluetooth).
  tasks::StartPersistent<tasks::Type::kAudio>(1, [=]() { task->Main(); });
  return task;
 }
@ -115,14 +120,12 @@ AudioTask::AudioTask(IAudioSource* source, IAudioSink* sink)
    : source_(source),
      sink_(sink),
      codec_(),
      mixer_(new SinkMixer(sink->stream())),
      timer_(),
      has_begun_decoding_(false),
      current_input_format_(),
      current_output_format_(),
-      sample_buffer_(reinterpret_cast<std::byte*>(
+      codec_buffer_(new RawStream(kSampleBufferSize)) {}
          heap_caps_malloc(kSampleBufferSize,
                           MALLOC_CAP_INTERNAL | MALLOC_CAP_8BIT))),
      sample_buffer_len_(kSampleBufferSize) {}
 void AudioTask::Main() {
  for (;;) {
@ -226,7 +229,7 @@ auto AudioTask::BeginDecoding(InputStream& stream) -> bool {
  codecs::ICodec::OutputFormat format = res.second.value();
  StreamInfo::Pcm new_format{
      .channels = format.num_channels,
-      .bits_per_sample = format.bits_per_sample,
+      .bits_per_sample = 32,
      .sample_rate = format.sample_rate_hz,
  };
@ -246,13 +249,18 @@ auto AudioTask::BeginDecoding(InputStream& stream) -> bool {
    return false;
  }
  OutputStream writer{codec_buffer_.get()};
  writer.prepare(new_format, {});
  return true;
 }
 auto AudioTask::ContinueDecoding(InputStream& stream) -> bool {
  while (!stream.data().empty()) {
-    auto res = codec_->ContinueStream(stream.data(),
+    OutputStream writer{codec_buffer_.get()};
-                                      {sample_buffer_, sample_buffer_len_});
+
    auto res =
        codec_->ContinueStream(stream.data(), writer.data_as<sample::Sample>());
    stream.consume(res.first);
@ -263,9 +271,10 @@ auto AudioTask::ContinueDecoding(InputStream& stream) -> bool {
        return false;
      }
    } else {
-      xStreamBufferSend(sink_->stream(), sample_buffer_,
+      writer.add(res.second->samples_written * sizeof(sample::Sample));
-                        res.second->bytes_written, portMAX_DELAY);
+
-      timer_->AddBytes(res.second->bytes_written);
+      InputStream reader{codec_buffer_.get()};
      SendToSink(reader);
    }
  }
  return true;
@ -284,21 +293,23 @@ auto AudioTask::FinishDecoding(InputStream& stream) -> void {
    std::unique_ptr<RawStream> mad_buffer;
    mad_buffer.reset(new RawStream(stream.data().size_bytes() + 8));
-    OutputStream writer{mad_buffer.get()};
+    OutputStream mad_writer{mad_buffer.get()};
    std::copy(stream.data().begin(), stream.data().end(),
-              writer.data().begin());
+              mad_writer.data().begin());
-    std::fill(writer.data().begin(), writer.data().end(), std::byte{0});
+    std::fill(mad_writer.data().begin(), mad_writer.data().end(), std::byte{0});
    InputStream padded_stream{mad_buffer.get()};
-    auto res = codec_->ContinueStream(stream.data(),
+    OutputStream writer{codec_buffer_.get()};
-                                      {sample_buffer_, sample_buffer_len_});
+    auto res =
        codec_->ContinueStream(stream.data(), writer.data_as<sample::Sample>());
    if (res.second.has_error()) {
      return;
    }
-    xStreamBufferSend(sink_->stream(), sample_buffer_,
+    writer.add(res.second->samples_written * sizeof(sample::Sample));
-                      res.second->bytes_written, portMAX_DELAY);
+
-    timer_->AddBytes(res.second->bytes_written);
+    InputStream reader{codec_buffer_.get()};
    SendToSink(reader);
  }
 }
@ -319,24 +330,31 @@ auto AudioTask::ForwardPcmStream(StreamInfo::Pcm& format,
  xStreamBufferSend(sink_->stream(), samples.data(), samples.size_bytes(),
                    portMAX_DELAY);
  timer_->AddBytes(samples.size_bytes());
  InputStream reader{codec_buffer_.get()};
  SendToSink(reader);
  return true;
 }
 auto AudioTask::ConfigureSink(const StreamInfo::Pcm& format,
                              const Duration& duration) -> bool {
  if (format != current_output_format_) {
    current_output_format_ = format;
    StreamInfo::Pcm new_sink_format = sink_->PrepareFormat(format);
    if (new_sink_format != current_sink_format_) {
      current_sink_format_ = new_sink_format;
      // The new format is different to the old one. Wait for the sink to drain
      // before continuing.
      while (!xStreamBufferIsEmpty(sink_->stream())) {
        ESP_LOGI(kTag, "waiting for sink stream to drain...");
        // TODO(jacqueline): Get the sink drain ISR to notify us of this
        // via semaphore instead of busy-ish waiting.
-      vTaskDelay(pdMS_TO_TICKS(100));
+        vTaskDelay(pdMS_TO_TICKS(10));
      }
      ESP_LOGI(kTag, "configuring sink");
-    if (!sink_->Configure(format)) {
+      sink_->Configure(new_sink_format);
      return false;
    }
  }
@ -345,4 +363,17 @@ auto AudioTask::ConfigureSink(const StreamInfo::Pcm& format,
  return true;
 }
 auto AudioTask::SendToSink(InputStream& stream) -> void {
  std::size_t bytes_to_send = stream.data().size_bytes();
  std::size_t bytes_sent;
  if (stream.info().format_as<StreamInfo::Pcm>() == current_sink_format_) {
    bytes_sent = xStreamBufferSend(sink_->stream(), stream.data().data(),
                                   bytes_to_send, portMAX_DELAY);
    stream.consume(bytes_sent);
  } else {
    bytes_sent = mixer_->MixAndSend(stream, current_sink_format_.value());
  }
  timer_->AddBytes(bytes_sent);
 }
 }  // namespace audio
--- a/src/audio/fatfs_audio_input.cpp
+++ b/src/audio/fatfs_audio_input.cpp
@ -30,6 +30,7 @@
 #include "freertos/portmacro.h"
 #include "freertos/projdefs.h"
 #include "future_fetcher.hpp"
 #include "idf_additions.h"
 #include "span.hpp"
 #include "stream_info.hpp"
 #include "tag_parser.hpp"
@ -40,8 +41,8 @@ static const char* kTag = "SRC";
 namespace audio {
-static constexpr UINT kFileBufferSize = 4096 * 2;
+static constexpr UINT kFileBufferSize = 8 * 1024;
-static constexpr UINT kStreamerBufferSize = 4096;
+static constexpr UINT kStreamerBufferSize = 64 * 1024;
 static StreamBufferHandle_t sForwardDest = nullptr;
@ -143,7 +144,9 @@ FatfsAudioInput::FatfsAudioInput(
    : IAudioSource(),
      tag_parser_(tag_parser),
      has_data_(xSemaphoreCreateBinary()),
-      streamer_buffer_(xStreamBufferCreate(kStreamerBufferSize, 1)),
+      streamer_buffer_(xStreamBufferCreateWithCaps(kStreamerBufferSize,
                                                   1,
                                                   MALLOC_CAP_SPIRAM)),
      streamer_(new FileStreamer(streamer_buffer_, has_data_)),
      input_buffer_(new RawStream(kFileBufferSize)),
      source_mutex_(),
--- a/src/audio/i2s_audio_output.cpp
+++ b/src/audio/i2s_audio_output.cpp
@ -115,10 +115,20 @@ auto I2SAudioOutput::AdjustVolumeDown() -> bool {
  return true;
 }
-auto I2SAudioOutput::Configure(const StreamInfo::Pcm& pcm) -> bool {
+auto I2SAudioOutput::PrepareFormat(const StreamInfo::Pcm& orig)
    -> StreamInfo::Pcm {
  return StreamInfo::Pcm{
      .channels = std::min<uint8_t>(orig.channels, 2),
      .bits_per_sample = std::clamp<uint8_t>(orig.bits_per_sample, 16, 32),
      .sample_rate = 44100,
      //.sample_rate = std::clamp<uint32_t>(orig.sample_rate, 8000, 96000),
  };
 }
 auto I2SAudioOutput::Configure(const StreamInfo::Pcm& pcm) -> void {
  if (current_config_ && pcm == *current_config_) {
    ESP_LOGI(kTag, "ignoring unchanged format");
-    return true;
+    return;
  }
  ESP_LOGI(kTag, "incoming audio stream: %u ch %u bpp @ %lu Hz", pcm.channels,
@ -134,7 +144,7 @@ auto I2SAudioOutput::Configure(const StreamInfo::Pcm& pcm) -> bool {
      break;
    default:
      ESP_LOGE(kTag, "dropping stream with out of bounds channels");
-      return false;
+      return;
  }
  drivers::I2SDac::BitsPerSample bps;
@ -150,30 +160,36 @@ auto I2SAudioOutput::Configure(const StreamInfo::Pcm& pcm) -> bool {
      break;
    default:
      ESP_LOGE(kTag, "dropping stream with unknown bps");
-      return false;
+      return;
  }
  drivers::I2SDac::SampleRate sample_rate;
  switch (pcm.sample_rate) {
    case 8000:
      sample_rate = drivers::I2SDac::SAMPLE_RATE_8;
      break;
    case 32000:
      sample_rate = drivers::I2SDac::SAMPLE_RATE_32;
      break;
    case 44100:
      sample_rate = drivers::I2SDac::SAMPLE_RATE_44_1;
      break;
    case 48000:
      sample_rate = drivers::I2SDac::SAMPLE_RATE_48;
      break;
    case 88200:
      sample_rate = drivers::I2SDac::SAMPLE_RATE_88_2;
      break;
    case 96000:
      sample_rate = drivers::I2SDac::SAMPLE_RATE_96;
      break;
    default:
      ESP_LOGE(kTag, "dropping stream with unknown rate");
-      return false;
+      return;
  }
  dac_->Reconfigure(ch, bps, sample_rate);
  current_config_ = pcm;
  return true;
 }
 auto I2SAudioOutput::Send(const cpp::span<std::byte>& data) -> void {
  dac_->WriteData(data);
 }
 }  // namespace audio
--- a/src/audio/include/audio_sink.hpp
+++ b/src/audio/include/audio_sink.hpp
@ -17,7 +17,6 @@ namespace audio {
 class IAudioSink {
 private:
  // TODO: tune. at least about 12KiB seems right for mp3
  static const std::size_t kDrainBufferSize = 24 * 1024;
  StreamBufferHandle_t stream_;
@ -38,8 +37,8 @@ class IAudioSink {
  virtual auto AdjustVolumeUp() -> bool = 0;
  virtual auto AdjustVolumeDown() -> bool = 0;
-  virtual auto Configure(const StreamInfo::Pcm& format) -> bool = 0;
+  virtual auto PrepareFormat(const StreamInfo::Pcm&) -> StreamInfo::Pcm = 0;
-  virtual auto Send(const cpp::span<std::byte>& data) -> void = 0;
+  virtual auto Configure(const StreamInfo::Pcm& format) -> void = 0;
  auto stream() -> StreamBufferHandle_t { return stream_; }
 };
--- a/src/audio/include/audio_task.hpp
+++ b/src/audio/include/audio_task.hpp
@ -14,6 +14,7 @@
 #include "audio_source.hpp"
 #include "codec.hpp"
 #include "pipeline.hpp"
 #include "sink_mixer.hpp"
 #include "stream_info.hpp"
 namespace audio {
@ -63,18 +64,20 @@ class AudioTask {
  auto ForwardPcmStream(StreamInfo::Pcm&, cpp::span<const std::byte>) -> bool;
  auto ConfigureSink(const StreamInfo::Pcm&, const Duration&) -> bool;
  auto SendToSink(InputStream&) -> void;
  IAudioSource* source_;
  IAudioSink* sink_;
  std::unique_ptr<codecs::ICodec> codec_;
  std::unique_ptr<SinkMixer> mixer_;
  std::unique_ptr<Timer> timer_;
  bool has_begun_decoding_;
  std::optional<StreamInfo::Format> current_input_format_;
  std::optional<StreamInfo::Pcm> current_output_format_;
  std::optional<StreamInfo::Pcm> current_sink_format_;
-  std::byte* sample_buffer_;
+  std::unique_ptr<RawStream> codec_buffer_;
  std::size_t sample_buffer_len_;
 };
 }  // namespace audio
--- a/src/audio/include/fir.h
+++ b/src/audio/include/fir.h
@ -0,0 +1,131 @@
 /*
 * FIR filter coefficients from resample-1.x smallfilter.h 
 * see Digital Audio Resampling Home Page located at
 * http://ccrma.stanford.edu/~jos/resample/
 */
 32767, 32766, 32764, 32760, 32755, 32749, 32741, 32731, 32721, 32708,
 32695, 32679, 32663, 32645, 32625, 32604, 32582, 32558, 32533, 32506,
 32478, 32448, 32417, 32385, 32351, 32316, 32279, 32241, 32202, 32161,
 32119, 32075, 32030, 31984, 31936, 31887, 31836, 31784, 31731, 31676,
 31620, 31563, 31504, 31444, 31383, 31320, 31256, 31191, 31124, 31056,
 30987, 30916, 30845, 30771, 30697, 30621, 30544, 30466, 30387, 30306,
 30224, 30141, 30057, 29971, 29884, 29796, 29707, 29617, 29525, 29433,
 29339, 29244, 29148, 29050, 28952, 28852, 28752, 28650, 28547, 28443,
 28338, 28232, 28125, 28017, 27908, 27797, 27686, 27574, 27461, 27346,
 27231, 27115, 26998, 26879, 26760, 26640, 26519, 26398, 26275, 26151,
 26027, 25901, 25775, 25648, 25520, 25391, 25262, 25131, 25000, 24868,
 24735, 24602, 24467, 24332, 24197, 24060, 23923, 23785, 23647, 23507,
 23368, 23227, 23086, 22944, 22802, 22659, 22515, 22371, 22226, 22081,
 21935, 21789, 21642, 21494, 21346, 21198, 21049, 20900, 20750, 20600,
 20449, 20298, 20146, 19995, 19842, 19690, 19537, 19383, 19230, 19076,
 18922, 18767, 18612, 18457, 18302, 18146, 17990, 17834, 17678, 17521,
 17365, 17208, 17051, 16894, 16737, 16579, 16422, 16264, 16106, 15949,
 15791, 15633, 15475, 15317, 15159, 15001, 14843, 14685, 14527, 14369,
 14212, 14054, 13896, 13739, 13581, 13424, 13266, 13109, 12952, 12795,
 12639, 12482, 12326, 12170, 12014, 11858, 11703, 11548, 11393, 11238,
 11084, 10929, 10776, 10622, 10469, 10316, 10164, 10011, 9860, 9708,
 9557, 9407, 9256, 9106, 8957, 8808, 8659, 8511, 8364, 8216, 8070,
 7924, 7778, 7633, 7488, 7344, 7200, 7057, 6914, 6773, 6631, 6490,
 6350, 6210, 6071, 5933, 5795, 5658, 5521, 5385, 5250, 5115, 4981,
 4848, 4716, 4584, 4452, 4322, 4192, 4063, 3935, 3807, 3680, 3554,
 3429, 3304, 3180, 3057, 2935, 2813, 2692, 2572, 2453, 2335, 2217,
 2101, 1985, 1870, 1755, 1642, 1529, 1418, 1307, 1197, 1088, 979, 872,
 765, 660, 555, 451, 348, 246, 145, 44, -54, -153, -250, -347, -443,
 -537, -631, -724, -816, -908, -998, -1087, -1175, -1263, -1349, -1435,
 -1519, -1603, -1685, -1767, -1848, -1928, -2006, -2084, -2161, -2237,
 -2312, -2386, -2459, -2531, -2603, -2673, -2742, -2810, -2878, -2944,
 -3009, -3074, -3137, -3200, -3261, -3322, -3381, -3440, -3498, -3554,
 -3610, -3665, -3719, -3772, -3824, -3875, -3925, -3974, -4022, -4069,
 -4116, -4161, -4205, -4249, -4291, -4333, -4374, -4413, -4452, -4490,
 -4527, -4563, -4599, -4633, -4666, -4699, -4730, -4761, -4791, -4820,
 -4848, -4875, -4901, -4926, -4951, -4974, -4997, -5019, -5040, -5060,
 -5080, -5098, -5116, -5133, -5149, -5164, -5178, -5192, -5205, -5217,
 -5228, -5238, -5248, -5257, -5265, -5272, -5278, -5284, -5289, -5293,
 -5297, -5299, -5301, -5303, -5303, -5303, -5302, -5300, -5298, -5295,
 -5291, -5287, -5282, -5276, -5270, -5263, -5255, -5246, -5237, -5228,
 -5217, -5206, -5195, -5183, -5170, -5157, -5143, -5128, -5113, -5097,
 -5081, -5064, -5047, -5029, -5010, -4991, -4972, -4952, -4931, -4910,
 -4889, -4867, -4844, -4821, -4797, -4774, -4749, -4724, -4699, -4673,
 -4647, -4620, -4593, -4566, -4538, -4510, -4481, -4452, -4422, -4393,
 -4363, -4332, -4301, -4270, -4238, -4206, -4174, -4142, -4109, -4076,
 -4042, -4009, -3975, -3940, -3906, -3871, -3836, -3801, -3765, -3729,
 -3693, -3657, -3620, -3584, -3547, -3510, -3472, -3435, -3397, -3360,
 -3322, -3283, -3245, -3207, -3168, -3129, -3091, -3052, -3013, -2973,
 -2934, -2895, -2855, -2816, -2776, -2736, -2697, -2657, -2617, -2577,
 -2537, -2497, -2457, -2417, -2377, -2337, -2297, -2256, -2216, -2176,
 -2136, -2096, -2056, -2016, -1976, -1936, -1896, -1856, -1817, -1777,
 -1737, -1698, -1658, -1619, -1579, -1540, -1501, -1462, -1423, -1384,
 -1345, -1306, -1268, -1230, -1191, -1153, -1115, -1077, -1040, -1002,
 -965, -927, -890, -854, -817, -780, -744, -708, -672, -636, -600,
 -565, -530, -494, -460, -425, -391, -356, -322, -289, -255, -222,
 -189, -156, -123, -91, -59, -27, 4, 35, 66, 97, 127, 158, 188, 218,
 247, 277, 306, 334, 363, 391, 419, 447, 474, 501, 528, 554, 581, 606,
 632, 657, 683, 707, 732, 756, 780, 803, 827, 850, 872, 895, 917, 939,
 960, 981, 1002, 1023, 1043, 1063, 1082, 1102, 1121, 1139, 1158, 1176,
 1194, 1211, 1228, 1245, 1262, 1278, 1294, 1309, 1325, 1340, 1354,
 1369, 1383, 1397, 1410, 1423, 1436, 1448, 1461, 1473, 1484, 1496,
 1507, 1517, 1528, 1538, 1548, 1557, 1566, 1575, 1584, 1592, 1600,
 1608, 1616, 1623, 1630, 1636, 1643, 1649, 1654, 1660, 1665, 1670,
 1675, 1679, 1683, 1687, 1690, 1694, 1697, 1700, 1702, 1704, 1706,
 1708, 1709, 1711, 1712, 1712, 1713, 1713, 1713, 1713, 1712, 1711,
 1710, 1709, 1708, 1706, 1704, 1702, 1700, 1697, 1694, 1691, 1688,
 1685, 1681, 1677, 1673, 1669, 1664, 1660, 1655, 1650, 1644, 1639,
 1633, 1627, 1621, 1615, 1609, 1602, 1596, 1589, 1582, 1575, 1567,
 1560, 1552, 1544, 1536, 1528, 1520, 1511, 1503, 1494, 1485, 1476,
 1467, 1458, 1448, 1439, 1429, 1419, 1409, 1399, 1389, 1379, 1368,
 1358, 1347, 1337, 1326, 1315, 1304, 1293, 1282, 1271, 1260, 1248,
 1237, 1225, 1213, 1202, 1190, 1178, 1166, 1154, 1142, 1130, 1118,
 1106, 1094, 1081, 1069, 1057, 1044, 1032, 1019, 1007, 994, 981, 969,
 956, 943, 931, 918, 905, 892, 879, 867, 854, 841, 828, 815, 802, 790,
 777, 764, 751, 738, 725, 713, 700, 687, 674, 662, 649, 636, 623, 611,
 598, 585, 573, 560, 548, 535, 523, 510, 498, 486, 473, 461, 449, 437,
 425, 413, 401, 389, 377, 365, 353, 341, 330, 318, 307, 295, 284, 272,
 261, 250, 239, 228, 217, 206, 195, 184, 173, 163, 152, 141, 131, 121,
 110, 100, 90, 80, 70, 60, 51, 41, 31, 22, 12, 3, -5, -14, -23, -32,
 -41, -50, -59, -67, -76, -84, -93, -101, -109, -117, -125, -133, -140,
 -148, -156, -163, -170, -178, -185, -192, -199, -206, -212, -219,
 -226, -232, -239, -245, -251, -257, -263, -269, -275, -280, -286,
 -291, -297, -302, -307, -312, -317, -322, -327, -332, -336, -341,
 -345, -349, -354, -358, -362, -366, -369, -373, -377, -380, -384,
 -387, -390, -394, -397, -400, -402, -405, -408, -411, -413, -416,
 -418, -420, -422, -424, -426, -428, -430, -432, -433, -435, -436,
 -438, -439, -440, -442, -443, -444, -445, -445, -446, -447, -447,
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 -427, -426, -424, -422, -420, -419, -417, -415, -413, -411, -409,
 -407, -405, -403, -400, -398, -396, -393, -391, -389, -386, -384,
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 -285, -282, -278, -275, -272, -269, -265, -262, -259, -256, -252,
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 -178, -175, -172, -168, -165, -162, -159, -156, -153, -150, -147,
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 -111, -108, -105, -102, -99, -97, -94, -91, -88, -86, -83, -80, -78,
 -75, -72, -70, -67, -65, -62, -59, -57, -55, -52, -50, -47, -45, -43,
 -40, -38, -36, -33, -31, -29, -27, -25, -22, -20, -18, -16, -14, -12,
 -10, -8, -6, -4, -2, 0, 0, 2, 4, 6, 8, 9, 11, 13, 14, 16, 17, 19, 21,
 22, 24, 25, 27, 28, 29, 31, 32, 33, 35, 36, 37, 38, 40, 41, 42, 43,
 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59,
 59, 60, 61, 62, 62, 63, 63, 64, 64, 65, 66, 66, 66, 67, 67, 68, 68,
 69, 69, 69, 70, 70, 70, 70, 71, 71, 71, 71, 71, 72, 72, 72, 72, 72,
 72, 72, 72, 72, 72, 72, 72, 72, 72, 72, 72, 72, 72, 72, 72, 72, 72,
 72, 71, 71, 71, 71, 71, 70, 70, 70, 70, 69, 69, 69, 69, 68, 68, 68,
 67, 67, 67, 66, 66, 66, 65, 65, 64, 64, 64, 63, 63, 62, 62, 62, 61,
 61, 60, 60, 59, 59, 58, 58, 58, 57, 57, 56, 56, 55, 55, 54, 54, 53,
 53, 52, 52, 51, 51, 50, 50, 49, 48, 48, 47, 47, 46, 46, 45, 45, 44,
 44, 43, 43, 42, 42, 41, 41, 40, 39, 39, 38, 38, 37, 37, 36, 36, 35,
 35, 34, 34, 33, 33, 32, 32, 31, 31, 30, 30, 29, 29, 28, 28, 27, 27,
 26, 26, 25, 25, 24, 24, 23, 23, 23, 22, 22, 21, 21, 20, 20, 20, 19,
 19, 18, 18, 17, 17, 17, 16, 16, 15, 15, 15, 14, 14, 14, 13, 13, 12,
 12, 12, 11, 11, 11, 10, 10, 10, 9, 9, 9, 9, 8, 8, 8, 7, 7, 7, 7, 6, 6,
 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 1, 1, 1,
 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, -1, -1, -1,
 -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -2, -2, -2, -2, -2, -2, -2,
 -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2,
 -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2,
 -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2, -2,
 -2, -2, -2, -2, -2, -2, -2, -2, -1, -1, -1, -1, -1, -1, -1, -1, -1,
 -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
--- a/src/audio/include/i2s_audio_output.hpp
+++ b/src/audio/include/i2s_audio_output.hpp
@ -35,8 +35,8 @@ class I2SAudioOutput : public IAudioSink {
  auto AdjustVolumeUp() -> bool override;
  auto AdjustVolumeDown() -> bool override;
-  auto Configure(const StreamInfo::Pcm& format) -> bool override;
+  auto PrepareFormat(const StreamInfo::Pcm&) -> StreamInfo::Pcm override;
-  auto Send(const cpp::span<std::byte>& data) -> void override;
+  auto Configure(const StreamInfo::Pcm& format) -> void override;
  I2SAudioOutput(const I2SAudioOutput&) = delete;
  I2SAudioOutput& operator=(const I2SAudioOutput&) = delete;
--- a/src/audio/include/resample.hpp
+++ b/src/audio/include/resample.hpp
@ -0,0 +1,44 @@
 #pragma once
 #include <sys/_stdint.h>
 #include <vector>
 #include "span.hpp"
 #include "sample.hpp"
 namespace audio {
 class Resampler {
 public:
  Resampler(uint32_t source_sample_rate,
            uint32_t target_sample_rate,
            uint8_t num_channels);
  ~Resampler();
  auto source_sample_rate() -> uint32_t { return source_sample_rate_; }
  auto target_sample_rate() -> uint32_t { return target_sample_rate_; }
  auto channels() -> uint_fast8_t { return num_channels_; }
  auto Process(cpp::span<const sample::Sample> input,
               cpp::span<sample::Sample> output,
               bool end_of_data) -> std::pair<size_t, size_t>;
 private:
  auto Subsample(int channel) -> float;
  auto ApplyFilter(cpp::span<float> filter, cpp::span<float> input) -> float;
  uint32_t source_sample_rate_;
  uint32_t target_sample_rate_;
  float factor_;
  uint8_t num_channels_;
  std::vector<float*> channel_buffers_;
  size_t channel_buffer_size_;
  float output_offset_;
  int32_t input_index_;
 };
 }  // namespace audio
--- a/src/audio/include/sink_mixer.hpp
+++ b/src/audio/include/sink_mixer.hpp
@ -0,0 +1,71 @@
 /*
 * Copyright 2023 jacqueline <me@jacqueline.id.au>
 *
 * SPDX-License-Identifier: GPL-3.0-only
 */
 #pragma once
 #include <sys/_stdint.h>
 #include <cstdint>
 #include <memory>
 #include "resample.hpp"
 #include "sample.hpp"
 #include "audio_decoder.hpp"
 #include "audio_sink.hpp"
 #include "audio_source.hpp"
 #include "codec.hpp"
 #include "pipeline.hpp"
 #include "stream_info.hpp"
 namespace audio {
 /*
 * Handles the final downmix + resample + quantisation stage of audio,
 * generation sending the result directly to an IAudioSink.
 */
 class SinkMixer {
 public:
  SinkMixer(StreamBufferHandle_t dest);
  ~SinkMixer();
  auto MixAndSend(InputStream&, const StreamInfo::Pcm&) -> std::size_t;
 private:
  auto Main() -> void;
  auto SetTargetFormat(const StreamInfo::Pcm& format) -> void;
  auto HandleBytes() -> void;
  auto Resample(InputStream&, OutputStream&) -> bool;
  auto ApplyDither(cpp::span<sample::Sample> samples, uint_fast8_t bits)
      -> void;
  auto Downscale(cpp::span<sample::Sample>, cpp::span<int16_t>) -> void;
  enum class Command {
    kReadBytes,
    kSetSourceFormat,
    kSetTargetFormat,
  };
  struct Args {
    Command cmd;
    StreamInfo::Pcm format;
  };
  QueueHandle_t commands_;
  SemaphoreHandle_t is_idle_;
  std::unique_ptr<Resampler> resampler_;
  std::unique_ptr<RawStream> input_stream_;
  std::unique_ptr<RawStream> resampled_stream_;
  StreamInfo::Pcm target_format_;
  StreamBufferHandle_t source_;
  StreamBufferHandle_t sink_;
 };
 }  // namespace audio
--- a/src/audio/include/stream_info.hpp
+++ b/src/audio/include/stream_info.hpp
@ -16,6 +16,7 @@
 #include <utility>
 #include <variant>
 #include "esp_heap_caps.h"
 #include "freertos/FreeRTOS.h"
 #include "freertos/ringbuf.h"
 #include "freertos/stream_buffer.h"
@ -56,6 +57,18 @@ class StreamInfo {
    bool operator==(const Encoded&) const = default;
  };
  /*
   * Two-channel, interleaved, 32-bit floating point pcm samples.
   */
  struct FloatingPointPcm {
    // Number of channels in this stream.
    uint8_t channels;
    // The sample rate.
    uint32_t sample_rate;
    bool operator==(const FloatingPointPcm&) const = default;
  };
  struct Pcm {
    // Number of channels in this stream.
    uint8_t channels;
@ -64,10 +77,14 @@ class StreamInfo {
    // The sample rate.
    uint32_t sample_rate;
    auto bytes_per_sample() const -> uint8_t {
      return bits_per_sample == 16 ? 2 : 4;
    }
    bool operator==(const Pcm&) const = default;
  };
-  typedef std::variant<std::monostate, Encoded, Pcm> Format;
+  typedef std::variant<std::monostate, Encoded, FloatingPointPcm, Pcm> Format;
  auto format() const -> const Format& { return format_; }
  auto set_format(Format f) -> void { format_ = f; }
@ -94,10 +111,17 @@ class OutputStream;
 class RawStream {
 public:
  explicit RawStream(std::size_t size);
  RawStream(std::size_t size, uint32_t);
  ~RawStream();
  auto info() -> StreamInfo& { return info_; }
  auto data() -> cpp::span<std::byte>;
  template <typename T>
  auto data_as() -> cpp::span<T> {
    auto orig = data();
    return {reinterpret_cast<T*>(orig.data()), orig.size_bytes() / sizeof(T)};
  }
  auto empty() const -> bool { return info_.bytes_in_stream() == 0; }
 private:
  StreamInfo info_;
@ -114,6 +138,12 @@ class InputStream {
  const StreamInfo& info() const;
  cpp::span<const std::byte> data() const;
  template <typename T>
  auto data_as() const -> cpp::span<const T> {
    auto orig = data();
    return {reinterpret_cast<const T*>(orig.data()),
            orig.size_bytes() / sizeof(T)};
  }
 private:
  RawStream* raw_;
@ -131,6 +161,11 @@ class OutputStream {
  const StreamInfo& info() const;
  cpp::span<std::byte> data() const;
  template <typename T>
  auto data_as() const -> cpp::span<T> {
    auto orig = data();
    return {reinterpret_cast<T*>(orig.data()), orig.size_bytes() / sizeof(T)};
  }
 private:
  RawStream* raw_;
--- a/src/audio/resample.cpp
+++ b/src/audio/resample.cpp
@ -0,0 +1,205 @@
 /*
 * Copyright 2023 jacqueline <me@jacqueline.id.au>
 *
 * SPDX-License-Identifier: GPL-3.0-only
 */
 #include "resample.hpp"
 /*
 * This file contains the implementation for a 32-bit floating point resampler.
 * It is largely based on David Bryant's ART resampler, which is BSD-licensed,
 * and available at https://github.com/dbry/audio-resampler/.
 *
 * This resampler uses windowed sinc interpolation filters, with an additional
 * lowpass filter to reduce aliasing.
 */
 #include <stdint.h>
 #include <stdlib.h>
 #include <string.h>
 #include <algorithm>
 #include <cmath>
 #include <numeric>
 #include "esp_log.h"
 #include "sample.hpp"
 #include "stream_info.hpp"
 namespace audio {
 static constexpr double kLowPassRatio = 0.5;
 static constexpr size_t kNumFilters = 64;
 static constexpr size_t kFilterSize = 16;
 typedef std::array<float, kFilterSize> Filter;
 static std::array<Filter, kNumFilters + 1> sFilters{};
 static bool sFiltersInitialised = false;
 auto InitFilter(int index) -> void;
 Resampler::Resampler(uint32_t source_sample_rate,
                     uint32_t target_sample_rate,
                     uint8_t num_channels)
    : source_sample_rate_(source_sample_rate),
      target_sample_rate_(target_sample_rate),
      factor_(static_cast<double>(target_sample_rate) /
              static_cast<double>(source_sample_rate)),
      num_channels_(num_channels) {
  channel_buffers_.resize(num_channels);
  channel_buffer_size_ = kFilterSize * 16;
  for (int i = 0; i < num_channels; i++) {
    channel_buffers_[i] =
        static_cast<float*>(calloc(sizeof(float), channel_buffer_size_));
  }
  output_offset_ = kFilterSize / 2.0f;
  input_index_ = kFilterSize;
  if (!sFiltersInitialised) {
    sFiltersInitialised = true;
    for (int i = 0; i < kNumFilters + 1; i++) {
      InitFilter(i);
    }
  }
 }
 Resampler::~Resampler() {}
 auto Resampler::Process(cpp::span<const sample::Sample> input,
                        cpp::span<sample::Sample> output,
                        bool end_of_data) -> std::pair<size_t, size_t> {
  size_t samples_used = 0;
  size_t samples_produced = 0;
  size_t input_frames = input.size() / num_channels_;
  size_t output_frames = output.size() / num_channels_;
  int half_taps = kFilterSize / 2;
  while (output_frames > 0) {
    if (output_offset_ >= input_index_ - half_taps) {
      if (input_frames > 0) {
        // Check whether the channel buffers will overflow with the addition of
        // this sample. If so, we need to move the remaining contents back to
        // the beginning of the buffer.
        if (input_index_ == channel_buffer_size_) {
          for (int i = 0; i < num_channels_; ++i) {
            memmove(channel_buffers_[i],
                    channel_buffers_[i] + channel_buffer_size_ - kFilterSize,
                    kFilterSize * sizeof(float));
          }
          output_offset_ -= channel_buffer_size_ - kFilterSize;
          input_index_ -= channel_buffer_size_ - kFilterSize;
        }
        for (int i = 0; i < num_channels_; ++i) {
          channel_buffers_[i][input_index_] =
              sample::ToFloat(input[samples_used++]);
        }
        input_index_++;
        input_frames--;
      } else {
        break;
      }
    } else {
      for (int i = 0; i < num_channels_; i++) {
        output[samples_produced++] = sample::FromFloat(Subsample(i));
      }
      // NOTE: floating point division here is potentially slow due to FPU
      // limitations. Consider explicitly bunding the xtensa libgcc divsion via
      // reciprocal implementation if we care about portability between
      // compilers.
      output_offset_ += 1.0f / factor_;
      output_frames--;
    }
  }
  return {samples_used, samples_produced};
 }
 /*
 * Constructs the filter in-place for the given index of sFilters. This only
 * needs to be done once, per-filter. 64-bit math is okay here, because filters
 * will not be initialised within a performance critical path.
 */
 auto InitFilter(int index) -> void {
  Filter& filter = sFilters[index];
  std::array<double, kFilterSize> working_buffer{};
  double fraction = index / static_cast<double>(kNumFilters);
  double filter_sum = 0.0;
  for (int i = 0; i < kFilterSize; ++i) {
    // "dist" is the absolute distance from the sinc maximum to the filter tap
    //  to be calculated, in radians.
    double dist = fabs((kFilterSize / 2.0 - 1.0) + fraction - i) * M_PI;
    // "ratio" is that distance divided by half the tap count such that it
    // reaches π at the window extremes
    double ratio = dist / (kFilterSize / 2.0);
    double value;
    if (dist != 0.0) {
      value = sin(dist * kLowPassRatio) / (dist * kLowPassRatio);
      // Hann window. We could alternatively use a Blackman Harris window,
      // however our unusually small filter size makes the Hann window's
      // steeper cutoff more important.
      value *= 0.5 * (1.0 + cos(ratio));
    } else {
      value = 1.0;
    }
    working_buffer[i] = value;
    filter_sum += value;
  }
  // Filter should have unity DC gain
  double scaler = 1.0 / filter_sum;
  double error = 0.0;
  for (int i = kFilterSize / 2; i < kFilterSize;
       i = kFilterSize - i - (i >= kFilterSize / 2)) {
    working_buffer[i] *= scaler;
    filter[i] = working_buffer[i] - error;
    error += static_cast<double>(filter[i]) - working_buffer[i];
  }
 }
 /*
 * Performs sub-sampling with interpolation for the given channel. Assumes that
 * the channel buffer has already been filled with samples.
 */
 auto Resampler::Subsample(int channel) -> float {
  cpp::span<float> source{channel_buffers_[channel], channel_buffer_size_};
  int offset_integral = std::floor(output_offset_);
  source = source.subspan(offset_integral);
  float offset_fractional = output_offset_ - offset_integral;
  offset_fractional *= kNumFilters;
  int filter_index = std::floor(offset_fractional);
  float sum1 = ApplyFilter(sFilters[filter_index],
                           {source.data() - kFilterSize / 2 + 1, kFilterSize});
  offset_fractional -= filter_index;
  float sum2 = ApplyFilter(sFilters[filter_index + 1],
                           {source.data() - kFilterSize / 2 + 1, kFilterSize});
  return (sum2 * offset_fractional) + (sum1 * (1.0f - offset_fractional));
 }
 auto Resampler::ApplyFilter(cpp::span<float> filter, cpp::span<float> input)
    -> float {
  float sum = 0.0;
  for (int i = 0; i < kFilterSize; i++) {
    sum += filter[i] * input[i];
  }
  return sum;
 }
 }  // namespace audio
--- a/src/audio/sink_mixer.cpp
+++ b/src/audio/sink_mixer.cpp
@ -0,0 +1,224 @@
 /*
 * Copyright 2023 jacqueline <me@jacqueline.id.au>
 *
 * SPDX-License-Identifier: GPL-3.0-only
 */
 #include "sink_mixer.hpp"
 #include <stdint.h>
 #include <cmath>
 #include "esp_heap_caps.h"
 #include "esp_log.h"
 #include "freertos/portmacro.h"
 #include "freertos/projdefs.h"
 #include "idf_additions.h"
 #include "resample.hpp"
 #include "sample.hpp"
 #include "stream_info.hpp"
 #include "tasks.hpp"
 static constexpr char kTag[] = "mixer";
 static constexpr std::size_t kSourceBufferLength = 8 * 1024;
 static constexpr std::size_t kSampleBufferLength = 240 * 2 * sizeof(int32_t);
 namespace audio {
 SinkMixer::SinkMixer(StreamBufferHandle_t dest)
    : commands_(xQueueCreate(1, sizeof(Args))),
      is_idle_(xSemaphoreCreateBinary()),
      resampler_(nullptr),
      source_(xStreamBufferCreateWithCaps(kSourceBufferLength,
                                          1,
                                          MALLOC_CAP_SPIRAM)),
      sink_(dest) {
  input_stream_.reset(new RawStream(kSampleBufferLength));
  resampled_stream_.reset(new RawStream(kSampleBufferLength));
  // Pin to CORE0 because we need the FPU.
  // FIXME: A fixed point implementation could run freely on either core,
  // which should lead to a big performance increase.
  tasks::StartPersistent<tasks::Type::kMixer>(0, [&]() { Main(); });
 }
 SinkMixer::~SinkMixer() {
  vQueueDelete(commands_);
  vSemaphoreDelete(is_idle_);
  vStreamBufferDelete(source_);
 }
 auto SinkMixer::MixAndSend(InputStream& input, const StreamInfo::Pcm& target)
    -> std::size_t {
  if (input.info().format_as<StreamInfo::Pcm>() !=
      input_stream_->info().format_as<StreamInfo::Pcm>()) {
    xSemaphoreTake(is_idle_, portMAX_DELAY);
    Args args{
        .cmd = Command::kSetSourceFormat,
        .format = input.info().format_as<StreamInfo::Pcm>().value(),
    };
    xQueueSend(commands_, &args, portMAX_DELAY);
    xSemaphoreGive(is_idle_);
  }
  if (target_format_ != target) {
    xSemaphoreTake(is_idle_, portMAX_DELAY);
    Args args{
        .cmd = Command::kSetTargetFormat,
        .format = target,
    };
    xQueueSend(commands_, &args, portMAX_DELAY);
    xSemaphoreGive(is_idle_);
  }
  Args args{
      .cmd = Command::kReadBytes,
      .format = {},
  };
  xQueueSend(commands_, &args, portMAX_DELAY);
  auto buf = input.data();
  std::size_t bytes_sent =
      xStreamBufferSend(source_, buf.data(), buf.size_bytes(), portMAX_DELAY);
  input.consume(bytes_sent);
  return bytes_sent;
 }
 auto SinkMixer::Main() -> void {
  OutputStream input_receiver{input_stream_.get()};
  xSemaphoreGive(is_idle_);
  for (;;) {
    Args args;
    while (!xQueueReceive(commands_, &args, portMAX_DELAY)) {
    }
    switch (args.cmd) {
      case Command::kSetSourceFormat:
        ESP_LOGI(kTag, "setting source format");
        input_receiver.prepare(args.format, {});
        resampler_.reset();
        break;
      case Command::kSetTargetFormat:
        ESP_LOGI(kTag, "setting target format");
        target_format_ = args.format;
        resampler_.reset();
        break;
      case Command::kReadBytes:
        xSemaphoreTake(is_idle_, 0);
        while (!xStreamBufferIsEmpty(source_)) {
          auto buf = input_receiver.data();
          std::size_t bytes_received = xStreamBufferReceive(
              source_, buf.data(), buf.size_bytes(), portMAX_DELAY);
          input_receiver.add(bytes_received);
          HandleBytes();
        }
        xSemaphoreGive(is_idle_);
        break;
    }
  }
 }
 auto SinkMixer::HandleBytes() -> void {
  InputStream input{input_stream_.get()};
  auto pcm = input.info().format_as<StreamInfo::Pcm>();
  if (!pcm) {
    ESP_LOGE(kTag, "mixer got unsupported data");
    return;
  }
  if (*pcm == target_format_) {
    // The happiest possible case: the input format matches the output
    // format already. Streams like this should probably have bypassed the
    // mixer.
    // TODO(jacqueline): Make this an error; it's slow to use the mixer in this
    // case, compared to just writing directly to the sink.
    auto buf = input.data();
    std::size_t bytes_sent =
        xStreamBufferSend(sink_, buf.data(), buf.size_bytes(), portMAX_DELAY);
    input.consume(bytes_sent);
    return;
  }
  while (input_stream_->info().bytes_in_stream() >= sizeof(sample::Sample)) {
    RawStream* output_source;
    if (pcm->sample_rate != target_format_.sample_rate) {
      OutputStream resampled_writer{resampled_stream_.get()};
      if (Resample(input, resampled_writer)) {
        // Zero samples used or written. We need more input.
        break;
      }
      output_source = resampled_stream_.get();
    } else {
      output_source = input_stream_.get();
    }
    size_t bytes_consumed = output_source->info().bytes_in_stream();
    size_t bytes_to_send = output_source->info().bytes_in_stream();
    if (target_format_.bits_per_sample == 16) {
      // This is slightly scary; we're basically reaching into the internals of
      // the stream buffer to do in-place conversion of samples. Saving an
      // extra buffer + copy into that buffer is certainly worth it however.
      cpp::span<sample::Sample> src =
          output_source->data_as<sample::Sample>().first(
              output_source->info().bytes_in_stream() / sizeof(sample::Sample));
      cpp::span<int16_t> dest{reinterpret_cast<int16_t*>(src.data()),
                              src.size()};
      ApplyDither(src, 16);
      Downscale(src, dest);
      bytes_consumed = src.size_bytes();
      bytes_to_send = src.size_bytes() / 2;
    }
    InputStream output{output_source};
    cpp::span<const std::byte> buf = output.data();
    size_t bytes_sent = 0;
    while (bytes_sent < bytes_to_send) {
      auto cropped = buf.subspan(bytes_sent, bytes_to_send - bytes_sent);
      bytes_sent += xStreamBufferSend(sink_, cropped.data(),
                                      cropped.size_bytes(), portMAX_DELAY);
    }
    output.consume(bytes_consumed);
  }
 }
 auto SinkMixer::Resample(InputStream& in, OutputStream& out) -> bool {
  if (resampler_ == nullptr) {
    ESP_LOGI(kTag, "creating new resampler");
    auto format = in.info().format_as<StreamInfo::Pcm>();
    resampler_.reset(new Resampler(
        format->sample_rate, target_format_.sample_rate, format->channels));
  }
  auto res = resampler_->Process(in.data_as<sample::Sample>(),
                                 out.data_as<sample::Sample>(), false);
  in.consume(res.first * sizeof(sample::Sample));
  out.add(res.second * sizeof(sample::Sample));
  return res.first == 0 && res.second == 0;
 }
 auto SinkMixer::Downscale(cpp::span<sample::Sample> samples,
                          cpp::span<int16_t> output) -> void {
  for (size_t i = 0; i < samples.size(); i++) {
    output[i] = sample::ToSigned16Bit(samples[i]);
  }
 }
 auto SinkMixer::ApplyDither(cpp::span<sample::Sample> samples,
                            uint_fast8_t bits) -> void {
  static uint32_t prnd;
  for (auto& s : samples) {
    prnd = (prnd * 0x19660dL + 0x3c6ef35fL) & 0xffffffffL;
    s = sample::Clip(
        static_cast<int64_t>(s) +
        (static_cast<int>(prnd) >> (sizeof(sample::Sample) - bits)));
  }
 }
 }  // namespace audio
--- a/src/audio/stream_info.cpp
+++ b/src/audio/stream_info.cpp
@ -30,6 +30,13 @@ RawStream::RawStream(std::size_t size)
  assert(buffer_ != NULL);
 }
 RawStream::RawStream(std::size_t size, uint32_t caps)
    : info_(),
      buffer_size_(size),
      buffer_(reinterpret_cast<std::byte*>(heap_caps_malloc(size, caps))) {
  assert(buffer_ != NULL);
 }
 RawStream::~RawStream() {
  heap_caps_free(buffer_);
 }
--- a/src/codecs/CMakeLists.txt
+++ b/src/codecs/CMakeLists.txt
@ -3,9 +3,9 @@
 # SPDX-License-Identifier: GPL-3.0-only
 idf_component_register(
-  SRCS "codec.cpp" "mad.cpp" "foxenflac.cpp" "stbvorbis.cpp"
+  SRCS "codec.cpp" "mad.cpp" "foxenflac.cpp"
  INCLUDE_DIRS "include"
-  REQUIRES "result" "span" "libmad" "libfoxenflac" "stb_vorbis")
+  REQUIRES "result" "span" "libmad" "libfoxenflac")
 target_compile_options("${COMPONENT_LIB}" PRIVATE ${EXTRA_WARNINGS})
--- a/src/codecs/codec.cpp
+++ b/src/codecs/codec.cpp
@ -11,7 +11,6 @@
 #include "foxenflac.hpp"
 #include "mad.hpp"
 #include "stbvorbis.hpp"
 #include "types.hpp"
 namespace codecs {
@ -22,8 +21,6 @@ auto CreateCodecForType(StreamType type) -> std::optional<ICodec*> {
      return new MadMp3Decoder();
    case StreamType::kFlac:
      return new FoxenFlacDecoder();
    case StreamType::kVorbis:
      return new StbVorbisDecoder();
    default:
      return {};
  }
--- a/src/codecs/foxenflac.cpp
+++ b/src/codecs/foxenflac.cpp
@ -12,6 +12,7 @@
 #include "esp_log.h"
 #include "foxen/flac.h"
 #include "sample.hpp"
 namespace codecs {
@ -47,7 +48,6 @@ auto FoxenFlacDecoder::BeginStream(const cpp::span<const std::byte> input)
  OutputFormat format{
      .num_channels = static_cast<uint8_t>(channels),
      .bits_per_sample = 32,  // libfoxenflac output is fixed-size.
      .sample_rate_hz = static_cast<uint32_t>(fs),
      .duration_seconds = {},
      .bits_per_second = {},
@ -62,7 +62,7 @@ auto FoxenFlacDecoder::BeginStream(const cpp::span<const std::byte> input)
 }
 auto FoxenFlacDecoder::ContinueStream(cpp::span<const std::byte> input,
-                                      cpp::span<std::byte> output)
+                                      cpp::span<sample::Sample> output)
    -> Result<OutputInfo> {
  cpp::span<int32_t> output_as_samples{
      reinterpret_cast<int32_t*>(output.data()), output.size_bytes() / 4};
@ -78,7 +78,7 @@ auto FoxenFlacDecoder::ContinueStream(cpp::span<const std::byte> input,
  if (samples_written > 0) {
    return {bytes_read,
-            OutputInfo{.bytes_written = samples_written * 4,
+            OutputInfo{.samples_written = samples_written,
                       .is_finished_writing = state == FLAC_END_OF_FRAME}};
  }
--- a/src/codecs/include/codec.hpp
+++ b/src/codecs/include/codec.hpp
@ -17,6 +17,7 @@
 #include <utility>
 #include "result.hpp"
 #include "sample.hpp"
 #include "span.hpp"
 #include "types.hpp"
@ -61,7 +62,6 @@ class ICodec {
  struct OutputFormat {
    uint8_t num_channels;
    uint8_t bits_per_sample;
    uint32_t sample_rate_hz;
    std::optional<uint32_t> duration_seconds;
@ -76,7 +76,7 @@ class ICodec {
      -> Result<OutputFormat> = 0;
  struct OutputInfo {
-    std::size_t bytes_written;
+    std::size_t samples_written;
    bool is_finished_writing;
  };
@ -84,7 +84,7 @@ class ICodec {
   * Writes PCM samples to the given output buffer.
   */
  virtual auto ContinueStream(cpp::span<const std::byte> input,
-                              cpp::span<std::byte> output)
+                              cpp::span<sample::Sample> output)
      -> Result<OutputInfo> = 0;
  virtual auto SeekStream(cpp::span<const std::byte> input,
--- a/src/codecs/include/foxenflac.hpp
+++ b/src/codecs/include/foxenflac.hpp
@ -14,6 +14,7 @@
 #include <utility>
 #include "foxen/flac.h"
 #include "sample.hpp"
 #include "span.hpp"
 #include "codec.hpp"
@ -26,7 +27,7 @@ class FoxenFlacDecoder : public ICodec {
  ~FoxenFlacDecoder();
  auto BeginStream(cpp::span<const std::byte>) -> Result<OutputFormat> override;
-  auto ContinueStream(cpp::span<const std::byte>, cpp::span<std::byte>)
+  auto ContinueStream(cpp::span<const std::byte>, cpp::span<sample::Sample>)
      -> Result<OutputInfo> override;
  auto SeekStream(cpp::span<const std::byte> input, std::size_t target_sample)
      -> Result<void> override;
--- a/src/codecs/include/mad.hpp
+++ b/src/codecs/include/mad.hpp
@ -13,6 +13,7 @@
 #include <utility>
 #include "mad.h"
 #include "sample.hpp"
 #include "span.hpp"
 #include "codec.hpp"
@ -35,7 +36,7 @@ class MadMp3Decoder : public ICodec {
   * Writes samples for the current frame.
   */
  auto ContinueStream(cpp::span<const std::byte> input,
-                      cpp::span<std::byte> output)
+                      cpp::span<sample::Sample> output)
      -> Result<OutputInfo> override;
  auto SeekStream(cpp::span<const std::byte> input, std::size_t target_sample)
--- a/src/codecs/include/sample.hpp
+++ b/src/codecs/include/sample.hpp
@ -0,0 +1,65 @@
 #pragma once
 #include <stdint.h>
 #include <algorithm>
 #include <mad.h>
 namespace sample {
 // A signed, 32-bit PCM sample.
 typedef int32_t Sample;
 constexpr auto Clip(int64_t v) -> Sample {
  if (v > INT32_MAX)
    return INT32_MAX;
  if (v < INT32_MIN)
    return INT32_MIN;
  return v;
 }
 constexpr auto FromSigned(int32_t src, uint_fast8_t bits) -> Sample {
  // Left-align samples, effectively scaling them up to 32 bits.
  return src << (sizeof(Sample) * 8 - bits);
 }
 constexpr auto FromUnsigned(uint32_t src, uint_fast8_t bits) -> Sample {
  // Left-align, then substract the max value / 2 to make the sample centred
  // around zero.
  return (src << (sizeof(uint32_t) * 8 - bits)) - (~0UL >> 1);
 }
 constexpr auto FromFloat(float src) -> Sample {
  return std::clamp<float>(src, -1.0f, 1.0f) * static_cast<float>(INT32_MAX);
 }
 constexpr auto FromDouble(double src) -> Sample {
  return std::clamp<double>(src, -1.0, 1.0) * static_cast<double>(INT32_MAX);
 }
 constexpr auto FromMad(mad_fixed_t src) -> Sample {
  // Round the bottom bits.
  src += (1L << (MAD_F_FRACBITS - 24));
  // Clip the leftover bits to within range.
  if (src >= MAD_F_ONE)
    src = MAD_F_ONE - 1;
  else if (src < -MAD_F_ONE)
    src = -MAD_F_ONE;
  // Quantize.
  return FromSigned(src >> (MAD_F_FRACBITS + 1 - 24), 24);
 }
 constexpr auto ToSigned16Bit(Sample src) -> int16_t {
  return src >> 16;
 }
 static constexpr float kFactor = 1.0f / static_cast<float>(INT32_MAX);
 constexpr auto ToFloat(Sample src) -> float {
  return src * kFactor;
 }
 }  // namespace sample
--- a/src/codecs/mad.cpp
+++ b/src/codecs/mad.cpp
@ -17,24 +17,11 @@
 #include "codec.hpp"
 #include "esp_log.h"
 #include "result.hpp"
 #include "sample.hpp"
 #include "types.hpp"
 namespace codecs {
 static uint32_t mad_fixed_to_pcm(mad_fixed_t sample, uint8_t bits) {
  // Round the bottom bits.
  sample += (1L << (MAD_F_FRACBITS - bits));
  // Clip the leftover bits to within range.
  if (sample >= MAD_F_ONE)
    sample = MAD_F_ONE - 1;
  else if (sample < -MAD_F_ONE)
    sample = -MAD_F_ONE;
  // Quantize.
  return sample >> (MAD_F_FRACBITS + 1 - bits);
 }
 MadMp3Decoder::MadMp3Decoder() {
  mad_stream_init(&stream_);
  mad_frame_init(&frame_);
@ -83,7 +70,6 @@ auto MadMp3Decoder::BeginStream(const cpp::span<const std::byte> input)
  uint8_t channels = MAD_NCHANNELS(&header);
  OutputFormat output{
      .num_channels = channels,
      .bits_per_sample = 24,  // We always scale to 24 bits
      .sample_rate_hz = header.samplerate,
      .duration_seconds = {},
      .bits_per_second = {},
@ -100,7 +86,7 @@ auto MadMp3Decoder::BeginStream(const cpp::span<const std::byte> input)
 }
 auto MadMp3Decoder::ContinueStream(cpp::span<const std::byte> input,
-                                   cpp::span<std::byte> output)
+                                   cpp::span<sample::Sample> output)
    -> Result<OutputInfo> {
  std::size_t bytes_read = 0;
  if (current_sample_ < 0) {
@ -133,32 +119,24 @@ auto MadMp3Decoder::ContinueStream(cpp::span<const std::byte> input,
    bytes_read = GetBytesUsed(input.size_bytes());
  }
-  size_t output_byte = 0;
+  size_t output_sample = 0;
  while (current_sample_ < synth_.pcm.length) {
-    if (output_byte + (4 * synth_.pcm.channels) >= output.size()) {
+    if (output_sample + synth_.pcm.channels >= output.size()) {
-      // We can't fit the next sample into the buffer. Stop now, and also avoid
+      // We can't fit the next full frame into the buffer.
-      // writing the sample for only half the channels.
+      return {bytes_read, OutputInfo{.samples_written = output_sample,
      return {bytes_read, OutputInfo{.bytes_written = output_byte,
                                     .is_finished_writing = false}};
    }
    for (int channel = 0; channel < synth_.pcm.channels; channel++) {
-      uint32_t sample_24 =
+      output[output_sample++] =
-          mad_fixed_to_pcm(synth_.pcm.samples[channel][current_sample_], 24);
+          sample::FromMad(synth_.pcm.samples[channel][current_sample_]);
      // 24 bit samples must still be aligned to 32 bits. The LSB is ignored.
      output[output_byte++] = static_cast<std::byte>(0);
      output[output_byte++] = static_cast<std::byte>((sample_24)&0xFF);
      output[output_byte++] = static_cast<std::byte>((sample_24 >> 8) & 0xFF);
      output[output_byte++] = static_cast<std::byte>((sample_24 >> 16) & 0xFF);
    }
    current_sample_++;
  }
  // We wrote everything! Reset, ready for the next frame.
  current_sample_ = -1;
-  return {bytes_read, OutputInfo{.bytes_written = output_byte,
+  return {bytes_read, OutputInfo{.samples_written = output_sample,
                                 .is_finished_writing = true}};
 }
--- a/src/drivers/CMakeLists.txt
+++ b/src/drivers/CMakeLists.txt
@ -5,6 +5,7 @@
 idf_component_register(
  SRCS "touchwheel.cpp" "i2s_dac.cpp" "gpios.cpp" "battery.cpp" "storage.cpp" "i2c.cpp"
  "spi.cpp" "display.cpp" "display_init.cpp" "samd.cpp" "relative_wheel.cpp" "wm8523.cpp"
  "nvs.cpp" "bluetooth.cpp"
  INCLUDE_DIRS "include"
-  REQUIRES "esp_adc" "fatfs" "result" "lvgl" "span" "tasks")
+  REQUIRES "esp_adc" "fatfs" "result" "lvgl" "span" "tasks" "nvs_flash" "bt" "tinyfsm")
 target_compile_options(${COMPONENT_LIB} PRIVATE ${EXTRA_WARNINGS})
--- a/src/drivers/bluetooth.cpp
+++ b/src/drivers/bluetooth.cpp
@ -0,0 +1,253 @@
 #include "bluetooth.hpp"
 #include <stdint.h>
 #include <atomic>
 #include <ostream>
 #include <sstream>
 #include "esp_a2dp_api.h"
 #include "esp_avrc_api.h"
 #include "esp_bt.h"
 #include "esp_bt_device.h"
 #include "esp_bt_main.h"
 #include "esp_gap_bt_api.h"
 #include "esp_log.h"
 #include "esp_mac.h"
 #include "tinyfsm/include/tinyfsm.hpp"
 namespace drivers {
 static constexpr char kTag[] = "bluetooth";
 static std::atomic<StreamBufferHandle_t> sStream;
 auto gap_cb(esp_bt_gap_cb_event_t event, esp_bt_gap_cb_param_t* param) -> void {
  tinyfsm::FsmList<bluetooth::BluetoothState>::dispatch(
      bluetooth::events::internal::Gap{.type = event, .param = param});
 }
 auto avrcp_cb(esp_avrc_ct_cb_event_t event, esp_avrc_ct_cb_param_t* param)
    -> void {
  tinyfsm::FsmList<bluetooth::BluetoothState>::dispatch(
      bluetooth::events::internal::Avrc{.type = event, .param = param});
 }
 auto a2dp_cb(esp_a2d_cb_event_t event, esp_a2d_cb_param_t* param) -> void {
  tinyfsm::FsmList<bluetooth::BluetoothState>::dispatch(
      bluetooth::events::internal::A2dp{.type = event, .param = param});
 }
 auto a2dp_data_cb(uint8_t* buf, int32_t buf_size) -> int32_t {
  if (buf == nullptr || buf_size <= 0) {
    return 0;
  }
  StreamBufferHandle_t stream = sStream.load();
  if (stream == nullptr) {
    return 0;
  }
  return xStreamBufferReceive(stream, buf, buf_size, 0);
 }
 Bluetooth::Bluetooth() {
  tinyfsm::FsmList<bluetooth::BluetoothState>::start();
 }
 auto Bluetooth::Enable() -> bool {
  tinyfsm::FsmList<bluetooth::BluetoothState>::dispatch(
      bluetooth::events::Enable{});
  return !bluetooth::BluetoothState::is_in_state<bluetooth::Disabled>();
 }
 auto Bluetooth::Disable() -> void {
  tinyfsm::FsmList<bluetooth::BluetoothState>::dispatch(
      bluetooth::events::Disable{});
 }
 auto DeviceName() -> std::string {
  uint8_t mac[8]{0};
  esp_efuse_mac_get_default(mac);
  std::ostringstream name;
  name << "TANGARA " << std::hex << mac[0] << mac[1];
  return name.str();
 }
 namespace bluetooth {
 static bool sIsFirstEntry = true;
 void Disabled::entry() {
  if (sIsFirstEntry) {
    // We only use BT Classic, to claw back ~60KiB from the BLE firmware.
    esp_bt_controller_mem_release(ESP_BT_MODE_BLE);
    sIsFirstEntry = false;
    return;
  }
  esp_bluedroid_disable();
  esp_bluedroid_deinit();
  esp_bt_controller_disable();
 }
 void Disabled::react(const events::Enable&) {
  esp_bt_controller_config_t config = BT_CONTROLLER_INIT_CONFIG_DEFAULT();
  if (esp_bt_controller_init(&config) != ESP_OK) {
    ESP_LOGE(kTag, "initialize controller failed");
    return;
  }
  if (esp_bt_controller_enable(ESP_BT_MODE_CLASSIC_BT) != ESP_OK) {
    ESP_LOGE(kTag, "enable controller failed");
    return;
  }
  if (esp_bluedroid_init() != ESP_OK) {
    ESP_LOGE(kTag, "initialize bluedroid failed");
    return;
  }
  if (esp_bluedroid_enable() != ESP_OK) {
    ESP_LOGE(kTag, "enable bluedroid failed");
    return;
  }
  // Enable Secure Simple Pairing
  esp_bt_sp_param_t param_type = ESP_BT_SP_IOCAP_MODE;
  esp_bt_io_cap_t iocap = ESP_BT_IO_CAP_IO;
  esp_bt_gap_set_security_param(param_type, &iocap, sizeof(uint8_t));
  // Set a reasonable name for the device.
  std::string name = DeviceName();
  esp_bt_dev_set_device_name(name.c_str());
  // Initialise GAP. This controls advertising our device, and scanning for
  // other devices.
  esp_bt_gap_register_callback(gap_cb);
  // Initialise AVRCP. This handles playback controls; play/pause/volume/etc.
  // esp_avrc_ct_init();
  // esp_avrc_ct_register_callback(avrcp_cb);
  // Initialise A2DP. This handles streaming audio. Currently ESP-IDF's SBC
  // encoder only supports 2 channels of interleaved 16 bit samples, at
  // 44.1kHz, so there is no additional configuration to be done for the
  // stream itself.
  esp_a2d_source_init();
  esp_a2d_register_callback(a2dp_cb);
  esp_a2d_source_register_data_callback(a2dp_data_cb);
  // Don't let anyone interact with us before we're ready.
  esp_bt_gap_set_scan_mode(ESP_BT_NON_CONNECTABLE, ESP_BT_NON_DISCOVERABLE);
  transit<Scanning>();
 }
 static constexpr uint8_t kDiscoveryTimeSeconds = 10;
 static constexpr uint8_t kDiscoveryMaxResults = 0;
 void Scanning::entry() {
  ESP_LOGI(kTag, "scanning for devices");
  esp_bt_gap_start_discovery(ESP_BT_INQ_MODE_GENERAL_INQUIRY,
                             kDiscoveryTimeSeconds, kDiscoveryMaxResults);
 }
 void Scanning::exit() {
  esp_bt_gap_cancel_discovery();
 }
 auto OnDeviceDiscovered(esp_bt_gap_cb_param_t* param) -> void {
  ESP_LOGI(kTag, "device discovered");
 }
 void Scanning::react(const events::internal::Gap& ev) {
  switch (ev.type) {
    case ESP_BT_GAP_DISC_RES_EVT:
      OnDeviceDiscovered(ev.param);
      break;
    case ESP_BT_GAP_DISC_STATE_CHANGED_EVT:
      if (ev.param->disc_st_chg.state == ESP_BT_GAP_DISCOVERY_STOPPED) {
        ESP_LOGI(kTag, "still scanning");
        esp_bt_gap_start_discovery(ESP_BT_INQ_MODE_GENERAL_INQUIRY,
                                   kDiscoveryTimeSeconds, kDiscoveryMaxResults);
      }
      break;
    case ESP_BT_GAP_MODE_CHG_EVT:
      // todo: mode change. is this important?
      ESP_LOGI(kTag, "GAP mode changed");
      break;
    default:
      ESP_LOGW(kTag, "unhandled GAP event: %u", ev.type);
  }
 }
 void Connecting::entry() {
  ESP_LOGI(kTag, "connecting to device");
  esp_a2d_source_connect(nullptr);
 }
 void Connecting::exit() {}
 void Connecting::react(const events::internal::Gap& ev) {
  switch (ev.type) {
    case ESP_BT_GAP_AUTH_CMPL_EVT:
      // todo: auth completed. check if we succeeded.
      break;
    case ESP_BT_GAP_PIN_REQ_EVT:
      // todo: device needs a pin to connect.
      break;
    case ESP_BT_GAP_CFM_REQ_EVT:
      // todo: device needs user to click okay.
      break;
    case ESP_BT_GAP_KEY_NOTIF_EVT:
      // todo: device is telling us a password?
      break;
    case ESP_BT_GAP_KEY_REQ_EVT:
      // todo: device needs a password
      break;
    case ESP_BT_GAP_MODE_CHG_EVT:
      // todo: mode change. is this important?
      break;
    default:
      ESP_LOGW(kTag, "unhandled GAP event: %u", ev.type);
  }
 }
 void Connecting::react(const events::internal::A2dp& ev) {
  switch (ev.type) {
    case ESP_A2D_CONNECTION_STATE_EVT:
      // todo: connection state changed. we might be connected!
      break;
    default:
      ESP_LOGW(kTag, "unhandled A2DP event: %u", ev.type);
  }
 }
 void Connected::react(const events::internal::A2dp& ev) {
  switch (ev.type) {
    case ESP_A2D_CONNECTION_STATE_EVT:
      // todo: connection state changed. we might have dropped
      break;
    case ESP_A2D_AUDIO_STATE_EVT:
      // todo: audio state changed. who knows, dude.
      break;
    default:
      ESP_LOGW(kTag, "unhandled A2DP event: %u", ev.type);
  }
 }
 void Connected::react(const events::internal::Avrc& ev) {
  switch (ev.type) {
    case ESP_AVRC_CT_CONNECTION_STATE_EVT:
      // todo: avrc connected. send our capabilities.
    default:
      ESP_LOGW(kTag, "unhandled AVRC event: %u", ev.type);
  }
 }
 }  // namespace bluetooth
 }  // namespace drivers
 FSM_INITIAL_STATE(drivers::bluetooth::BluetoothState,
                  drivers::bluetooth::Disabled)
--- a/src/drivers/i2s_dac.cpp
+++ b/src/drivers/i2s_dac.cpp
@ -161,7 +161,6 @@ auto I2SDac::Reconfigure(Channels ch, BitsPerSample bps, SampleRate rate)
      word_length = 0b10;
      break;
    case BPS_32:
      // TODO(jacqueline): Error on this? It's not supported anymore.
      slot_config_.data_bit_width = I2S_DATA_BIT_WIDTH_32BIT;
      slot_config_.ws_width = 32;
      word_length = 0b11;
--- a/src/drivers/include/bluetooth.hpp
+++ b/src/drivers/include/bluetooth.hpp
@ -0,0 +1,108 @@
 #pragma once
 #include <string>
 #include <vector>
 #include <freertos/FreeRTOS.h>
 #include <freertos/stream_buffer.h>
 #include "esp_a2dp_api.h"
 #include "esp_avrc_api.h"
 #include "esp_gap_bt_api.h"
 #include "tinyfsm.hpp"
 #include "tinyfsm/include/tinyfsm.hpp"
 namespace drivers {
 /*
 * A handle used to interact with the bluetooth state machine.
 */
 class Bluetooth {
 public:
  Bluetooth();
  auto Enable() -> bool;
  auto Disable() -> void;
  auto SetSource(StreamBufferHandle_t) -> void;
 };
 namespace bluetooth {
 namespace events {
 struct Enable : public tinyfsm::Event {};
 struct Disable : public tinyfsm::Event {};
 namespace internal {
 struct Gap : public tinyfsm::Event {
  esp_bt_gap_cb_event_t type;
  esp_bt_gap_cb_param_t* param;
 };
 struct A2dp : public tinyfsm::Event {
  esp_a2d_cb_event_t type;
  esp_a2d_cb_param_t* param;
 };
 struct Avrc : public tinyfsm::Event {
  esp_avrc_ct_cb_event_t type;
  esp_avrc_ct_cb_param_t* param;
 };
 }  // namespace internal
 }  // namespace events
 class BluetoothState : public tinyfsm::Fsm<BluetoothState> {
 public:
  virtual ~BluetoothState(){};
  virtual void entry() {}
  virtual void exit() {}
  virtual void react(const events::Enable& ev){};
  virtual void react(const events::Disable& ev) = 0;
  virtual void react(const events::internal::Gap& ev) = 0;
  virtual void react(const events::internal::A2dp& ev) = 0;
  virtual void react(const events::internal::Avrc& ev){};
 };
 class Disabled : public BluetoothState {
  void entry() override;
  void react(const events::Enable& ev) override;
  void react(const events::Disable& ev) override{};
  void react(const events::internal::Gap& ev) override {}
  void react(const events::internal::A2dp& ev) override {}
 };
 class Scanning : public BluetoothState {
  void entry() override;
  void exit() override;
  void react(const events::Disable& ev) override;
  void react(const events::internal::Gap& ev) override;
  void react(const events::internal::A2dp& ev) override;
 };
 class Connecting : public BluetoothState {
  void entry() override;
  void exit() override;
  void react(const events::Disable& ev) override;
  void react(const events::internal::Gap& ev) override;
  void react(const events::internal::A2dp& ev) override;
 };
 class Connected : public BluetoothState {
  void entry() override;
  void exit() override;
  void react(const events::Disable& ev) override;
  void react(const events::internal::Gap& ev) override;
  void react(const events::internal::A2dp& ev) override;
  void react(const events::internal::Avrc& ev) override;
 };
 }  // namespace bluetooth
 }  // namespace drivers
--- a/src/drivers/include/i2s_dac.hpp
+++ b/src/drivers/include/i2s_dac.hpp
@ -51,14 +51,12 @@ class I2SDac {
    BPS_32 = I2S_DATA_BIT_WIDTH_32BIT,
  };
  enum SampleRate {
-    SAMPLE_RATE_11_025 = 11025,
+    SAMPLE_RATE_8 = 8000,
    SAMPLE_RATE_16 = 16000,
    SAMPLE_RATE_22_05 = 22050,
    SAMPLE_RATE_32 = 32000,
    SAMPLE_RATE_44_1 = 44100,
    SAMPLE_RATE_48 = 48000,
    SAMPLE_RATE_88_2 = 88200,
    SAMPLE_RATE_96 = 96000,
    SAMPLE_RATE_192 = 192000,
  };
  auto Reconfigure(Channels ch, BitsPerSample bps, SampleRate rate) -> void;
--- a/src/drivers/include/nvs.hpp
+++ b/src/drivers/include/nvs.hpp
@ -0,0 +1,27 @@
 /*
 * Copyright 2023 jacqueline <me@jacqueline.id.au>
 *
 * SPDX-License-Identifier: GPL-3.0-only
 */
 #pragma once
 #include "esp_err.h"
 #include "nvs.h"
 namespace drivers {
 class NvsStorage {
 public:
  static auto Open() -> NvsStorage*;
  auto SchemaVersion() -> uint8_t;
  explicit NvsStorage(nvs_handle_t);
  ~NvsStorage();
 private:
  nvs_handle_t handle_;
 };
 }  // namespace drivers
--- a/src/drivers/nvs.cpp
+++ b/src/drivers/nvs.cpp
@ -0,0 +1,73 @@
 /*
 * Copyright 2023 jacqueline <me@jacqueline.id.au>
 *
 * SPDX-License-Identifier: GPL-3.0-only
 */
 #include "nvs.hpp"
 #include <stdint.h>
 #include <cstdint>
 #include <memory>
 #include "esp_log.h"
 #include "nvs.h"
 #include "nvs_flash.h"
 namespace drivers {
 static constexpr char kTag[] = "nvm";
 static constexpr uint8_t kSchemaVersion = 1;
 static constexpr char kKeyVersion[] = "ver";
 auto NvsStorage::Open() -> NvsStorage* {
  esp_err_t err = nvs_flash_init();
  if (err == ESP_ERR_NVS_NO_FREE_PAGES) {
    ESP_LOGW(kTag, "partition needs initialisation");
    nvs_flash_erase();
    err = nvs_flash_init();
  }
  if (err != ESP_OK) {
    ESP_LOGE(kTag, "failed to init nvm");
    return nullptr;
  }
  nvs_handle_t handle;
  if ((err = nvs_open("tangara", NVS_READWRITE, &handle)) != ESP_OK) {
    ESP_LOGE(kTag, "failed to open nvs namespace");
    return nullptr;
  }
  std::unique_ptr<NvsStorage> instance = std::make_unique<NvsStorage>(handle);
  if (instance->SchemaVersion() < kSchemaVersion) {
    ESP_LOGW(kTag, "namespace needs downgrading");
    nvs_erase_all(handle);
    nvs_set_u8(handle, kKeyVersion, kSchemaVersion);
    err = nvs_commit(handle);
    if (err != ESP_OK) {
      ESP_LOGW(kTag, "failed to init namespace");
      return nullptr;
    }
  }
  ESP_LOGI(kTag, "nvm storage initialised okay");
  return instance.release();
 }
 NvsStorage::NvsStorage(nvs_handle_t handle) : handle_(handle) {}
 NvsStorage::~NvsStorage() {
  nvs_close(handle_);
  nvs_flash_deinit();
 }
 auto NvsStorage::SchemaVersion() -> uint8_t {
  uint8_t ret;
  if (nvs_get_u8(handle_, kKeyVersion, &ret) != ESP_OK) {
    return UINT8_MAX;
  }
  return ret;
 }
 }  // namespace drivers
--- a/src/drivers/storage.cpp
+++ b/src/drivers/storage.cpp
@ -63,7 +63,7 @@ auto SdStorage::Create(IGpios* gpio) -> cpp::result<SdStorage*, Error> {
  // Will return ESP_ERR_INVALID_RESPONSE if there is no card
  esp_err_t err = sdmmc_card_init(host.get(), card.get());
  if (err != ESP_OK) {
-    ESP_LOGW(kTag, "Failed to read, err: %d", err);
+    ESP_LOGW(kTag, "Failed to read, err: %s", esp_err_to_name(err));
    return cpp::fail(Error::FAILED_TO_READ);
  }
@ -74,7 +74,21 @@ auto SdStorage::Create(IGpios* gpio) -> cpp::result<SdStorage*, Error> {
  // Mount right now, not on first operation.
  FRESULT ferr = f_mount(fs, "", 1);
  if (ferr != FR_OK) {
-    ESP_LOGW(kTag, "Failed to mount, err: %d", ferr);
+    std::string err_str;
    switch (ferr) {
      case FR_DISK_ERR:
        err_str = "FR_DISK_ERR";
        break;
      case FR_NOT_READY:
        err_str = "FR_NOT_READY";
        break;
      case FR_NO_FILESYSTEM:
        err_str = "FR_NO_FILESYSTEM";
        break;
      default:
        err_str = std::to_string(ferr);
    }
    ESP_LOGW(kTag, "Failed to mount, err: %s", err_str.c_str());
    return cpp::fail(Error::FAILED_TO_MOUNT);
  }
--- a/src/system_fsm/booting.cpp
+++ b/src/system_fsm/booting.cpp
@ -13,6 +13,7 @@
 #include "event_queue.hpp"
 #include "gpios.hpp"
 #include "lvgl/lvgl.h"
 #include "nvs.hpp"
 #include "relative_wheel.hpp"
 #include "spi.hpp"
 #include "system_events.hpp"
@ -50,9 +51,10 @@ auto Booting::entry() -> void {
  ESP_LOGI(kTag, "installing remaining drivers");
  sSamd.reset(drivers::Samd::Create());
  sBattery.reset(drivers::Battery::Create());
  sNvs.reset(drivers::NvsStorage::Open());
  sTagParser.reset(new database::TagParserImpl());
-  if (!sSamd || !sBattery) {
+  if (!sSamd || !sBattery || !sNvs) {
    events::System().Dispatch(FatalError{});
    events::Ui().Dispatch(FatalError{});
    return;
--- a/src/system_fsm/include/system_fsm.hpp
+++ b/src/system_fsm/include/system_fsm.hpp
@ -13,6 +13,7 @@
 #include "database.hpp"
 #include "display.hpp"
 #include "gpios.hpp"
 #include "nvs.hpp"
 #include "relative_wheel.hpp"
 #include "samd.hpp"
 #include "storage.hpp"
@ -54,6 +55,7 @@ class SystemState : public tinyfsm::Fsm<SystemState> {
 protected:
  static std::shared_ptr<drivers::Gpios> sGpios;
  static std::shared_ptr<drivers::Samd> sSamd;
  static std::shared_ptr<drivers::NvsStorage> sNvs;
  static std::shared_ptr<drivers::TouchWheel> sTouch;
  static std::shared_ptr<drivers::RelativeWheel> sRelativeTouch;
--- a/src/system_fsm/system_fsm.cpp
+++ b/src/system_fsm/system_fsm.cpp
@ -17,6 +17,7 @@ namespace system_fsm {
 std::shared_ptr<drivers::Gpios> SystemState::sGpios;
 std::shared_ptr<drivers::Samd> SystemState::sSamd;
 std::shared_ptr<drivers::NvsStorage> SystemState::sNvs;
 std::shared_ptr<drivers::TouchWheel> SystemState::sTouch;
 std::shared_ptr<drivers::RelativeWheel> SystemState::sRelativeTouch;
--- a/src/tasks/tasks.cpp
+++ b/src/tasks/tasks.cpp
@ -34,6 +34,10 @@ auto Name<Type::kAudio>() -> std::string {
  return "AUDIO";
 }
 template <>
 auto Name<Type::kMixer>() -> std::string {
  return "MIXER";
 }
 template <>
 auto Name<Type::kDatabase>() -> std::string {
  return "DB";
 }
@ -77,6 +81,14 @@ auto AllocateStack<Type::kFileStreamer>() -> cpp::span<StackType_t> {
          size};
 }
 template <>
 auto AllocateStack<Type::kMixer>() -> cpp::span<StackType_t> {
  std::size_t size = 4 * 1024;
  return {static_cast<StackType_t*>(
              heap_caps_malloc(size, MALLOC_CAP_INTERNAL | MALLOC_CAP_8BIT)),
          size};
 }
 // Leveldb is designed for non-embedded use cases, where stack space isn't so
 // much of a concern. It therefore uses an eye-wateringly large amount of stack.
 template <>
@ -105,6 +117,10 @@ auto Priority() -> UBaseType_t;
 // Realtime audio is the entire point of this device, so give this task the
 // highest priority.
 template <>
 auto Priority<Type::kMixer>() -> UBaseType_t {
  return 12;
 }
 template <>
 auto Priority<Type::kAudio>() -> UBaseType_t {
  return 11;
 }
--- a/src/tasks/tasks.hpp
+++ b/src/tasks/tasks.hpp
@ -36,6 +36,8 @@ enum class Type {
  kFileStreamer,
  // The main audio pipeline task.
  kAudio,
  // TODO
  kMixer,
  // Task for running database queries.
  kDatabase,
  // Task for internal database operations
@ -57,9 +59,19 @@ template <Type t>
 auto StartPersistent(const std::function<void(void)>& fn) -> void {
  StaticTask_t* task_buffer = new StaticTask_t;
  cpp::span<StackType_t> stack = AllocateStack<t>();
  xTaskCreateStatic(&PersistentMain, Name<t>().c_str(), stack.size(),
                    new std::function<void(void)>(fn), Priority<t>(),
                    stack.data(), task_buffer);
 }
 template <Type t>
 auto StartPersistent(BaseType_t core, const std::function<void(void)>& fn)
    -> void {
  StaticTask_t* task_buffer = new StaticTask_t;
  cpp::span<StackType_t> stack = AllocateStack<t>();
  xTaskCreateStaticPinnedToCore(&PersistentMain, Name<t>().c_str(),
                                stack.size(), new std::function<void(void)>(fn),
-                                Priority<t>(), stack.data(), task_buffer, 0);
+                                Priority<t>(), stack.data(), task_buffer, core);
 }
 class Worker {
--- a/src/ui/lvgl_task.cpp
+++ b/src/ui/lvgl_task.cpp
@ -48,12 +48,25 @@
 namespace ui {
 static const char* kTag = "lv_task";
 static const TickType_t kMaxFrameRate = pdMS_TO_TICKS(33);
 static int sTimerId;
 static SemaphoreHandle_t sFrameSemaphore;
 auto next_frame(TimerHandle_t) {
  xSemaphoreGive(sFrameSemaphore);
 }
 void LvglMain(std::weak_ptr<drivers::RelativeWheel> weak_touch_wheel,
              std::weak_ptr<drivers::Display> weak_display) {
  ESP_LOGI(kTag, "init lvgl");
  lv_init();
  sFrameSemaphore = xSemaphoreCreateBinary();
  auto timer =
      xTimerCreate("lvgl_frame", kMaxFrameRate, pdTRUE, &sTimerId, next_frame);
  xTimerStart(timer, portMAX_DELAY);
  lv_theme_t* base_theme = lv_theme_basic_init(NULL);
  lv_disp_set_theme(NULL, base_theme);
  static themes::Theme sTheme{};
@ -80,9 +93,9 @@ void LvglMain(std::weak_ptr<drivers::RelativeWheel> weak_touch_wheel,
    }
    lv_task_handler();
-    // 30 FPS
+
-    // TODO(jacqueline): make this dynamic
+    // Wait for the signal to loop again.
-    vTaskDelay(pdMS_TO_TICKS(33));
+    xSemaphoreTake(sFrameSemaphore, portMAX_DELAY);
  }
 }
--- a/tools/cmake/common.cmake
+++ b/tools/cmake/common.cmake
@ -20,7 +20,6 @@ list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/lvgl")
 list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/result")
 list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/shared_string")
 list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/span")
 list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/stb_vorbis")
 list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/tinyfsm")
 include($ENV{IDF_PATH}/tools/cmake/project.cmake)