audio: low latency playback optimizations
Add support to configure input and output streams as
legacy (faster than realtime) or realtime
Change-Id: Idebe3624bd14d5384b8c42d4f74d1874537b6028
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index b95ddf6..5b449bb 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -15,6 +15,7 @@
*/
#define LOG_TAG "audio_hw_primary"
+#define ATRACE_TAG ATRACE_TAG_AUDIO
/*#define LOG_NDEBUG 0*/
/*#define VERY_VERY_VERBOSE_LOGGING*/
#ifdef VERY_VERY_VERBOSE_LOGGING
@@ -34,6 +35,7 @@
#include <sys/prctl.h>
#include <cutils/log.h>
+#include <cutils/trace.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <cutils/atomic.h>
@@ -73,6 +75,8 @@
#define STR(x) #x
#endif
+#define ULL_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000)
+
static unsigned int configured_low_latency_capture_period_size =
LOW_LATENCY_CAPTURE_PERIOD_SIZE;
@@ -103,6 +107,20 @@
.avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
};
+static int af_period_multiplier = 4;
+struct pcm_config pcm_config_rt = {
+ .channels = DEFAULT_CHANNEL_COUNT,
+ .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+ .period_size = ULL_PERIOD_SIZE, //1 ms
+ .period_count = 512, //=> buffer size is 512ms
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = ULL_PERIOD_SIZE*8, //8ms
+ .stop_threshold = INT_MAX,
+ .silence_threshold = 0,
+ .silence_size = 0,
+ .avail_min = ULL_PERIOD_SIZE, //1 ms
+};
+
struct pcm_config pcm_config_hdmi_multi = {
.channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
.rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
@@ -122,6 +140,19 @@
.avail_min = 0,
};
+struct pcm_config pcm_config_audio_capture_rt = {
+ .channels = DEFAULT_CHANNEL_COUNT,
+ .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+ .period_size = ULL_PERIOD_SIZE,
+ .period_count = 512,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 0,
+ .stop_threshold = INT_MAX,
+ .silence_threshold = 0,
+ .silence_size = 0,
+ .avail_min = ULL_PERIOD_SIZE, //1 ms
+};
+
#define AFE_PROXY_CHANNEL_COUNT 2
#define AFE_PROXY_SAMPLING_RATE 48000
@@ -201,6 +232,120 @@
static pthread_mutex_t adev_init_lock;
static unsigned int audio_device_ref_count;
+static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
+ int flags __unused)
+{
+ int dir = 0;
+ switch (uc_id) {
+ case USECASE_AUDIO_RECORD_LOW_LATENCY:
+ dir = 1;
+ case USECASE_AUDIO_PLAYBACK_ULL:
+ break;
+ default:
+ return false;
+ }
+
+ int dev_id = platform_get_pcm_device_id(uc_id, dir == 0 ?
+ PCM_PLAYBACK : PCM_CAPTURE);
+ if (adev->adm_is_noirq_avail)
+ return adev->adm_is_noirq_avail(adev->adm_data,
+ adev->snd_card, dev_id, dir);
+ return false;
+}
+
+static void register_out_stream(struct stream_out *out)
+{
+ struct audio_device *adev = out->dev;
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD)
+ return;
+
+ if (!adev->adm_register_output_stream)
+ return;
+
+ adev->adm_register_output_stream(adev->adm_data,
+ out->handle,
+ out->flags);
+
+ if (!adev->adm_set_config)
+ return;
+
+ if (out->realtime) {
+ adev->adm_set_config(adev->adm_data,
+ out->handle,
+ out->pcm, &out->config);
+ }
+}
+
+static void register_in_stream(struct stream_in *in)
+{
+ struct audio_device *adev = in->dev;
+ if (!adev->adm_register_input_stream)
+ return;
+
+ adev->adm_register_input_stream(adev->adm_data,
+ in->capture_handle,
+ in->flags);
+
+ if (!adev->adm_set_config)
+ return;
+
+ if (in->realtime) {
+ adev->adm_set_config(adev->adm_data,
+ in->capture_handle,
+ in->pcm,
+ &in->config);
+ }
+}
+
+static void request_out_focus(struct stream_out *out, long ns)
+{
+ struct audio_device *adev = out->dev;
+
+ if (out->routing_change) {
+ out->routing_change = false;
+ if (adev->adm_on_routing_change)
+ adev->adm_on_routing_change(adev->adm_data, out->handle);
+ }
+
+ if (adev->adm_request_focus_v2) {
+ adev->adm_request_focus_v2(adev->adm_data, out->handle, ns);
+ } else if (adev->adm_request_focus) {
+ adev->adm_request_focus(adev->adm_data, out->handle);
+ }
+}
+
+static void request_in_focus(struct stream_in *in, long ns)
+{
+ struct audio_device *adev = in->dev;
+
+ if (in->routing_change) {
+ in->routing_change = false;
+ if (adev->adm_on_routing_change)
+ adev->adm_on_routing_change(adev->adm_data, in->capture_handle);
+ }
+
+ if (adev->adm_request_focus_v2) {
+ adev->adm_request_focus_v2(adev->adm_data, in->capture_handle, ns);
+ } else if (adev->adm_request_focus) {
+ adev->adm_request_focus(adev->adm_data, in->capture_handle);
+ }
+}
+
+static void release_out_focus(struct stream_out *out, long ns __unused)
+{
+ struct audio_device *adev = out->dev;
+
+ if (adev->adm_abandon_focus)
+ adev->adm_abandon_focus(adev->adm_data, out->handle);
+}
+
+static void release_in_focus(struct stream_in *in, long ns __unused)
+{
+ struct audio_device *adev = in->dev;
+ if (adev->adm_abandon_focus)
+ adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+}
+
__attribute__ ((visibility ("default")))
bool audio_hw_send_gain_dep_calibration(int level) {
bool ret_val = false;
@@ -236,6 +381,12 @@
return false;
}
+static inline bool is_mmap_usecase(audio_usecase_t uc_id)
+{
+ return (uc_id == USECASE_AUDIO_RECORD_AFE_PROXY) ||
+ (uc_id == USECASE_AUDIO_PLAYBACK_AFE_PROXY);
+}
+
static int get_snd_codec_id(audio_format_t format)
{
int id = 0;
@@ -862,6 +1013,8 @@
if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
flags |= PCM_MMAP | PCM_NOIRQ;
pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ } else if (in->realtime) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
}
while (1) {
@@ -891,7 +1044,10 @@
in->pcm = NULL;
goto error_open;
}
-
+ register_in_stream(in);
+ if (in->realtime) {
+ ret = pcm_start(in->pcm);
+ }
audio_extn_perf_lock_release();
ALOGV("%s: exit", __func__);
@@ -1225,9 +1381,12 @@
if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
unsigned int flags = PCM_OUT;
unsigned int pcm_open_retry_count = 0;
+
if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
flags |= PCM_MMAP | PCM_NOIRQ;
pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ } else if (out->realtime) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
} else
flags |= PCM_MONOTONIC;
@@ -1278,9 +1437,14 @@
if (adev->offload_effects_start_output != NULL)
adev->offload_effects_start_output(out->handle, out->pcm_device_id);
}
+ ret = 0;
+ register_out_stream(out);
+ if (out->realtime) {
+ ret = pcm_start(out->pcm);
+ }
audio_extn_perf_lock_release();
ALOGV("%s: exit", __func__);
- return 0;
+ return ret;
error_open:
audio_extn_perf_lock_release();
stop_output_stream(out);
@@ -1369,7 +1533,7 @@
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
return out->compr_config.fragment_size;
}
- return out->config.period_size *
+ return out->config.period_size * out->af_period_multiplier *
audio_stream_out_frame_size((const struct audio_stream_out *)stream);
}
@@ -1522,11 +1686,18 @@
* Because select_devices() must be called to switch back the music
* playback to headset.
*/
- if (val != 0) {
- out->devices = val;
+ audio_devices_t new_dev = val;
+ if (new_dev != AUDIO_DEVICE_NONE) {
+ bool same_dev = out->devices == new_dev;
+ out->devices = new_dev;
- if (!out->standby)
+ if (!out->standby) {
+ if (!same_dev) {
+ ALOGV("update routing change");
+ out->routing_change = true;
+ }
select_devices(adev, out->usecase);
+ }
if (output_drives_call(adev, out)) {
if (!voice_is_in_call(adev)) {
@@ -1598,10 +1769,19 @@
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
+ uint32_t hw_delay, period_ms;
struct stream_out *out = (struct stream_out *)stream;
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD)
return COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
+ else if (out->realtime) {
+ // since the buffer won't be filled up faster than realtime,
+ // return a smaller number
+ period_ms = (out->af_period_multiplier * out->config.period_size *
+ 1000) / (out->config.rate);
+ hw_delay = platform_render_latency(out->usecase)/1000;
+ return period_ms + hw_delay;
+ }
return (out->config.period_count * out->config.period_size * 1000) /
(out->config.rate);
@@ -1679,8 +1859,6 @@
out->standby = true;
goto exit;
}
- if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD && adev->adm_register_output_stream)
- adev->adm_register_output_stream(adev->adm_data, out->handle, out->flags);
}
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
@@ -1724,20 +1902,21 @@
memset((void *)buffer, 0, bytes);
ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
- if (adev->adm_request_focus)
- adev->adm_request_focus(adev->adm_data, out->handle);
- if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
+ long ns = pcm_bytes_to_frames(out->pcm, bytes)*1000000000LL/
+ out->config.rate;
+ request_out_focus(out, ns);
+
+ bool use_mmap = is_mmap_usecase(out->usecase) || out->realtime;
+ if (use_mmap)
ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
- }
else
ret = pcm_write(out->pcm, (void *)buffer, bytes);
if (ret == 0)
out->written += bytes / (out->config.channels * sizeof(short));
- if (adev->adm_abandon_focus)
- adev->adm_abandon_focus(adev->adm_data, out->handle);
+ release_out_focus(out, ns);
}
}
@@ -1931,8 +2110,8 @@
{
struct stream_in *in = (struct stream_in *)stream;
- return in->config.period_size *
- audio_stream_in_frame_size((const struct audio_stream_in *)stream);
+ return in->config.period_size * in->af_period_multiplier *
+ audio_stream_in_frame_size((const struct audio_stream_in *)stream);
}
static uint32_t in_get_channels(const struct audio_stream *stream)
@@ -2025,8 +2204,11 @@
if (((int)in->device != val) && (val != 0)) {
in->device = val;
/* If recording is in progress, change the tx device to new device */
- if (!in->standby)
- status = select_devices(adev, in->usecase);
+ if (!in->standby) {
+ ALOGV("update input routing change");
+ in->routing_change = true;
+ select_devices(adev, in->usecase);
+ }
}
}
@@ -2074,22 +2256,27 @@
goto exit;
}
in->standby = 0;
- if (adev->adm_register_input_stream)
- adev->adm_register_input_stream(adev->adm_data, in->capture_handle, in->flags);
}
- if (adev->adm_request_focus)
- adev->adm_request_focus(adev->adm_data, in->capture_handle);
+ //what's the duration requested by the client?
+ long ns = pcm_bytes_to_frames(in->pcm, bytes)*1000000000LL/
+ in->config.rate;
+ request_in_focus(in, ns);
+ bool use_mmap = is_mmap_usecase(in->usecase) || in->realtime;
if (in->pcm) {
- if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
+ if (use_mmap) {
ret = pcm_mmap_read(in->pcm, buffer, bytes);
} else
ret = pcm_read(in->pcm, buffer, bytes);
+
+ if (ret < 0) {
+ ALOGE("Failed to read w/err %s", strerror(errno));
+ ret = -errno;
+ }
}
- if (adev->adm_abandon_focus)
- adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+ release_in_focus(in, ns);
/*
* Instead of writing zeroes here, we could trust the hardware
@@ -2342,7 +2529,9 @@
out->config = pcm_config_deep_buffer;
} else if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
- out->config = pcm_config_low_latency;
+ out->realtime = may_use_noirq_mode(adev, USECASE_AUDIO_PLAYBACK_ULL, out->flags);
+ out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
+ out->config = out->realtime ? pcm_config_rt : pcm_config_low_latency;
} else {
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
out->config = pcm_config_low_latency;
@@ -2410,6 +2599,7 @@
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
+ out->af_period_multiplier = out->realtime ? af_period_multiplier : 1;
out->standby = 1;
/* out->muted = false; by calloc() */
/* out->written = 0; by calloc() */
@@ -2719,18 +2909,29 @@
#if LOW_LATENCY_CAPTURE_USE_CASE
in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
#endif
+ in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags);
}
- in->config = pcm_config_audio_capture;
- frame_size = audio_stream_in_frame_size(&in->stream);
- buffer_size = get_input_buffer_size(config->sample_rate,
- config->format,
- channel_count,
+ in->config = in->realtime ? pcm_config_audio_capture_rt :
+ pcm_config_audio_capture;
+
+ if (!in->realtime) {
+ frame_size = audio_stream_in_frame_size(&in->stream);
+ buffer_size = get_input_buffer_size(config->sample_rate,
+ config->format,
+ channel_count,
is_low_latency);
- in->config.period_size = buffer_size / frame_size;
+ in->config.period_size = buffer_size / frame_size;
+ } // period size is left untouched for rt mode playback
}
+
in->config.channels = channel_count;
- in->config.rate = config->sample_rate;
+ if (in->realtime) {
+ in->af_period_multiplier = af_period_multiplier;
+ } else {
+ in->config.rate = config->sample_rate;
+ in->af_period_multiplier = 1;
+ }
/* This stream could be for sound trigger lab,
get sound trigger pcm if present */
@@ -3062,6 +3263,14 @@
dlsym(adev->adm_lib, "adm_request_focus");
adev->adm_abandon_focus = (adm_abandon_focus_t)
dlsym(adev->adm_lib, "adm_abandon_focus");
+ adev->adm_set_config = (adm_set_config_t)
+ dlsym(adev->adm_lib, "adm_set_config");
+ adev->adm_request_focus_v2 = (adm_request_focus_v2_t)
+ dlsym(adev->adm_lib, "adm_request_focus_v2");
+ adev->adm_is_noirq_avail = (adm_is_noirq_avail_t)
+ dlsym(adev->adm_lib, "adm_is_noirq_avail");
+ adev->adm_on_routing_change = (adm_on_routing_change_t)
+ dlsym(adev->adm_lib, "adm_on_routing_change");
}
adev->bt_wb_speech_enabled = false;
@@ -3091,6 +3300,17 @@
}
audio_device_ref_count++;
+
+ if (property_get("audio_hal.period_multiplier", value, NULL) > 0) {
+ af_period_multiplier = atoi(value);
+ if (af_period_multiplier < 0) {
+ af_period_multiplier = 2;
+ } else if (af_period_multiplier > 4) {
+ af_period_multiplier = 4;
+ }
+ ALOGV("new period_multiplier = %d", af_period_multiplier);
+ }
+
pthread_mutex_unlock(&adev_init_lock);
if (adev->adm_init)
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 1a1e0ef..64ac794 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -177,7 +177,9 @@
void *offload_cookie;
struct compr_gapless_mdata gapless_mdata;
int send_new_metadata;
-
+ bool realtime;
+ int af_period_multiplier;
+ bool routing_change;
struct audio_device *dev;
};
@@ -201,7 +203,9 @@
audio_input_flags_t flags;
bool is_st_session;
bool is_st_session_active;
-
+ bool realtime;
+ int af_period_multiplier;
+ bool routing_change;
struct audio_device *dev;
};
@@ -234,6 +238,12 @@
typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t);
typedef void (*adm_request_focus_t)(void *, audio_io_handle_t);
typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t);
+typedef void (*adm_set_config_t)(void *, audio_io_handle_t,
+ struct pcm *,
+ struct pcm_config *);
+typedef void (*adm_request_focus_v2_t)(void *, audio_io_handle_t, long);
+typedef bool (*adm_is_noirq_avail_t)(void *, int, int, int);
+typedef void (*adm_on_routing_change_t)(void *, audio_io_handle_t);
struct audio_device {
struct audio_hw_device device;
@@ -285,10 +295,13 @@
adm_deregister_stream_t adm_deregister_stream;
adm_request_focus_t adm_request_focus;
adm_abandon_focus_t adm_abandon_focus;
+ adm_set_config_t adm_set_config;
+ adm_request_focus_v2_t adm_request_focus_v2;
+ adm_is_noirq_avail_t adm_is_noirq_avail;
+ adm_on_routing_change_t adm_on_routing_change;
/* logging */
snd_device_t last_logged_snd_device[AUDIO_USECASE_MAX][2]; /* [out, in] */
-
};
int select_devices(struct audio_device *adev,
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index f4f2f73..789824b 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -491,6 +491,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+#define ULL_PLATFORM_DELAY (7*1000LL)
static pthread_once_t check_op_once_ctl = PTHREAD_ONCE_INIT;
static bool is_tmus = false;
@@ -2524,6 +2525,8 @@
return DEEP_BUFFER_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
return LOW_LATENCY_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_ULL:
+ return ULL_PLATFORM_DELAY;
default:
return 0;
}
