audio: Copy generic HAL and bt-audio configs from Cuttlefish am: 456949d476
Change-Id: I4e1806c4f3fe9fb691239f0913535b8c91688b5f
diff --git a/audio/Android.mk b/audio/Android.mk
index afa804f..228ccf2 100644
--- a/audio/Android.mk
+++ b/audio/Android.mk
@@ -24,8 +24,7 @@
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
-LOCAL_MODULE_PATH_32 := $(TARGET_OUT_VENDOR)/lib/hw
-LOCAL_MODULE_PATH_64 := $(TARGET_OUT_VENDOR)/lib64/hw
+LOCAL_MODULE_RELATIVE_PATH := hw
LOCAL_VENDOR_MODULE := true
LOCAL_SRC_FILES := audio_hw.c
@@ -38,4 +37,3 @@
system/media/audio_effects/include
include $(BUILD_SHARED_LIBRARY)
-
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,
},
diff --git a/device-common.mk b/device-common.mk
index 7be64f9..41eaa72 100644
--- a/device-common.mk
+++ b/device-common.mk
@@ -121,6 +121,7 @@
$(LOCAL_PATH)/etc/audio_policy_configuration_bluetooth_legacy_hal.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration_bluetooth_legacy_hal.xml \
frameworks/av/services/audiopolicy/config/a2dp_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/a2dp_audio_policy_configuration.xml \
frameworks/av/services/audiopolicy/config/a2dp_in_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/a2dp_in_audio_policy_configuration.xml \
+ frameworks/av/services/audiopolicy/config/primary_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/primary_audio_policy_configuration.xml \
frameworks/av/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/bluetooth_audio_policy_configuration.xml \
frameworks/av/services/audiopolicy/config/r_submix_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/r_submix_audio_policy_configuration.xml \
frameworks/av/services/audiopolicy/config/usb_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/usb_audio_policy_configuration.xml \
@@ -130,6 +131,7 @@
# Copy media codecs config file
PRODUCT_COPY_FILES += \
$(LOCAL_PATH)/etc/media_codecs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/media_codecs.xml \
+ frameworks/av/media/libeffects/data/audio_effects.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.xml \
frameworks/av/media/libstagefright/data/media_codecs_google_video.xml:$(TARGET_COPY_OUT_VENDOR)/etc/media_codecs_google_video.xml \
frameworks/av/media/libstagefright/data/media_codecs_google_audio.xml:$(TARGET_COPY_OUT_VENDOR)/etc/media_codecs_google_audio.xml
diff --git a/manifest.xml b/manifest.xml
index 0ff0eb2..fee2945 100644
--- a/manifest.xml
+++ b/manifest.xml
@@ -27,6 +27,24 @@
</interface>
</hal>
<hal format="hidl">
+ <name>android.hardware.bluetooth.a2dp</name>
+ <transport>hwbinder</transport>
+ <version>1.0</version>
+ <interface>
+ <name>IBluetoothAudioOffload</name>
+ <instance>default</instance>
+ </interface>
+ </hal>
+ <hal format="hidl">
+ <name>android.hardware.bluetooth.audio</name>
+ <transport>hwbinder</transport>
+ <version>2.0</version>
+ <interface>
+ <name>IBluetoothAudioProvidersFactory</name>
+ <instance>default</instance>
+ </interface>
+ </hal>
+ <hal format="hidl">
<name>android.hardware.configstore</name>
<transport>hwbinder</transport>
<version>1.1</version>