audio: Copy generic HAL and bt-audio configs from Cuttlefish
Existing HAL copied from goldfish days seem to be obsolete
and didn't work as expected with both analog speakers (WIP)
and BT audio profiles. So copied generic audio HAL from
Cuttlefish and verified BT and analog audio playback.
Change-Id: Ib7fbea1c02240663e3406865f3a21544b1aec793
Signed-off-by: Amit Pundir <amit.pundir@linaro.org>
diff --git a/audio/audio_hw.c b/audio/audio_hw.c
index d601ea8..805e2cd 100644
--- a/audio/audio_hw.c
+++ b/audio/audio_hw.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2016 The Android Open Source Project
+ * Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -12,352 +12,812 @@
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
+ *
+ * Copied as it is from device/google/cuttlefish/guest/hals/audio/audio_hw.c
+ * and fixed couple of typos pointed out by Lint during review.
*/
-#define LOG_TAG "audio_hw_dragonboard"
-//#define LOG_NDEBUG 0
+#define LOG_TAG "audio_hw_generic"
+#include <assert.h>
#include <errno.h>
-#include <malloc.h>
+#include <inttypes.h>
#include <pthread.h>
#include <stdint.h>
-#include <sys/time.h>
#include <stdlib.h>
+#include <sys/time.h>
+#include <dlfcn.h>
+#include <fcntl.h>
#include <unistd.h>
#include <log/log.h>
+#include <cutils/list.h>
#include <cutils/str_parms.h>
-#include <cutils/properties.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <hardware/audio.h>
-
-#include <sound/asound.h>
#include <tinyalsa/asoundlib.h>
-#include <audio_utils/resampler.h>
-#include <audio_utils/echo_reference.h>
-#include <hardware/audio_effect.h>
-#include <hardware/audio_alsaops.h>
-#include <audio_effects/effect_aec.h>
+
+#define PCM_CARD 0
+#define PCM_DEVICE 0
-#define CARD_OUT 0
-#define PORT_CODEC 0
-/* Minimum granularity - Arbitrary but small value */
-#define CODEC_BASE_FRAME_COUNT 32
+#define OUT_PERIOD_MS 15
+#define OUT_PERIOD_COUNT 4
-/* number of base blocks in a short period (low latency) */
-#define PERIOD_MULTIPLIER 32 /* 21 ms */
-/* number of frames per short period (low latency) */
-#define PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * PERIOD_MULTIPLIER)
-/* number of pseudo periods for low latency playback */
-#define PLAYBACK_PERIOD_COUNT 2
-#define PLAYBACK_PERIOD_START_THRESHOLD 2
-#define CODEC_SAMPLING_RATE 48000
-#define CHANNEL_STEREO 2
-#define MIN_WRITE_SLEEP_US 5000
+#define IN_PERIOD_MS 15
+#define IN_PERIOD_COUNT 4
-struct stub_stream_in {
- struct audio_stream_in stream;
+struct generic_audio_device {
+ struct audio_hw_device device; // Constant after init
+ pthread_mutex_t lock;
+ bool mic_mute; // Protected by this->lock
+ struct mixer* mixer; // Protected by this->lock
+ struct listnode out_streams; // Record for output streams, protected by this->lock
+ struct listnode in_streams; // Record for input streams, protected by this->lock
+ audio_patch_handle_t next_patch_handle; // Protected by this->lock
};
-struct alsa_audio_device {
- struct audio_hw_device hw_device;
-
- pthread_mutex_t lock; /* see note below on mutex acquisition order */
- int devices;
- struct alsa_stream_in *active_input;
- struct alsa_stream_out *active_output;
- bool mic_mute;
-};
-
-struct alsa_stream_out {
- struct audio_stream_out stream;
-
- pthread_mutex_t lock; /* see note below on mutex acquisition order */
- struct pcm_config config;
- struct pcm *pcm;
- bool unavailable;
- int standby;
- struct alsa_audio_device *dev;
- int write_threshold;
- unsigned int written;
-};
+/* If not NULL, this is a pointer to the fallback module.
+ * This really is the original goldfish audio device /dev/eac which we will use
+ * if no alsa devices are detected.
+ */
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state);
+static int adev_get_microphones(const audio_hw_device_t *dev,
+ struct audio_microphone_characteristic_t *mic_array,
+ size_t *mic_count);
-/* must be called with hw device and output stream mutexes locked */
-static int start_output_stream(struct alsa_stream_out *out)
-{
- struct alsa_audio_device *adev = out->dev;
+typedef struct audio_vbuffer {
+ pthread_mutex_t lock;
+ uint8_t * data;
+ size_t frame_size;
+ size_t frame_count;
+ size_t head;
+ size_t tail;
+ size_t live;
+} audio_vbuffer_t;
- if (out->unavailable)
- return -ENODEV;
-
- /* default to low power: will be corrected in out_write if necessary before first write to
- * tinyalsa.
- */
- out->write_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE;
- out->config.start_threshold = PLAYBACK_PERIOD_START_THRESHOLD * PERIOD_SIZE;
- out->config.avail_min = PERIOD_SIZE;
-
- out->pcm = pcm_open(CARD_OUT, PORT_CODEC, PCM_OUT | PCM_MMAP | PCM_NOIRQ | PCM_MONOTONIC, &out->config);
-
- if (!pcm_is_ready(out->pcm)) {
- ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm));
- pcm_close(out->pcm);
- adev->active_output = NULL;
- out->unavailable = true;
- return -ENODEV;
+static int audio_vbuffer_init (audio_vbuffer_t * audio_vbuffer, size_t frame_count,
+ size_t frame_size) {
+ if (!audio_vbuffer) {
+ return -EINVAL;
}
-
- adev->active_output = out;
+ audio_vbuffer->frame_size = frame_size;
+ audio_vbuffer->frame_count = frame_count;
+ size_t bytes = frame_count * frame_size;
+ audio_vbuffer->data = calloc(bytes, 1);
+ if (!audio_vbuffer->data) {
+ return -ENOMEM;
+ }
+ audio_vbuffer->head = 0;
+ audio_vbuffer->tail = 0;
+ audio_vbuffer->live = 0;
+ pthread_mutex_init (&audio_vbuffer->lock, (const pthread_mutexattr_t *) NULL);
return 0;
}
+static int audio_vbuffer_destroy (audio_vbuffer_t * audio_vbuffer) {
+ if (!audio_vbuffer) {
+ return -EINVAL;
+ }
+ free(audio_vbuffer->data);
+ pthread_mutex_destroy(&audio_vbuffer->lock);
+ return 0;
+}
+
+static int audio_vbuffer_live (audio_vbuffer_t * audio_vbuffer) {
+ if (!audio_vbuffer) {
+ return -EINVAL;
+ }
+ pthread_mutex_lock (&audio_vbuffer->lock);
+ int live = audio_vbuffer->live;
+ pthread_mutex_unlock (&audio_vbuffer->lock);
+ return live;
+}
+
+#define MIN(a,b) (((a)<(b))?(a):(b))
+static size_t audio_vbuffer_write (audio_vbuffer_t * audio_vbuffer, const void * buffer, size_t frame_count) {
+ size_t frames_written = 0;
+ pthread_mutex_lock (&audio_vbuffer->lock);
+
+ while (frame_count != 0) {
+ int frames = 0;
+ if (audio_vbuffer->live == 0 || audio_vbuffer->head > audio_vbuffer->tail) {
+ frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->head);
+ } else if (audio_vbuffer->head < audio_vbuffer->tail) {
+ frames = MIN(frame_count, audio_vbuffer->tail - (audio_vbuffer->head));
+ } else {
+ // Full
+ break;
+ }
+ memcpy(&audio_vbuffer->data[audio_vbuffer->head*audio_vbuffer->frame_size],
+ &((uint8_t*)buffer)[frames_written*audio_vbuffer->frame_size],
+ frames*audio_vbuffer->frame_size);
+ audio_vbuffer->live += frames;
+ frames_written += frames;
+ frame_count -= frames;
+ audio_vbuffer->head = (audio_vbuffer->head + frames) % audio_vbuffer->frame_count;
+ }
+
+ pthread_mutex_unlock (&audio_vbuffer->lock);
+ return frames_written;
+}
+
+static size_t audio_vbuffer_read (audio_vbuffer_t * audio_vbuffer, void * buffer, size_t frame_count) {
+ size_t frames_read = 0;
+ pthread_mutex_lock (&audio_vbuffer->lock);
+
+ while (frame_count != 0) {
+ int frames = 0;
+ if (audio_vbuffer->live == audio_vbuffer->frame_count ||
+ audio_vbuffer->tail > audio_vbuffer->head) {
+ frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->tail);
+ } else if (audio_vbuffer->tail < audio_vbuffer->head) {
+ frames = MIN(frame_count, audio_vbuffer->head - audio_vbuffer->tail);
+ } else {
+ break;
+ }
+ memcpy(&((uint8_t*)buffer)[frames_read*audio_vbuffer->frame_size],
+ &audio_vbuffer->data[audio_vbuffer->tail*audio_vbuffer->frame_size],
+ frames*audio_vbuffer->frame_size);
+ audio_vbuffer->live -= frames;
+ frames_read += frames;
+ frame_count -= frames;
+ audio_vbuffer->tail = (audio_vbuffer->tail + frames) % audio_vbuffer->frame_count;
+ }
+
+ pthread_mutex_unlock (&audio_vbuffer->lock);
+ return frames_read;
+}
+
+struct generic_stream_out {
+ struct audio_stream_out stream; // Constant after init
+ pthread_mutex_t lock;
+ struct generic_audio_device *dev; // Constant after init
+ uint32_t num_devices; // Protected by this->lock
+ audio_devices_t devices[AUDIO_PATCH_PORTS_MAX]; // Protected by this->lock
+ struct audio_config req_config; // Constant after init
+ struct pcm_config pcm_config; // Constant after init
+ audio_vbuffer_t buffer; // Constant after init
+
+ // Time & Position Keeping
+ bool standby; // Protected by this->lock
+ uint64_t underrun_position; // Protected by this->lock
+ struct timespec underrun_time; // Protected by this->lock
+ uint64_t last_write_time_us; // Protected by this->lock
+ uint64_t frames_total_buffered; // Protected by this->lock
+ uint64_t frames_written; // Protected by this->lock
+ uint64_t frames_rendered; // Protected by this->lock
+
+ // Worker
+ pthread_t worker_thread; // Constant after init
+ pthread_cond_t worker_wake; // Protected by this->lock
+ bool worker_standby; // Protected by this->lock
+ bool worker_exit; // Protected by this->lock
+
+ audio_io_handle_t handle; // Constant after init
+ audio_patch_handle_t patch_handle; // Protected by this->dev->lock
+
+ struct listnode stream_node; // Protected by this->dev->lock
+};
+
+struct generic_stream_in {
+ struct audio_stream_in stream; // Constant after init
+ pthread_mutex_t lock;
+ struct generic_audio_device *dev; // Constant after init
+ audio_devices_t device; // Protected by this->lock
+ struct audio_config req_config; // Constant after init
+ struct pcm *pcm; // Protected by this->lock
+ struct pcm_config pcm_config; // Constant after init
+ int16_t *stereo_to_mono_buf; // Protected by this->lock
+ size_t stereo_to_mono_buf_size; // Protected by this->lock
+ audio_vbuffer_t buffer; // Protected by this->lock
+
+ // Time & Position Keeping
+ bool standby; // Protected by this->lock
+ int64_t standby_position; // Protected by this->lock
+ struct timespec standby_exit_time;// Protected by this->lock
+ int64_t standby_frames_read; // Protected by this->lock
+
+ // Worker
+ pthread_t worker_thread; // Constant after init
+ pthread_cond_t worker_wake; // Protected by this->lock
+ bool worker_standby; // Protected by this->lock
+ bool worker_exit; // Protected by this->lock
+
+ audio_io_handle_t handle; // Constant after init
+ audio_patch_handle_t patch_handle; // Protected by this->dev->lock
+
+ struct listnode stream_node; // Protected by this->dev->lock
+};
+
+static struct pcm_config pcm_config_out = {
+ .channels = 2,
+ .rate = 0,
+ .period_size = 0,
+ .period_count = OUT_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 0,
+};
+
+static struct pcm_config pcm_config_in = {
+ .channels = 2,
+ .rate = 0,
+ .period_size = 0,
+ .period_count = IN_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 0,
+ .stop_threshold = INT_MAX,
+};
+
+static pthread_mutex_t adev_init_lock = PTHREAD_MUTEX_INITIALIZER;
+static unsigned int audio_device_ref_count = 0;
+
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- return out->config.rate;
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ return out->req_config.sample_rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
- ALOGV("out_set_sample_rate: %d", 0);
return -ENOSYS;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
- ALOGV("out_get_buffer_size: %d", 4096);
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ int size = out->pcm_config.period_size *
+ audio_stream_out_frame_size(&out->stream);
- /* return the closest majoring multiple of 16 frames, as
- * audioflinger expects audio buffers to be a multiple of 16 frames */
- size_t size = PERIOD_SIZE;
- size = ((size + 15) / 16) * 16;
- return size * audio_stream_out_frame_size((struct audio_stream_out *)stream);
+ return size;
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
- ALOGV("out_get_channels");
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- return audio_channel_out_mask_from_count(out->config.channels);
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ return out->req_config.channel_mask;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
- ALOGV("out_get_format");
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- return audio_format_from_pcm_format(out->config.format);
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+
+ return out->req_config.format;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
- ALOGV("out_set_format: %d",format);
return -ENOSYS;
}
-static int do_output_standby(struct alsa_stream_out *out)
-{
- struct alsa_audio_device *adev = out->dev;
-
- if (!out->standby) {
- pcm_close(out->pcm);
- out->pcm = NULL;
- adev->active_output = NULL;
- out->standby = 1;
- }
- return 0;
-}
-
-static int out_standby(struct audio_stream *stream)
-{
- ALOGV("out_standby");
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- int status;
-
- pthread_mutex_lock(&out->dev->lock);
- pthread_mutex_lock(&out->lock);
- status = do_output_standby(out);
- pthread_mutex_unlock(&out->lock);
- pthread_mutex_unlock(&out->dev->lock);
- return status;
-}
-
static int out_dump(const struct audio_stream *stream, int fd)
{
- ALOGV("out_dump");
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ pthread_mutex_lock(&out->lock);
+ dprintf(fd, "\tout_dump:\n"
+ "\t\tsample rate: %u\n"
+ "\t\tbuffer size: %zu\n"
+ "\t\tchannel mask: %08x\n"
+ "\t\tformat: %d\n"
+ "\t\tdevice(s): ",
+ out_get_sample_rate(stream),
+ out_get_buffer_size(stream),
+ out_get_channels(stream),
+ out_get_format(stream));
+ if (out->num_devices == 0) {
+ dprintf(fd, "%08x\n", AUDIO_DEVICE_NONE);
+ } else {
+ for (uint32_t i = 0; i < out->num_devices; i++) {
+ if (i != 0) {
+ dprintf(fd, ", ");
+ }
+ dprintf(fd, "%08x", out->devices[i]);
+ }
+ dprintf(fd, "\n");
+ }
+ dprintf(fd, "\t\taudio dev: %p\n\n", out->dev);
+ pthread_mutex_unlock(&out->lock);
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
- ALOGV("out_set_parameters");
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- struct alsa_audio_device *adev = out->dev;
struct str_parms *parms;
char value[32];
- int ret, val = 0;
+ int success;
+ int ret = -EINVAL;
+ if (kvpairs == NULL || kvpairs[0] == 0) {
+ return 0;
+ }
parms = str_parms_create_str(kvpairs);
+ success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
+ value, sizeof(value));
+ // As the hal version is 3.0, it must not use set parameters API to set audio devices.
+ // Instead, it should use create_audio_patch API.
+ assert(("Must not use set parameters API to set audio devices", success < 0));
- ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
- if (ret >= 0) {
- val = atoi(value);
- pthread_mutex_lock(&adev->lock);
- pthread_mutex_lock(&out->lock);
- if (((adev->devices & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) {
- adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
- adev->devices |= val;
- }
- pthread_mutex_unlock(&out->lock);
- pthread_mutex_unlock(&adev->lock);
+ if (str_parms_has_key(parms, AUDIO_PARAMETER_STREAM_FORMAT)) {
+ // match the return value of out_set_format
+ ret = -ENOSYS;
}
str_parms_destroy(parms);
+
+ if (ret == -EINVAL) {
+ ALOGW("%s(), unsupported parameter %s", __func__, kvpairs);
+ // There is not any key supported for set_parameters API.
+ // Return error when there is non-null value passed in.
+ }
return ret;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
- ALOGV("out_get_parameters");
- return strdup("");
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ struct str_parms *query = str_parms_create_str(keys);
+ char *str = NULL;
+ char value[256];
+ struct str_parms *reply = str_parms_create();
+ int ret;
+ bool get = false;
+
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+ if (ret >= 0) {
+ pthread_mutex_lock(&out->lock);
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+ for (uint32_t i = 0; i < out->num_devices; i++) {
+ device |= out->devices[i];
+ }
+ str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, device);
+ pthread_mutex_unlock(&out->lock);
+ get = true;
+ }
+
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+ value[0] = 0;
+ strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+ str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
+ get = true;
+ }
+
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
+ value[0] = 0;
+ strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+ str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
+ get = true;
+ }
+
+ if (get) {
+ str = str_parms_to_str(reply);
+ }
+ else {
+ ALOGD("%s Unsupported parameter: %s", __FUNCTION__, keys);
+ }
+
+ str_parms_destroy(query);
+ str_parms_destroy(reply);
+ return str;
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
- ALOGV("out_get_latency");
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- return (PERIOD_SIZE * PLAYBACK_PERIOD_COUNT * 1000) / out->config.rate;
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ return (out->pcm_config.period_size * 1000) / out->pcm_config.rate;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
- float right)
+ float right)
{
- ALOGV("out_set_volume: Left:%f Right:%f", left, right);
- return 0;
+ return -ENOSYS;
}
-static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
- size_t bytes)
+static void *out_write_worker(void * args)
{
- int ret;
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- struct alsa_audio_device *adev = out->dev;
- size_t frame_size = audio_stream_out_frame_size(stream);
- size_t out_frames = bytes / frame_size;
-
- /* acquiring hw device mutex systematically is useful if a low priority thread is waiting
- * on the output stream mutex - e.g. executing select_mode() while holding the hw device
- * mutex
- */
- pthread_mutex_lock(&adev->lock);
- pthread_mutex_lock(&out->lock);
- if (out->standby) {
- ret = start_output_stream(out);
- if (ret != 0) {
- pthread_mutex_unlock(&adev->lock);
- goto exit;
+ struct generic_stream_out *out = (struct generic_stream_out *)args;
+ struct pcm *pcm = NULL;
+ uint8_t *buffer = NULL;
+ int buffer_frames;
+ int buffer_size;
+ bool restart = false;
+ bool shutdown = false;
+ while (true) {
+ pthread_mutex_lock(&out->lock);
+ while (out->worker_standby || restart) {
+ restart = false;
+ if (pcm) {
+ pcm_close(pcm); // Frees pcm
+ pcm = NULL;
+ free(buffer);
+ buffer=NULL;
+ }
+ if (out->worker_exit) {
+ break;
+ }
+ pthread_cond_wait(&out->worker_wake, &out->lock);
}
- out->standby = 0;
+
+ if (out->worker_exit) {
+ if (!out->worker_standby) {
+ ALOGE("Out worker not in standby before exiting");
+ }
+ shutdown = true;
+ }
+
+ while (!shutdown && audio_vbuffer_live(&out->buffer) == 0) {
+ pthread_cond_wait(&out->worker_wake, &out->lock);
+ }
+
+ if (shutdown) {
+ pthread_mutex_unlock(&out->lock);
+ break;
+ }
+
+ if (!pcm) {
+ pcm = pcm_open(PCM_CARD, PCM_DEVICE,
+ PCM_OUT | PCM_MONOTONIC, &out->pcm_config);
+ if (!pcm_is_ready(pcm)) {
+ ALOGE("pcm_open(out) failed: %s: channels %d format %d rate %d",
+ pcm_get_error(pcm),
+ out->pcm_config.channels,
+ out->pcm_config.format,
+ out->pcm_config.rate
+ );
+ pthread_mutex_unlock(&out->lock);
+ break;
+ }
+ buffer_frames = out->pcm_config.period_size;
+ buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
+ buffer = malloc(buffer_size);
+ if (!buffer) {
+ ALOGE("could not allocate write buffer");
+ pthread_mutex_unlock(&out->lock);
+ break;
+ }
+ }
+ int frames = audio_vbuffer_read(&out->buffer, buffer, buffer_frames);
+ pthread_mutex_unlock(&out->lock);
+ int ret = pcm_write(pcm, buffer, pcm_frames_to_bytes(pcm, frames));
+ if (ret != 0) {
+ ALOGE("pcm_write failed %s", pcm_get_error(pcm));
+ restart = true;
+ }
+ }
+ if (buffer) {
+ free(buffer);
}
- pthread_mutex_unlock(&adev->lock);
+ return NULL;
+}
- ret = pcm_mmap_write(out->pcm, buffer, out_frames * frame_size);
- if (ret == 0) {
- out->written += out_frames;
+// Call with in->lock held
+static void get_current_output_position(struct generic_stream_out *out,
+ uint64_t * position,
+ struct timespec * timestamp) {
+ struct timespec curtime = { .tv_sec = 0, .tv_nsec = 0 };
+ clock_gettime(CLOCK_MONOTONIC, &curtime);
+ const int64_t now_us = (curtime.tv_sec * 1000000000LL + curtime.tv_nsec) / 1000;
+ if (timestamp) {
+ *timestamp = curtime;
}
-exit:
+ int64_t position_since_underrun;
+ if (out->standby) {
+ position_since_underrun = 0;
+ } else {
+ const int64_t first_us = (out->underrun_time.tv_sec * 1000000000LL +
+ out->underrun_time.tv_nsec) / 1000;
+ position_since_underrun = (now_us - first_us) *
+ out_get_sample_rate(&out->stream.common) /
+ 1000000;
+ if (position_since_underrun < 0) {
+ position_since_underrun = 0;
+ }
+ }
+ *position = out->underrun_position + position_since_underrun;
+
+ // The device will reuse the same output stream leading to periods of
+ // underrun.
+ if (*position > out->frames_written) {
+ ALOGW("Not supplying enough data to HAL, expected position %" PRIu64 " , only wrote "
+ "%" PRIu64,
+ *position, out->frames_written);
+
+ *position = out->frames_written;
+ out->underrun_position = *position;
+ out->underrun_time = curtime;
+ out->frames_total_buffered = 0;
+ }
+}
+
+
+static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
+ size_t bytes)
+{
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ const size_t frames = bytes / audio_stream_out_frame_size(stream);
+
+ pthread_mutex_lock(&out->lock);
+
+ if (out->worker_standby) {
+ out->worker_standby = false;
+ }
+
+ uint64_t current_position;
+ struct timespec current_time;
+
+ get_current_output_position(out, ¤t_position, ¤t_time);
+ const uint64_t now_us = (current_time.tv_sec * 1000000000LL +
+ current_time.tv_nsec) / 1000;
+ if (out->standby) {
+ out->standby = false;
+ out->underrun_time = current_time;
+ out->frames_rendered = 0;
+ out->frames_total_buffered = 0;
+ }
+
+ size_t frames_written = audio_vbuffer_write(&out->buffer, buffer, frames);
+ pthread_cond_signal(&out->worker_wake);
+
+ /* Implementation just consumes bytes if we start getting backed up */
+ out->frames_written += frames;
+ out->frames_rendered += frames;
+ out->frames_total_buffered += frames;
+
+ // We simulate the audio device blocking when it's write buffers become
+ // full.
+
+ // At the beginning or after an underrun, try to fill up the vbuffer.
+ // This will be throttled by the PlaybackThread
+ int frames_sleep = out->frames_total_buffered < out->buffer.frame_count ? 0 : frames;
+
+ uint64_t sleep_time_us = frames_sleep * 1000000LL /
+ out_get_sample_rate(&stream->common);
+
+ // If the write calls are delayed, subtract time off of the sleep to
+ // compensate
+ uint64_t time_since_last_write_us = now_us - out->last_write_time_us;
+ if (time_since_last_write_us < sleep_time_us) {
+ sleep_time_us -= time_since_last_write_us;
+ } else {
+ sleep_time_us = 0;
+ }
+ out->last_write_time_us = now_us + sleep_time_us;
+
pthread_mutex_unlock(&out->lock);
- if (ret != 0) {
- usleep((int64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
- out_get_sample_rate(&stream->common));
+ if (sleep_time_us > 0) {
+ usleep(sleep_time_us);
}
- return bytes;
-}
+ if (frames_written < frames) {
+ ALOGW("Hardware backing HAL too slow, could only write %zu of %zu frames", frames_written, frames);
+ }
-static int out_get_render_position(const struct audio_stream_out *stream,
- uint32_t *dsp_frames)
-{
- *dsp_frames = 0;
- ALOGV("out_get_render_position: dsp_frames: %p", dsp_frames);
- return -EINVAL;
+ /* Always consume all bytes */
+ return bytes;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
+
{
- struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
- int ret = -1;
+ if (stream == NULL || frames == NULL || timestamp == NULL) {
+ return -EINVAL;
+ }
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
- if (out->pcm) {
- unsigned int avail;
- if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
- size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
- int64_t signed_frames = out->written - kernel_buffer_size + avail;
- if (signed_frames >= 0) {
- *frames = signed_frames;
- ret = 0;
- }
- }
- }
+ pthread_mutex_lock(&out->lock);
+ get_current_output_position(out, frames, timestamp);
+ pthread_mutex_unlock(&out->lock);
- return ret;
+ return 0;
}
+static int out_get_render_position(const struct audio_stream_out *stream,
+ uint32_t *dsp_frames)
+{
+ if (stream == NULL || dsp_frames == NULL) {
+ return -EINVAL;
+ }
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ pthread_mutex_lock(&out->lock);
+ *dsp_frames = out->frames_rendered;
+ pthread_mutex_unlock(&out->lock);
+ return 0;
+}
+
+// Must be called with out->lock held
+static void do_out_standby(struct generic_stream_out *out)
+{
+ int frames_sleep = 0;
+ uint64_t sleep_time_us = 0;
+ if (out->standby) {
+ return;
+ }
+ while (true) {
+ get_current_output_position(out, &out->underrun_position, NULL);
+ frames_sleep = out->frames_written - out->underrun_position;
+
+ if (frames_sleep == 0) {
+ break;
+ }
+
+ sleep_time_us = frames_sleep * 1000000LL /
+ out_get_sample_rate(&out->stream.common);
+
+ pthread_mutex_unlock(&out->lock);
+ usleep(sleep_time_us);
+ pthread_mutex_lock(&out->lock);
+ }
+ out->worker_standby = true;
+ out->standby = true;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ pthread_mutex_lock(&out->lock);
+ do_out_standby(out);
+ pthread_mutex_unlock(&out->lock);
+ return 0;
+}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
- ALOGV("out_add_audio_effect: %p", effect);
+ // out_add_audio_effect is a no op
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
- ALOGV("out_remove_audio_effect: %p", effect);
+ // out_remove_audio_effect is a no op
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
- int64_t *timestamp)
+ int64_t *timestamp)
{
- *timestamp = 0;
- ALOGV("out_get_next_write_timestamp: %ld", (long int)(*timestamp));
- return -EINVAL;
+ return -ENOSYS;
}
-/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
- ALOGV("in_get_sample_rate");
- return 8000;
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ return in->req_config.sample_rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
- ALOGV("in_set_sample_rate: %d", rate);
return -ENOSYS;
}
+static int refine_output_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
+{
+ static const uint32_t sample_rates [] = {8000,11025,16000,22050,24000,32000,
+ 44100,48000};
+ static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
+ bool inval = false;
+ if (*format != AUDIO_FORMAT_PCM_16_BIT) {
+ *format = AUDIO_FORMAT_PCM_16_BIT;
+ inval = true;
+ }
+
+ int channel_count = popcount(*channel_mask);
+ if (channel_count != 1 && channel_count != 2) {
+ *channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ inval = true;
+ }
+
+ int i;
+ for (i = 0; i < sample_rates_count; i++) {
+ if (*sample_rate < sample_rates[i]) {
+ *sample_rate = sample_rates[i];
+ inval=true;
+ break;
+ }
+ else if (*sample_rate == sample_rates[i]) {
+ break;
+ }
+ else if (i == sample_rates_count-1) {
+ // Cap it to the highest rate we support
+ *sample_rate = sample_rates[i];
+ inval=true;
+ }
+ }
+
+ if (inval) {
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int refine_input_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
+{
+ static const uint32_t sample_rates [] = {8000, 11025, 16000, 22050, 44100, 48000};
+ static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
+ bool inval = false;
+ // Only PCM_16_bit is supported. If this is changed, stereo to mono drop
+ // must be fixed in in_read
+ if (*format != AUDIO_FORMAT_PCM_16_BIT) {
+ *format = AUDIO_FORMAT_PCM_16_BIT;
+ inval = true;
+ }
+
+ int channel_count = popcount(*channel_mask);
+ if (channel_count != 1 && channel_count != 2) {
+ *channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ inval = true;
+ }
+
+ int i;
+ for (i = 0; i < sample_rates_count; i++) {
+ if (*sample_rate < sample_rates[i]) {
+ *sample_rate = sample_rates[i];
+ inval=true;
+ break;
+ }
+ else if (*sample_rate == sample_rates[i]) {
+ break;
+ }
+ else if (i == sample_rates_count-1) {
+ // Cap it to the highest rate we support
+ *sample_rate = sample_rates[i];
+ inval=true;
+ }
+ }
+
+ if (inval) {
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int check_input_parameters(uint32_t sample_rate, audio_format_t format,
+ audio_channel_mask_t channel_mask)
+{
+ return refine_input_parameters(&sample_rate, &format, &channel_mask);
+}
+
+static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format,
+ audio_channel_mask_t channel_mask)
+{
+ size_t size;
+ int channel_count = popcount(channel_mask);
+ if (check_input_parameters(sample_rate, format, channel_mask) != 0)
+ return 0;
+
+ size = sample_rate*IN_PERIOD_MS/1000;
+ // Audioflinger expects audio buffers to be multiple of 16 frames
+ size = ((size + 15) / 16) * 16;
+ size *= sizeof(short) * channel_count;
+
+ return size;
+}
+
+
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
- ALOGV("in_get_buffer_size: %d", 320);
- return 320;
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ int size = get_input_buffer_size(in->req_config.sample_rate,
+ in->req_config.format,
+ in->req_config.channel_mask);
+
+ return size;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
- ALOGV("in_get_channels: %d", AUDIO_CHANNEL_IN_MONO);
- return AUDIO_CHANNEL_IN_MONO;
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ return in->req_config.channel_mask;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
- return AUDIO_FORMAT_PCM_16_BIT;
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ return in->req_config.format;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
@@ -365,40 +825,326 @@
return -ENOSYS;
}
-static int in_standby(struct audio_stream *stream)
-{
- return 0;
-}
-
static int in_dump(const struct audio_stream *stream, int fd)
{
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+
+ pthread_mutex_lock(&in->lock);
+ dprintf(fd, "\tin_dump:\n"
+ "\t\tsample rate: %u\n"
+ "\t\tbuffer size: %zu\n"
+ "\t\tchannel mask: %08x\n"
+ "\t\tformat: %d\n"
+ "\t\tdevice: %08x\n"
+ "\t\taudio dev: %p\n\n",
+ in_get_sample_rate(stream),
+ in_get_buffer_size(stream),
+ in_get_channels(stream),
+ in_get_format(stream),
+ in->device,
+ in->dev);
+ pthread_mutex_unlock(&in->lock);
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
- return 0;
+ struct str_parms *parms;
+ char value[32];
+ int success;
+ int ret = -EINVAL;
+
+ if (kvpairs == NULL || kvpairs[0] == 0) {
+ return 0;
+ }
+ parms = str_parms_create_str(kvpairs);
+ success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
+ value, sizeof(value));
+ // As the hal version is 3.0, it must not use set parameters API to set audio device.
+ // Instead, it should use create_audio_patch API.
+ assert(("Must not use set parameters API to set audio devices", success < 0));
+
+ if (str_parms_has_key(parms, AUDIO_PARAMETER_STREAM_FORMAT)) {
+ // match the return value of in_set_format
+ ret = -ENOSYS;
+ }
+
+ str_parms_destroy(parms);
+
+ if (ret == -EINVAL) {
+ ALOGW("%s(), unsupported parameter %s", __func__, kvpairs);
+ // There is not any key supported for set_parameters API.
+ // Return error when there is non-null value passed in.
+ }
+ return ret;
}
static char * in_get_parameters(const struct audio_stream *stream,
- const char *keys)
+ const char *keys)
{
- return strdup("");
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ struct str_parms *query = str_parms_create_str(keys);
+ char *str = NULL;
+ char value[256];
+ struct str_parms *reply = str_parms_create();
+ int ret;
+ bool get = false;
+
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+ if (ret >= 0) {
+ str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, in->device);
+ get = true;
+ }
+
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+ value[0] = 0;
+ strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+ str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
+ get = true;
+ }
+
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
+ value[0] = 0;
+ strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+ str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
+ get = true;
+ }
+
+ if (get) {
+ str = str_parms_to_str(reply);
+ }
+ else {
+ ALOGD("%s Unsupported parameter: %s", __FUNCTION__, keys);
+ }
+
+ str_parms_destroy(query);
+ str_parms_destroy(reply);
+ return str;
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
+ // in_set_gain is a no op
return 0;
}
-static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
- size_t bytes)
+// Call with in->lock held
+static void get_current_input_position(struct generic_stream_in *in,
+ int64_t * position,
+ struct timespec * timestamp) {
+ struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
+ clock_gettime(CLOCK_MONOTONIC, &t);
+ const int64_t now_us = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
+ if (timestamp) {
+ *timestamp = t;
+ }
+ int64_t position_since_standby;
+ if (in->standby) {
+ position_since_standby = 0;
+ } else {
+ const int64_t first_us = (in->standby_exit_time.tv_sec * 1000000000LL +
+ in->standby_exit_time.tv_nsec) / 1000;
+ position_since_standby = (now_us - first_us) *
+ in_get_sample_rate(&in->stream.common) /
+ 1000000;
+ if (position_since_standby < 0) {
+ position_since_standby = 0;
+ }
+ }
+ *position = in->standby_position + position_since_standby;
+}
+
+// Must be called with in->lock held
+static void do_in_standby(struct generic_stream_in *in)
{
- ALOGV("in_read: bytes %zu", bytes);
- /* XXX: fake timing for audio input */
- usleep((int64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
- in_get_sample_rate(&stream->common));
- memset(buffer, 0, bytes);
+ if (in->standby) {
+ return;
+ }
+ in->worker_standby = true;
+ get_current_input_position(in, &in->standby_position, NULL);
+ in->standby = true;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ pthread_mutex_lock(&in->lock);
+ do_in_standby(in);
+ pthread_mutex_unlock(&in->lock);
+ return 0;
+}
+
+static void *in_read_worker(void * args)
+{
+ struct generic_stream_in *in = (struct generic_stream_in *)args;
+ struct pcm *pcm = NULL;
+ uint8_t *buffer = NULL;
+ size_t buffer_frames;
+ int buffer_size;
+
+ bool restart = false;
+ bool shutdown = false;
+ while (true) {
+ pthread_mutex_lock(&in->lock);
+ while (in->worker_standby || restart) {
+ restart = false;
+ if (pcm) {
+ pcm_close(pcm); // Frees pcm
+ pcm = NULL;
+ free(buffer);
+ buffer=NULL;
+ }
+ if (in->worker_exit) {
+ break;
+ }
+ pthread_cond_wait(&in->worker_wake, &in->lock);
+ }
+
+ if (in->worker_exit) {
+ if (!in->worker_standby) {
+ ALOGE("In worker not in standby before exiting");
+ }
+ shutdown = true;
+ }
+ if (shutdown) {
+ pthread_mutex_unlock(&in->lock);
+ break;
+ }
+ if (!pcm) {
+ pcm = pcm_open(PCM_CARD, PCM_DEVICE,
+ PCM_IN | PCM_MONOTONIC, &in->pcm_config);
+ if (!pcm_is_ready(pcm)) {
+ ALOGE("pcm_open(in) failed: %s: channels %d format %d rate %d",
+ pcm_get_error(pcm),
+ in->pcm_config.channels,
+ in->pcm_config.format,
+ in->pcm_config.rate
+ );
+ pthread_mutex_unlock(&in->lock);
+ break;
+ }
+ buffer_frames = in->pcm_config.period_size;
+ buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
+ buffer = malloc(buffer_size);
+ if (!buffer) {
+ ALOGE("could not allocate worker read buffer");
+ pthread_mutex_unlock(&in->lock);
+ break;
+ }
+ }
+ pthread_mutex_unlock(&in->lock);
+ int ret = pcm_read(pcm, buffer, pcm_frames_to_bytes(pcm, buffer_frames));
+ if (ret != 0) {
+ ALOGW("pcm_read failed %s", pcm_get_error(pcm));
+ restart = true;
+ continue;
+ }
+
+ pthread_mutex_lock(&in->lock);
+ size_t frames_written = audio_vbuffer_write(&in->buffer, buffer, buffer_frames);
+ pthread_mutex_unlock(&in->lock);
+
+ if (frames_written != buffer_frames) {
+ ALOGW("in_read_worker only could write %zu / %zu frames", frames_written, buffer_frames);
+ }
+ }
+ if (buffer) {
+ free(buffer);
+ }
+ return NULL;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
+ size_t bytes)
+{
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ struct generic_audio_device *adev = in->dev;
+ const size_t frames = bytes / audio_stream_in_frame_size(stream);
+ bool mic_mute = false;
+ size_t read_bytes = 0;
+
+ adev_get_mic_mute(&adev->device, &mic_mute);
+ pthread_mutex_lock(&in->lock);
+
+ if (in->worker_standby) {
+ in->worker_standby = false;
+ }
+ pthread_cond_signal(&in->worker_wake);
+
+ int64_t current_position;
+ struct timespec current_time;
+
+ get_current_input_position(in, ¤t_position, ¤t_time);
+ if (in->standby) {
+ in->standby = false;
+ in->standby_exit_time = current_time;
+ in->standby_frames_read = 0;
+ }
+
+ const int64_t frames_available = current_position - in->standby_position - in->standby_frames_read;
+ assert(frames_available >= 0);
+
+ const size_t frames_wait = ((uint64_t)frames_available > frames) ? 0 : frames - frames_available;
+
+ int64_t sleep_time_us = frames_wait * 1000000LL /
+ in_get_sample_rate(&stream->common);
+
+ pthread_mutex_unlock(&in->lock);
+
+ if (sleep_time_us > 0) {
+ usleep(sleep_time_us);
+ }
+
+ pthread_mutex_lock(&in->lock);
+ int read_frames = 0;
+ if (in->standby) {
+ ALOGW("Input put to sleep while read in progress");
+ goto exit;
+ }
+ in->standby_frames_read += frames;
+
+ if (popcount(in->req_config.channel_mask) == 1 &&
+ in->pcm_config.channels == 2) {
+ // Need to resample to mono
+ if (in->stereo_to_mono_buf_size < bytes*2) {
+ in->stereo_to_mono_buf = realloc(in->stereo_to_mono_buf,
+ bytes*2);
+ if (!in->stereo_to_mono_buf) {
+ ALOGE("Failed to allocate stereo_to_mono_buff");
+ goto exit;
+ }
+ }
+
+ read_frames = audio_vbuffer_read(&in->buffer, in->stereo_to_mono_buf, frames);
+
+ // Currently only pcm 16 is supported.
+ uint16_t *src = (uint16_t *)in->stereo_to_mono_buf;
+ uint16_t *dst = (uint16_t *)buffer;
+ size_t i;
+ // Resample stereo 16 to mono 16 by dropping one channel.
+ // The stereo stream is interleaved L-R-L-R
+ for (i = 0; i < frames; i++) {
+ *dst = *src;
+ src += 2;
+ dst += 1;
+ }
+ } else {
+ read_frames = audio_vbuffer_read(&in->buffer, buffer, frames);
+ }
+
+exit:
+ read_bytes = read_frames*audio_stream_in_frame_size(stream);
+
+ if (mic_mute) {
+ read_bytes = 0;
+ }
+
+ if (read_bytes < bytes) {
+ memset (&((uint8_t *)buffer)[read_bytes], 0, bytes-read_bytes);
+ }
+
+ pthread_mutex_unlock(&in->lock);
+
return bytes;
}
@@ -407,36 +1153,58 @@
return 0;
}
+static int in_get_capture_position(const struct audio_stream_in *stream,
+ int64_t *frames, int64_t *time)
+{
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ pthread_mutex_lock(&in->lock);
+ struct timespec current_time;
+ get_current_input_position(in, frames, ¤t_time);
+ *time = (current_time.tv_sec * 1000000000LL + current_time.tv_nsec);
+ pthread_mutex_unlock(&in->lock);
+ return 0;
+}
+
+static int in_get_active_microphones(const struct audio_stream_in *stream,
+ struct audio_microphone_characteristic_t *mic_array,
+ size_t *mic_count)
+{
+ return adev_get_microphones(NULL, mic_array, mic_count);
+}
+
static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
+ // in_add_audio_effect is a no op
return 0;
}
static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
+ // in_add_audio_effect is a no op
return 0;
}
static int adev_open_output_stream(struct audio_hw_device *dev,
- audio_io_handle_t handle,
- audio_devices_t devices,
- audio_output_flags_t flags,
- struct audio_config *config,
- struct audio_stream_out **stream_out,
- const char *address __unused)
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address __unused)
{
- ALOGV("adev_open_output_stream...");
-
- struct alsa_audio_device *ladev = (struct alsa_audio_device *)dev;
- struct alsa_stream_out *out;
- struct pcm_params *params;
+ struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+ struct generic_stream_out *out;
int ret = 0;
- params = pcm_params_get(CARD_OUT, PORT_CODEC, PCM_OUT);
- if (!params)
- return -ENOSYS;
+ if (refine_output_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
+ ALOGE("Error opening output stream format %d, channel_mask %04x, sample_rate %u",
+ config->format, config->channel_mask, config->sample_rate);
+ ret = -EINVAL;
+ goto error;
+ }
- out = (struct alsa_stream_out *)calloc(1, sizeof(struct alsa_stream_out));
+ out = (struct generic_stream_out *)calloc(1, sizeof(struct generic_stream_out));
+
if (!out)
return -ENOMEM;
@@ -456,221 +1224,598 @@
out->stream.set_volume = out_set_volume;
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
- out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
+ out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
- out->config.channels = CHANNEL_STEREO;
- out->config.rate = CODEC_SAMPLING_RATE;
- out->config.format = PCM_FORMAT_S16_LE;
- out->config.period_size = PERIOD_SIZE;
- out->config.period_count = PLAYBACK_PERIOD_COUNT;
+ out->handle = handle;
- if (out->config.rate != config->sample_rate ||
- audio_channel_count_from_out_mask(config->channel_mask) != CHANNEL_STEREO ||
- out->config.format != pcm_format_from_audio_format(config->format) ) {
- config->sample_rate = out->config.rate;
- config->format = audio_format_from_pcm_format(out->config.format);
- config->channel_mask = audio_channel_out_mask_from_count(CHANNEL_STEREO);
- ret = -EINVAL;
+ pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
+ out->dev = adev;
+ // Only 1 device is expected despite the argument being named 'devices'
+ out->num_devices = 1;
+ out->devices[0] = devices;
+ memcpy(&out->req_config, config, sizeof(struct audio_config));
+ memcpy(&out->pcm_config, &pcm_config_out, sizeof(struct pcm_config));
+ out->pcm_config.rate = config->sample_rate;
+ out->pcm_config.period_size = out->pcm_config.rate*OUT_PERIOD_MS/1000;
+
+ out->standby = true;
+ out->underrun_position = 0;
+ out->underrun_time.tv_sec = 0;
+ out->underrun_time.tv_nsec = 0;
+ out->last_write_time_us = 0;
+ out->frames_total_buffered = 0;
+ out->frames_written = 0;
+ out->frames_rendered = 0;
+
+ ret = audio_vbuffer_init(&out->buffer,
+ out->pcm_config.period_size*out->pcm_config.period_count,
+ out->pcm_config.channels *
+ pcm_format_to_bits(out->pcm_config.format) >> 3);
+ if (ret == 0) {
+ pthread_cond_init(&out->worker_wake, NULL);
+ out->worker_standby = true;
+ out->worker_exit = false;
+ pthread_create(&out->worker_thread, NULL, out_write_worker, out);
+
}
- ALOGI("adev_open_output_stream selects channels=%d rate=%d format=%d",
- out->config.channels, out->config.rate, out->config.format);
-
- out->dev = ladev;
- out->standby = 1;
- out->unavailable = false;
-
- config->format = out_get_format(&out->stream.common);
- config->channel_mask = out_get_channels(&out->stream.common);
- config->sample_rate = out_get_sample_rate(&out->stream.common);
+ pthread_mutex_lock(&adev->lock);
+ list_add_tail(&adev->out_streams, &out->stream_node);
+ pthread_mutex_unlock(&adev->lock);
*stream_out = &out->stream;
- /* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
- ret = 0;
+error:
return ret;
}
+// This must be called with adev->lock held.
+struct generic_stream_out *get_stream_out_by_io_handle_l(
+ struct generic_audio_device *adev, audio_io_handle_t handle) {
+ struct listnode *node;
+
+ list_for_each(node, &adev->out_streams) {
+ struct generic_stream_out *out = node_to_item(
+ node, struct generic_stream_out, stream_node);
+ if (out->handle == handle) {
+ return out;
+ }
+ }
+ return NULL;
+}
+
static void adev_close_output_stream(struct audio_hw_device *dev,
- struct audio_stream_out *stream)
+ struct audio_stream_out *stream)
{
- ALOGV("adev_close_output_stream...");
+ struct generic_stream_out *out = (struct generic_stream_out *)stream;
+ pthread_mutex_lock(&out->lock);
+ do_out_standby(out);
+
+ out->worker_exit = true;
+ pthread_cond_signal(&out->worker_wake);
+ pthread_mutex_unlock(&out->lock);
+
+ pthread_join(out->worker_thread, NULL);
+ pthread_mutex_destroy(&out->lock);
+ audio_vbuffer_destroy(&out->buffer);
+
+ struct generic_audio_device *adev = (struct generic_audio_device *) dev;
+ pthread_mutex_lock(&adev->lock);
+ list_remove(&out->stream_node);
+ pthread_mutex_unlock(&adev->lock);
free(stream);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
- ALOGV("adev_set_parameters");
- return -ENOSYS;
+ return 0;
}
static char * adev_get_parameters(const struct audio_hw_device *dev,
- const char *keys)
+ const char *keys)
{
- ALOGV("adev_get_parameters");
return strdup("");
}
static int adev_init_check(const struct audio_hw_device *dev)
{
- ALOGV("adev_init_check");
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
- ALOGV("adev_set_voice_volume: %f", volume);
- return -ENOSYS;
+ // adev_set_voice_volume is a no op (simulates phones)
+ return 0;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
- ALOGV("adev_set_master_volume: %f", volume);
return -ENOSYS;
}
static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
{
- ALOGV("adev_get_master_volume: %f", *volume);
return -ENOSYS;
}
static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
{
- ALOGV("adev_set_master_mute: %d", muted);
return -ENOSYS;
}
static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
{
- ALOGV("adev_get_master_mute: %d", *muted);
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
- ALOGV("adev_set_mode: %d", mode);
+ // adev_set_mode is a no op (simulates phones)
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
- ALOGV("adev_set_mic_mute: %d",state);
- return -ENOSYS;
+ struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+ pthread_mutex_lock(&adev->lock);
+ adev->mic_mute = state;
+ pthread_mutex_unlock(&adev->lock);
+ return 0;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
- ALOGV("adev_get_mic_mute");
- return -ENOSYS;
+ struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+ pthread_mutex_lock(&adev->lock);
+ *state = adev->mic_mute;
+ pthread_mutex_unlock(&adev->lock);
+ return 0;
}
+
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
- const struct audio_config *config)
+ const struct audio_config *config)
{
- ALOGV("adev_get_input_buffer_size: %d", 320);
- return 320;
+ return get_input_buffer_size(config->sample_rate, config->format, config->channel_mask);
}
-static int adev_open_input_stream(struct audio_hw_device __unused *dev,
- audio_io_handle_t handle,
- audio_devices_t devices,
- struct audio_config *config,
- struct audio_stream_in **stream_in,
- audio_input_flags_t flags __unused,
- const char *address __unused,
- audio_source_t source __unused)
+// This must be called with adev->lock held.
+struct generic_stream_in *get_stream_in_by_io_handle_l(
+ struct generic_audio_device *adev, audio_io_handle_t handle) {
+ struct listnode *node;
+
+ list_for_each(node, &adev->in_streams) {
+ struct generic_stream_in *in = node_to_item(
+ node, struct generic_stream_in, stream_node);
+ if (in->handle == handle) {
+ return in;
+ }
+ }
+ return NULL;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+ struct audio_stream_in *stream)
{
- struct stub_stream_in *in;
+ struct generic_stream_in *in = (struct generic_stream_in *)stream;
+ pthread_mutex_lock(&in->lock);
+ do_in_standby(in);
- ALOGV("adev_open_input_stream...");
+ in->worker_exit = true;
+ pthread_cond_signal(&in->worker_wake);
+ pthread_mutex_unlock(&in->lock);
+ pthread_join(in->worker_thread, NULL);
- in = (struct stub_stream_in *)calloc(1, sizeof(struct stub_stream_in));
- if (!in)
- return -ENOMEM;
+ if (in->stereo_to_mono_buf != NULL) {
+ free(in->stereo_to_mono_buf);
+ in->stereo_to_mono_buf_size = 0;
+ }
+
+ pthread_mutex_destroy(&in->lock);
+ audio_vbuffer_destroy(&in->buffer);
+
+ struct generic_audio_device *adev = (struct generic_audio_device *) dev;
+ pthread_mutex_lock(&adev->lock);
+ list_remove(&in->stream_node);
+ pthread_mutex_unlock(&adev->lock);
+ free(stream);
+}
+
+
+static int adev_open_input_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ struct audio_stream_in **stream_in,
+ audio_input_flags_t flags __unused,
+ const char *address __unused,
+ audio_source_t source __unused)
+{
+ struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+ struct generic_stream_in *in;
+ int ret = 0;
+ if (refine_input_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
+ ALOGE("Error opening input stream format %d, channel_mask %04x, sample_rate %u",
+ config->format, config->channel_mask, config->sample_rate);
+ ret = -EINVAL;
+ goto error;
+ }
+
+ in = (struct generic_stream_in *)calloc(1, sizeof(struct generic_stream_in));
+ if (!in) {
+ ret = -ENOMEM;
+ goto error;
+ }
in->stream.common.get_sample_rate = in_get_sample_rate;
- in->stream.common.set_sample_rate = in_set_sample_rate;
+ in->stream.common.set_sample_rate = in_set_sample_rate; // no op
in->stream.common.get_buffer_size = in_get_buffer_size;
in->stream.common.get_channels = in_get_channels;
in->stream.common.get_format = in_get_format;
- in->stream.common.set_format = in_set_format;
+ in->stream.common.set_format = in_set_format; // no op
in->stream.common.standby = in_standby;
in->stream.common.dump = in_dump;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
- in->stream.common.add_audio_effect = in_add_audio_effect;
- in->stream.common.remove_audio_effect = in_remove_audio_effect;
- in->stream.set_gain = in_set_gain;
+ in->stream.common.add_audio_effect = in_add_audio_effect; // no op
+ in->stream.common.remove_audio_effect = in_remove_audio_effect; // no op
+ in->stream.set_gain = in_set_gain; // no op
in->stream.read = in_read;
- in->stream.get_input_frames_lost = in_get_input_frames_lost;
+ in->stream.get_input_frames_lost = in_get_input_frames_lost; // no op
+ in->stream.get_capture_position = in_get_capture_position;
+ in->stream.get_active_microphones = in_get_active_microphones;
+
+ pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
+ in->dev = adev;
+ in->device = devices;
+ memcpy(&in->req_config, config, sizeof(struct audio_config));
+ memcpy(&in->pcm_config, &pcm_config_in, sizeof(struct pcm_config));
+ in->pcm_config.rate = config->sample_rate;
+ in->pcm_config.period_size = in->pcm_config.rate*IN_PERIOD_MS/1000;
+
+ in->stereo_to_mono_buf = NULL;
+ in->stereo_to_mono_buf_size = 0;
+
+ in->standby = true;
+ in->standby_position = 0;
+ in->standby_exit_time.tv_sec = 0;
+ in->standby_exit_time.tv_nsec = 0;
+ in->standby_frames_read = 0;
+
+ ret = audio_vbuffer_init(&in->buffer,
+ in->pcm_config.period_size*in->pcm_config.period_count,
+ in->pcm_config.channels *
+ pcm_format_to_bits(in->pcm_config.format) >> 3);
+ if (ret == 0) {
+ pthread_cond_init(&in->worker_wake, NULL);
+ in->worker_standby = true;
+ in->worker_exit = false;
+ pthread_create(&in->worker_thread, NULL, in_read_worker, in);
+ }
+ in->handle = handle;
+
+ pthread_mutex_lock(&adev->lock);
+ list_add_tail(&adev->in_streams, &in->stream_node);
+ pthread_mutex_unlock(&adev->lock);
*stream_in = &in->stream;
+
+error:
+ return ret;
+}
+
+
+static int adev_dump(const audio_hw_device_t *dev, int fd)
+{
return 0;
}
-static void adev_close_input_stream(struct audio_hw_device *dev,
- struct audio_stream_in *in)
+static int adev_get_microphones(const audio_hw_device_t *dev,
+ struct audio_microphone_characteristic_t *mic_array,
+ size_t *mic_count)
{
- ALOGV("adev_close_input_stream...");
- return;
-}
+ if (mic_count == NULL) {
+ return -ENOSYS;
+ }
-static int adev_dump(const audio_hw_device_t *device, int fd)
-{
- ALOGV("adev_dump");
+ if (*mic_count == 0) {
+ *mic_count = 1;
+ return 0;
+ }
+
+ if (mic_array == NULL) {
+ return -ENOSYS;
+ }
+
+ strncpy(mic_array->device_id, "mic_goldfish", AUDIO_MICROPHONE_ID_MAX_LEN - 1);
+ mic_array->device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ strncpy(mic_array->address, AUDIO_BOTTOM_MICROPHONE_ADDRESS,
+ AUDIO_DEVICE_MAX_ADDRESS_LEN - 1);
+ memset(mic_array->channel_mapping, AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED,
+ sizeof(mic_array->channel_mapping));
+ mic_array->location = AUDIO_MICROPHONE_LOCATION_UNKNOWN;
+ mic_array->group = 0;
+ mic_array->index_in_the_group = 0;
+ mic_array->sensitivity = AUDIO_MICROPHONE_SENSITIVITY_UNKNOWN;
+ mic_array->max_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
+ mic_array->min_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
+ mic_array->directionality = AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN;
+ mic_array->num_frequency_responses = 0;
+ mic_array->geometric_location.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+ mic_array->geometric_location.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+ mic_array->geometric_location.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+ mic_array->orientation.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+ mic_array->orientation.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+ mic_array->orientation.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+
+ *mic_count = 1;
return 0;
}
-static int adev_close(hw_device_t *device)
-{
- ALOGV("adev_close");
- free(device);
+static int adev_create_audio_patch(struct audio_hw_device *dev,
+ unsigned int num_sources,
+ const struct audio_port_config *sources,
+ unsigned int num_sinks,
+ const struct audio_port_config *sinks,
+ audio_patch_handle_t *handle) {
+ if (num_sources != 1 || num_sinks == 0 || num_sinks > AUDIO_PATCH_PORTS_MAX) {
+ return -EINVAL;
+ }
+
+ if (sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ // If source is a device, the number of sinks should be 1.
+ if (num_sinks != 1 || sinks[0].type != AUDIO_PORT_TYPE_MIX) {
+ return -EINVAL;
+ }
+ } else if (sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ // If source is a mix, all sinks should be device.
+ for (unsigned int i = 0; i < num_sinks; i++) {
+ if (sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+ ALOGE("%s() invalid sink type %#x for mix source", __func__, sinks[i].type);
+ return -EINVAL;
+ }
+ }
+ } else {
+ // All other cases are invalid.
+ return -EINVAL;
+ }
+
+ struct generic_audio_device* adev = (struct generic_audio_device*) dev;
+ int ret = 0;
+ bool generatedPatchHandle = false;
+ pthread_mutex_lock(&adev->lock);
+ if (*handle == AUDIO_PATCH_HANDLE_NONE) {
+ *handle = ++adev->next_patch_handle;
+ generatedPatchHandle = true;
+ }
+
+ // Only handle patches for mix->devices and device->mix case.
+ if (sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ struct generic_stream_in *in =
+ get_stream_in_by_io_handle_l(adev, sinks[0].ext.mix.handle);
+ if (in == NULL) {
+ ALOGE("%s()can not find stream with handle(%d)", __func__, sources[0].ext.mix.handle);
+ ret = -EINVAL;
+ goto error;
+ }
+
+ // Check if the patch handle match the recorded one if a valid patch handle is passed.
+ if (!generatedPatchHandle && in->patch_handle != *handle) {
+ ALOGE("%s() the patch handle(%d) does not match recorded one(%d) for stream "
+ "with handle(%d) when creating audio patch for device->mix",
+ __func__, *handle, in->patch_handle, in->handle);
+ ret = -EINVAL;
+ goto error;
+ }
+ pthread_mutex_lock(&in->lock);
+ in->device = sources[0].ext.device.type;
+ pthread_mutex_unlock(&in->lock);
+ in->patch_handle = *handle;
+ } else {
+ struct generic_stream_out *out =
+ get_stream_out_by_io_handle_l(adev, sources[0].ext.mix.handle);
+ if (out == NULL) {
+ ALOGE("%s()can not find stream with handle(%d)", __func__, sources[0].ext.mix.handle);
+ ret = -EINVAL;
+ goto error;
+ }
+
+ // Check if the patch handle match the recorded one if a valid patch handle is passed.
+ if (!generatedPatchHandle && out->patch_handle != *handle) {
+ ALOGE("%s() the patch handle(%d) does not match recorded one(%d) for stream "
+ "with handle(%d) when creating audio patch for mix->device",
+ __func__, *handle, out->patch_handle, out->handle);
+ ret = -EINVAL;
+ pthread_mutex_unlock(&out->lock);
+ goto error;
+ }
+ pthread_mutex_lock(&out->lock);
+ for (out->num_devices = 0; out->num_devices < num_sinks; out->num_devices++) {
+ out->devices[out->num_devices] = sinks[out->num_devices].ext.device.type;
+ }
+ pthread_mutex_unlock(&out->lock);
+ out->patch_handle = *handle;
+ }
+
+error:
+ if (ret != 0 && generatedPatchHandle) {
+ *handle = AUDIO_PATCH_HANDLE_NONE;
+ }
+ pthread_mutex_unlock(&adev->lock);
return 0;
}
+// This must be called with adev->lock held.
+struct generic_stream_out *get_stream_out_by_patch_handle_l(
+ struct generic_audio_device *adev, audio_patch_handle_t patch_handle) {
+ struct listnode *node;
+
+ list_for_each(node, &adev->out_streams) {
+ struct generic_stream_out *out = node_to_item(
+ node, struct generic_stream_out, stream_node);
+ if (out->patch_handle == patch_handle) {
+ return out;
+ }
+ }
+ return NULL;
+}
+
+// This must be called with adev->lock held.
+struct generic_stream_in *get_stream_in_by_patch_handle_l(
+ struct generic_audio_device *adev, audio_patch_handle_t patch_handle) {
+ struct listnode *node;
+
+ list_for_each(node, &adev->in_streams) {
+ struct generic_stream_in *in = node_to_item(
+ node, struct generic_stream_in, stream_node);
+ if (in->patch_handle == patch_handle) {
+ return in;
+ }
+ }
+ return NULL;
+}
+
+static int adev_release_audio_patch(struct audio_hw_device *dev,
+ audio_patch_handle_t patch_handle) {
+ struct generic_audio_device *adev = (struct generic_audio_device *) dev;
+
+ pthread_mutex_lock(&adev->lock);
+ struct generic_stream_out *out = get_stream_out_by_patch_handle_l(adev, patch_handle);
+ if (out != NULL) {
+ pthread_mutex_lock(&out->lock);
+ out->num_devices = 0;
+ memset(out->devices, 0, sizeof(out->devices));
+ pthread_mutex_unlock(&out->lock);
+ out->patch_handle = AUDIO_PATCH_HANDLE_NONE;
+ pthread_mutex_unlock(&adev->lock);
+ return 0;
+ }
+ struct generic_stream_in *in = get_stream_in_by_patch_handle_l(adev, patch_handle);
+ if (in != NULL) {
+ pthread_mutex_lock(&in->lock);
+ in->device = AUDIO_DEVICE_NONE;
+ pthread_mutex_unlock(&in->lock);
+ in->patch_handle = AUDIO_PATCH_HANDLE_NONE;
+ pthread_mutex_unlock(&adev->lock);
+ return 0;
+ }
+
+ pthread_mutex_unlock(&adev->lock);
+ ALOGW("%s() cannot find stream for patch handle: %d", __func__, patch_handle);
+ return -EINVAL;
+}
+
+static int adev_close(hw_device_t *dev)
+{
+ struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+ int ret = 0;
+ if (!adev)
+ return 0;
+
+ pthread_mutex_lock(&adev_init_lock);
+
+ if (audio_device_ref_count == 0) {
+ ALOGE("adev_close called when ref_count 0");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ if ((--audio_device_ref_count) == 0) {
+ if (adev->mixer) {
+ mixer_close(adev->mixer);
+ }
+ free(adev);
+ }
+
+error:
+ pthread_mutex_unlock(&adev_init_lock);
+ return ret;
+}
+
static int adev_open(const hw_module_t* module, const char* name,
- hw_device_t** device)
+ hw_device_t** device)
{
- struct alsa_audio_device *adev;
-
- ALOGV("adev_open: %s", name);
+ static struct generic_audio_device *adev;
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
- adev = calloc(1, sizeof(struct alsa_audio_device));
- if (!adev)
- return -ENOMEM;
+ pthread_mutex_lock(&adev_init_lock);
+ if (audio_device_ref_count != 0) {
+ *device = &adev->device.common;
+ audio_device_ref_count++;
+ ALOGV("%s: returning existing instance of adev", __func__);
+ ALOGV("%s: exit", __func__);
+ goto unlock;
+ }
+ adev = calloc(1, sizeof(struct generic_audio_device));
- adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
- adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
- adev->hw_device.common.module = (struct hw_module_t *) module;
- adev->hw_device.common.close = adev_close;
- adev->hw_device.init_check = adev_init_check;
- adev->hw_device.set_voice_volume = adev_set_voice_volume;
- adev->hw_device.set_master_volume = adev_set_master_volume;
- adev->hw_device.get_master_volume = adev_get_master_volume;
- adev->hw_device.set_master_mute = adev_set_master_mute;
- adev->hw_device.get_master_mute = adev_get_master_mute;
- adev->hw_device.set_mode = adev_set_mode;
- adev->hw_device.set_mic_mute = adev_set_mic_mute;
- adev->hw_device.get_mic_mute = adev_get_mic_mute;
- adev->hw_device.set_parameters = adev_set_parameters;
- adev->hw_device.get_parameters = adev_get_parameters;
- adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
- adev->hw_device.open_output_stream = adev_open_output_stream;
- adev->hw_device.close_output_stream = adev_close_output_stream;
- adev->hw_device.open_input_stream = adev_open_input_stream;
- adev->hw_device.close_input_stream = adev_close_input_stream;
- adev->hw_device.dump = adev_dump;
+ pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
- adev->devices = AUDIO_DEVICE_NONE;
+ adev->device.common.tag = HARDWARE_DEVICE_TAG;
+ adev->device.common.version = AUDIO_DEVICE_API_VERSION_3_0;
+ adev->device.common.module = (struct hw_module_t *) module;
+ adev->device.common.close = adev_close;
- *device = &adev->hw_device.common;
+ adev->device.init_check = adev_init_check; // no op
+ adev->device.set_voice_volume = adev_set_voice_volume; // no op
+ adev->device.set_master_volume = adev_set_master_volume; // no op
+ adev->device.get_master_volume = adev_get_master_volume; // no op
+ adev->device.set_master_mute = adev_set_master_mute; // no op
+ adev->device.get_master_mute = adev_get_master_mute; // no op
+ adev->device.set_mode = adev_set_mode; // no op
+ adev->device.set_mic_mute = adev_set_mic_mute;
+ adev->device.get_mic_mute = adev_get_mic_mute;
+ adev->device.set_parameters = adev_set_parameters; // no op
+ adev->device.get_parameters = adev_get_parameters; // no op
+ adev->device.get_input_buffer_size = adev_get_input_buffer_size;
+ adev->device.open_output_stream = adev_open_output_stream;
+ adev->device.close_output_stream = adev_close_output_stream;
+ adev->device.open_input_stream = adev_open_input_stream;
+ adev->device.close_input_stream = adev_close_input_stream;
+ adev->device.dump = adev_dump;
+ adev->device.get_microphones = adev_get_microphones;
+ adev->device.create_audio_patch = adev_create_audio_patch;
+ adev->device.release_audio_patch = adev_release_audio_patch;
+ *device = &adev->device.common;
+
+ adev->next_patch_handle = AUDIO_PATCH_HANDLE_NONE;
+ list_init(&adev->out_streams);
+ list_init(&adev->in_streams);
+
+ adev->mixer = mixer_open(PCM_CARD);
+ struct mixer_ctl *ctl;
+
+ // Set default mixer ctls
+ // Enable channels and set volume
+ for (int i = 0; i < (int)mixer_get_num_ctls(adev->mixer); i++) {
+ ctl = mixer_get_ctl(adev->mixer, i);
+ ALOGD("mixer %d name %s", i, mixer_ctl_get_name(ctl));
+ if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Volume") ||
+ !strcmp(mixer_ctl_get_name(ctl), "Capture Volume")) {
+ for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
+ ALOGD("set ctl %d to %d", z, 100);
+ mixer_ctl_set_percent(ctl, z, 100);
+ }
+ continue;
+ }
+ if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Switch") ||
+ !strcmp(mixer_ctl_get_name(ctl), "Capture Switch")) {
+ for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
+ ALOGD("set ctl %d to %d", z, 1);
+ mixer_ctl_set_value(ctl, z, 1);
+ }
+ continue;
+ }
+ }
+
+ audio_device_ref_count++;
+
+unlock:
+ pthread_mutex_unlock(&adev_init_lock);
return 0;
}
@@ -684,7 +1829,7 @@
.module_api_version = AUDIO_MODULE_API_VERSION_0_1,
.hal_api_version = HARDWARE_HAL_API_VERSION,
.id = AUDIO_HARDWARE_MODULE_ID,
- .name = "Generic Audio HAL for dragonboards",
+ .name = "Generic audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},