From 76f06a045a019bdf9623c492ebf6252ac03397b7 Mon Sep 17 00:00:00 2001
From: Dirk Helbig <42479019+hegdi@users.noreply.github.com>
Date: Fri, 28 Feb 2025 09:42:11 +0100
Subject: [PATCH] platform/linux: add ALSA support

---
 CHANGELOG.md                        |   2 +
 platform/linux/btstack_audio_alsa.c | 423 ++++++++++++++++++++++++++++
 port/linux/CMakeLists.txt           |  23 +-
 port/linux/main.c                   |   4 +-
 src/btstack_audio.h                 |   2 +
 5 files changed, 446 insertions(+), 8 deletions(-)
 create mode 100644 platform/linux/btstack_audio_alsa.c

diff --git a/CHANGELOG.md b/CHANGELOG.md
index 33d5d1fbf..0d1aeb5f8 100644
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -15,6 +15,8 @@ and this project adheres to [Semantic Versioning](http://semver.org/spec/v2.0.0.
 - HCI: support newer AIROC Controller that require Download Mode with ENABLE_AIROC_DOWNLOAD_MODE
 - Zephyr: provide hal_flash_bank implementation for native flash driver
 - POSIX: support error condition for file descriptors in btstack_run_loop
+- Linux: HCI Transport for Linux HCI Kernel Socket
+- Linux: Audio sink implementation for Linux ALSA
 
 ### Fixed
 - GAP: store link key for standard/non-SSP pairing
diff --git a/platform/linux/btstack_audio_alsa.c b/platform/linux/btstack_audio_alsa.c
new file mode 100644
index 000000000..a926842ea
--- /dev/null
+++ b/platform/linux/btstack_audio_alsa.c
@@ -0,0 +1,423 @@
+/*
+ * Copyright (C) 2025 BlueKitchen GmbH
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of the copyright holders nor the names of
+ *    contributors may be used to endorse or promote products derived
+ *    from this software without specific prior written permission.
+ * 4. Any redistribution, use, or modification is done solely for
+ *    personal benefit and not for any commercial purpose or for
+ *    monetary gain.
+ *
+ * THIS SOFTWARE IS PROVIDED BY BLUEKITCHEN GMBH AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ * FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL BLUEKITCHEN
+ * GMBH OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
+ * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS
+ * OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
+ * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF
+ * THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * Please inquire about commercial licensing options at
+ * contact@bluekitchen-gmbh.com
+ *
+ */
+#ifdef HAVE_ALSA
+#define BTSTACK_FILE__ "btstack_audio_alsa.c"
+
+#include <btstack_run_loop.h>
+#include <alsa/asoundlib.h>
+#include <stdio.h>
+#include <stdint.h>
+#include <string.h>
+#include <poll.h>
+
+#include "btstack_debug.h"
+#include "btstack_audio.h"
+
+static const char* device = "default";
+static const char* simple_mixer_name = "Master";
+static snd_pcm_t *pcm_handle = NULL;
+static snd_mixer_t *mixer_handle = NULL;
+static snd_mixer_elem_t* master_volume = NULL;
+static long volume_max;
+static uint32_t current_sample_rate;
+static uint8_t num_channels;
+static void (*playback_callback)(int16_t *buffer, uint16_t num_samples);
+
+static unsigned int buffer_time = 500000;   /* ring buffer length in us */
+static unsigned int period_time = 100000;   /* period time in us */
+static int resample = 1;                    /* enable alsa-lib resampling */
+static snd_output_t *output = NULL;
+static snd_pcm_sframes_t buffer_size;
+static snd_pcm_sframes_t period_size;
+static int period_event = 0;                /* produce poll event after each period */
+
+static struct pollfd *ufds = NULL;
+static int ufds_count;
+// TODO: corner case handling
+#if 0
+/*
+ *   Underrun and suspend recovery
+ */
+static int xrun_recovery(snd_pcm_t *handle, int err)
+{
+    printf("stream recovery\n");
+    if (err == -EPIPE) {    /* under-run */
+        err = snd_pcm_prepare(handle);
+        if (err < 0)
+            printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err));
+        return 0;
+    } else if (err == -ESTRPIPE) {
+        while ((err = snd_pcm_resume(handle)) == -EAGAIN)
+            sleep(1);   /* wait until the suspend flag is released */
+        if (err < 0) {
+            err = snd_pcm_prepare(handle);
+            if (err < 0)
+                printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err));
+        }
+        return 0;
+    }
+    return err;
+}
+#endif
+
+static btstack_data_source_t *btstack_audio_alsa_data_sources;
+
+static void btstack_audio_alsa_handler(btstack_data_source_t * ds, btstack_data_source_callback_type_t callback_type) {
+    int source_fd = ds->source.fd;
+    int err;
+
+    for(int i=0; i<ufds_count; ++i ) {
+        int fd = ufds[i].fd;
+        short revents = 0;
+        if( source_fd == fd ) {
+            if( callback_type == DATA_SOURCE_CALLBACK_WRITE ) {
+                revents |= POLLOUT;
+            } else if( callback_type == DATA_SOURCE_CALLBACK_READ ) {
+                revents |= POLLIN;
+            }
+        }
+        ufds[i].revents = revents;
+    }
+    unsigned short revents;
+    err = snd_pcm_poll_descriptors_revents( pcm_handle, ufds, ufds_count, &revents );
+    if( err < 0 ) {
+        fprintf(stderr, "Can't translate event poll results: %s\n", snd_strerror(err));
+        return;
+    }
+
+    if( ( revents & POLLOUT ) != POLLOUT ) {
+        return;
+    }
+
+    int16_t buffer[1024] = { 0 };
+    playback_callback( buffer, 512 );
+    err = snd_pcm_writei( pcm_handle, buffer, 512 );
+    if (err < 512) {
+        fprintf(stderr, "Write error: %s\n", snd_strerror(err));
+        return;
+    }
+}
+
+static int alsa_mixer_init() {
+    long min;
+    int err;
+    snd_mixer_selem_id_t *sid;
+
+    err = snd_mixer_open(&mixer_handle, 0);
+     if( err < 0 ) {
+        fprintf(stderr, "Cannot open mixer device: %s\n", snd_strerror(err));
+        return err;
+    }
+    err = snd_mixer_attach(mixer_handle, device);
+    if( err < 0 ) {
+        fprintf(stderr, "Cannot attach mixer to device %s: %s\n", device, snd_strerror(err));
+        return err;
+    }
+    err = snd_mixer_selem_register(mixer_handle, NULL, NULL);
+      if( err < 0 ) {
+        fprintf(stderr, "Can't register simple mixer device: %s\n", snd_strerror(err));
+        return err;
+    }
+    err = snd_mixer_load(mixer_handle);
+    if( err < 0 ) {
+        fprintf(stderr, "Failed to load mixer: %s\n", snd_strerror(err));
+        return err;
+    }
+    snd_mixer_selem_id_alloca(&sid);
+    if( sid == NULL ) {
+        fprintf(stderr, "Failed to allocate simple mixer ID: %s\n", snd_strerror(err));
+        return err;
+    }
+
+    snd_mixer_selem_id_set_index(sid, 0);
+    snd_mixer_selem_id_set_name(sid, simple_mixer_name);
+    master_volume = snd_mixer_find_selem(mixer_handle, sid);
+    if( master_volume == NULL ) {
+        fprintf(stderr, "Master volume not found: %s\n", snd_strerror(err));
+        return err;
+    }
+    err = snd_mixer_selem_get_playback_volume_range(master_volume, &min, &volume_max);
+    if( err < 0 ) {
+        fprintf(stderr, "Failed to query mixer range: %s\n", snd_strerror(err));
+        return err;
+    }
+    return 0;
+}
+
+static int alsa_init(uint8_t channels, uint32_t samplerate, void (*playback)(int16_t *, uint16_t)) {
+    int err, dir;
+    snd_pcm_hw_params_t *hw_params;
+    snd_pcm_sw_params_t *sw_params;
+    unsigned int rrate;
+    snd_pcm_uframes_t size;
+
+    err = snd_output_stdio_attach(&output, stdout, 0);
+    if (err < 0) {
+        fprintf(stderr, "Output failed: %s\n", snd_strerror(err));
+        return err;
+    }
+
+    err = snd_pcm_open(&pcm_handle, device, SND_PCM_STREAM_PLAYBACK, 0);
+    if(err < 0) {
+        fprintf(stderr, "Cannot open audio device: %s\n", snd_strerror(err));
+        return err;
+    }
+
+    snd_pcm_hw_params_alloca(&hw_params);
+    if( hw_params == NULL ) {
+       fprintf(stderr, "Failed to allocate simple mixer ID: %s\n", snd_strerror(err));
+       return err;
+    }
+    /* choose all parameters */
+    err = snd_pcm_hw_params_any(pcm_handle, hw_params);
+    if (err < 0) {
+        fprintf(stderr, "Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* set hardware resampling */
+    err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hw_params, resample);
+    if (err < 0) {
+        fprintf(stderr, "Resampling setup failed for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* set the interleaved read/write format */
+    err = snd_pcm_hw_params_set_access(pcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
+    if (err < 0) {
+        fprintf(stderr, "Access type not available for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* set the sample format */
+    err = snd_pcm_hw_params_set_format(pcm_handle, hw_params, SND_PCM_FORMAT_S16_LE);
+    if (err < 0) {
+        fprintf(stderr, "Sample format not available for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* set the count of channels */
+    err = snd_pcm_hw_params_set_channels(pcm_handle, hw_params, channels);
+    if (err < 0) {
+        fprintf(stderr, "Channels count (%u) not available for playbacks: %s\n", channels, snd_strerror(err));
+        return err;
+    }
+    /* set the stream rate */
+    rrate = samplerate;
+    err = snd_pcm_hw_params_set_rate_near(pcm_handle, hw_params, &rrate, 0);
+    if (err < 0) {
+        fprintf(stderr, "Rate %uHz not available for playback: %s\n", samplerate, snd_strerror(err));
+        return err;
+    }
+    if (rrate != samplerate) {
+        fprintf(stderr, "Rate doesn't match (requested %uHz, get %iHz)\n", samplerate, err);
+        return -EINVAL;
+    }
+    /* set the buffer time */
+    err = snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hw_params, &buffer_time, &dir);
+    if (err < 0) {
+        fprintf(stderr, "Unable to set buffer time %u for playback: %s\n", buffer_time, snd_strerror(err));
+        return err;
+    }
+    err = snd_pcm_hw_params_get_buffer_size(hw_params, &size);
+    if (err < 0) {
+        fprintf(stderr, "Unable to get buffer size for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    buffer_size = size;
+    /* set the period time */
+    err = snd_pcm_hw_params_set_period_time_near(pcm_handle, hw_params, &period_time, &dir);
+    if (err < 0) {
+        fprintf(stderr, "Unable to set period time %u for playback: %s\n", period_time, snd_strerror(err));
+        return err;
+    }
+    err = snd_pcm_hw_params_get_period_size(hw_params, &size, &dir);
+    if (err < 0) {
+        fprintf(stderr, "Unable to get period size for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    period_size = size;
+
+    err = snd_pcm_hw_params(pcm_handle, hw_params);
+    if (err < 0) {
+        fprintf(stderr, "Cannot set hardware parameters: %s\n", snd_strerror(err));
+        return err;
+    }
+
+    snd_pcm_sw_params_alloca(&sw_params);
+
+    /* get the current sw params */
+    err = snd_pcm_sw_params_current(pcm_handle, sw_params);
+    if (err < 0) {
+        fprintf(stderr, "Unable to determine current swparams for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* start the transfer when the buffer is almost full: */
+    /* (buffer_size / avail_min) * avail_min */
+    err = snd_pcm_sw_params_set_start_threshold(pcm_handle, sw_params, (buffer_size / period_size) * period_size);
+    if (err < 0) {
+        fprintf(stderr, "Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* allow the transfer when at least period_size samples can be processed */
+    /* or disable this mechanism when period event is enabled (aka interrupt like style processing) */
+    err = snd_pcm_sw_params_set_avail_min(pcm_handle, sw_params, period_event ? buffer_size : period_size);
+    if (err < 0) {
+        fprintf(stderr, "Unable to set avail min for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* enable period events when requested */
+    if (period_event) {
+        err = snd_pcm_sw_params_set_period_event(pcm_handle, sw_params, 1);
+        if (err < 0) {
+            fprintf(stderr, "Unable to set period event: %s\n", snd_strerror(err));
+            return err;
+        }
+    }
+    /* write the parameters to the playback device */
+    err = snd_pcm_sw_params(pcm_handle, sw_params);
+    if (err < 0) {
+        fprintf(stderr, "Unable to set sw params for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+
+    current_sample_rate = samplerate;
+    num_channels = channels;
+    playback_callback = playback;
+
+    ufds_count = snd_pcm_poll_descriptors_count (pcm_handle);
+    if (ufds_count <= 0) {
+        fprintf(stderr, "Invalid poll descriptors count\n");
+        return ufds_count;
+    }
+
+    ufds = malloc(sizeof(struct pollfd) * ufds_count);
+    if (ufds == NULL) {
+        fprintf(stderr, "No enough memory for pollfd\n");
+        return -ENOMEM;
+    }
+
+    btstack_audio_alsa_data_sources = malloc(sizeof(btstack_data_source_t) * ufds_count);
+    if (ufds == NULL) {
+        fprintf(stderr, "No enough memory for data sources\n");
+        return -ENOMEM;
+    }
+
+    err = snd_pcm_poll_descriptors(pcm_handle, ufds, ufds_count);
+    if(err < 0) {
+        fprintf(stderr, "Unable to obtain poll descriptors for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+
+    // poll -> data sources
+    for(int i=0; i<ufds_count; ++i) {
+        int flags = 0;
+        int fd = ufds[i].fd;
+        short events = ufds[i].events;
+        ufds[i].revents = 0;
+        btstack_data_source_t *data_source = &btstack_audio_alsa_data_sources[i];
+        if( (events & POLLOUT) == POLLOUT ) {
+            flags |= DATA_SOURCE_CALLBACK_WRITE;
+        }
+        if( (events & POLLIN) == POLLIN ) {
+            flags |= DATA_SOURCE_CALLBACK_READ;
+        }
+        btstack_run_loop_set_data_source_fd(data_source, fd);
+        btstack_run_loop_set_data_source_handler(data_source, &btstack_audio_alsa_handler);
+        btstack_run_loop_add_data_source(data_source);
+        btstack_run_loop_enable_data_source_callbacks(data_source, flags);
+    }
+
+    // TODO: remove once pipe error handling / priming is implemented
+    int16_t samples[48000] = { 0 };
+    printf("period size: %ld\n", period_size );
+    snd_pcm_writei( pcm_handle, samples, period_size );
+
+    alsa_mixer_init();
+    return 1;
+}
+
+static uint32_t alsa_get_samplerate(void) {
+    return current_sample_rate;
+}
+
+static void alsa_set_volume(uint8_t volume) {
+    int err;
+
+    btstack_assert( volume <= 127 );
+    if( master_volume == NULL ) {
+        return;
+    }
+    err = snd_mixer_selem_set_playback_volume_all(master_volume, volume * volume_max / 127);
+    if( err < 0 ) {
+        fprintf(stderr, "Failed to set volume: %s\n", snd_strerror(err));
+        return;
+    }
+}
+
+static void alsa_start_stream(void) {
+    snd_pcm_start(pcm_handle);
+}
+
+static void alsa_stop_stream(void) {
+    snd_pcm_drop(pcm_handle);
+}
+
+static void alsa_close(void) {
+    if( pcm_handle != NULL ) {
+        snd_pcm_close(pcm_handle);
+    }
+    if( mixer_handle != NULL ) {
+        snd_mixer_close(mixer_handle);
+    }
+    if( ufds != NULL ) {
+        free( ufds );
+    }
+    if( btstack_audio_alsa_data_sources != NULL ) {
+        free( btstack_audio_alsa_data_sources );
+    }
+}
+
+btstack_audio_sink_t btstack_audio_alsa_sink = {
+    .init = alsa_init,
+    .get_samplerate = alsa_get_samplerate,
+    .set_volume = alsa_set_volume,
+    .start_stream = alsa_start_stream,
+    .stop_stream = alsa_stop_stream,
+    .close = alsa_close,
+};
+
+const btstack_audio_sink_t * btstack_audio_alsa_sink_get_instance(void){
+    return &btstack_audio_alsa_sink;
+}
+#endif // HAVE_ALSA
diff --git a/port/linux/CMakeLists.txt b/port/linux/CMakeLists.txt
index 9dff28380..fee10b9e8 100644
--- a/port/linux/CMakeLists.txt
+++ b/port/linux/CMakeLists.txt
@@ -149,13 +149,22 @@ find_package(PkgConfig)
 # portaudio
 if (PkgConfig_FOUND)
 	pkg_check_modules(PORTAUDIO portaudio-2.0)
-	if(PORTAUDIO_FOUND)
-		message("HAVE_PORTAUDIO")
-		target_include_directories(btstack PUBLIC ${PORTAUDIO_INCLUDE_DIRS})
-		target_link_directories(btstack PUBLIC ${PORTAUDIO_LIBRARY_DIRS})
-		target_link_libraries(btstack ${PORTAUDIO_LIBRARIES})
-		add_compile_definitions(HAVE_PORTAUDIO)
-	endif()
+endif()
+
+find_package(ALSA)
+
+if(PORTAUDIO_FOUND)
+    message("PortAudio audio port selected")
+    target_include_directories(btstack PUBLIC ${PORTAUDIO_INCLUDE_DIRS})
+    target_link_directories(btstack PUBLIC ${PORTAUDIO_LIBRARY_DIRS})
+    target_link_libraries(btstack PUBLIC ${PORTAUDIO_LIBRARIES})
+    target_compile_definitions(btstack PUBLIC HAVE_PORTAUDIO)
+elseif(ALSA_FOUND)
+    message("ALSA audio port selected")
+    target_include_directories(btstack PUBLIC ${ALSA_INCLUDE_DIRS})
+    target_link_directories(btstack PUBLIC ${ALSA_LIBRARY_DIRS})
+    target_link_libraries(btstack PUBLIC ${ALSA_LIBRARIES})
+    target_compile_definitions(btstack PUBLIC HAVE_ALSA)
 endif()
 
 # pthread
diff --git a/port/linux/main.c b/port/linux/main.c
index e411aa44d..05a66140f 100644
--- a/port/linux/main.c
+++ b/port/linux/main.c
@@ -268,9 +268,11 @@ int main(int argc, const char * argv[]){
 	const hci_transport_t * transport = hci_transport_linux_instance();
 	hci_init(transport, (void*) &transport_config);
 
-#ifdef HAVE_PORTAUDIO
+#if HAVE_PORTAUDIO
     btstack_audio_sink_set_instance(btstack_audio_portaudio_sink_get_instance());
     btstack_audio_source_set_instance(btstack_audio_portaudio_source_get_instance());
+#elif HAVE_ALSA
+    btstack_audio_sink_set_instance(btstack_audio_alsa_sink_get_instance());
 #endif
 
     // inform about BTstack state
diff --git a/src/btstack_audio.h b/src/btstack_audio.h
index e19099837..6dca1cd77 100644
--- a/src/btstack_audio.h
+++ b/src/btstack_audio.h
@@ -171,6 +171,8 @@ void btstack_audio_source_set_instance(const btstack_audio_source_t * audio_sour
 const btstack_audio_sink_t *   btstack_audio_portaudio_sink_get_instance(void);
 const btstack_audio_source_t * btstack_audio_portaudio_source_get_instance(void);
 
+const btstack_audio_sink_t *   btstack_audio_alsa_sink_get_instance(void);
+
 const btstack_audio_sink_t *   btstack_audio_embedded_sink_get_instance(void);
 const btstack_audio_source_t * btstack_audio_embedded_source_get_instance(void);