/*
* Copyright (C) 2013 Texas Instruments
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* 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.
*/
#define LOG_TAG "audio_hw_primary"
//#define LOG_NDEBUG 0
//#define VERY_VERBOSE_LOGGING
#ifdef VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
#include <errno.h>
#include <stdint.h>
#include <stdlib.h>
#include <sys/time.h>
#include <cutils/log.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <audio_utils/resampler.h>
#include <audio_route/audio_route.h>
#include <system/audio.h>
#include <hardware/hardware.h>
#include <hardware/audio.h>
#include <hardware/audio_effect.h>
#include <tinyalsa/asoundlib.h>
/* yet another definition of ARRAY_SIZE macro) */
#define ARRAY_SIZE(x) (sizeof(x)/sizeof(x[0]))
/*
* additional space in resampler buffer allowing for extra samples to be returned
* by speex resampler when sample rates ratio is not an integer
*/
#define RESAMPLER_HEADROOM_FRAMES 10
/* buffer_remix: functor for doing in-place buffer manipulations.
*
* NB. When remix_func is called, the memory at `buf` must be at least
* as large as frames * sample_size * MAX(in_chans, out_chans).
*/
struct buffer_remix {
void (*remix_func)(struct buffer_remix *data, void *buf, size_t frames);
size_t sample_size; /* size of one audio sample, in bytes */
size_t in_chans; /* number of input channels */
size_t out_chans; /* number of output channels */
};
struct j6_voice_stream {
struct j6_audio_device *dev;
struct pcm *pcm_in;
struct pcm *pcm_out;
struct pcm_config in_config;
struct pcm_config out_config;
struct resampler_itfe *resampler;
struct resampler_buffer_provider buf_provider;
struct buffer_remix *remix;
pthread_t thread;
int16_t *in_buffer;
int16_t *out_buffer;
size_t in_frames;
size_t out_frames;
size_t frame_size;
char *name;
};
struct j6_voice {
struct j6_voice_stream ul;
struct j6_voice_stream dl;
};
struct j6_audio_device {
struct audio_hw_device device;
struct j6_stream_in *in;
struct j6_stream_out *out;
struct j6_voice voice;
struct audio_route *route;
audio_devices_t in_device;
audio_devices_t out_device;
pthread_mutex_t lock;
unsigned int card;
unsigned int in_port;
unsigned int out_port;
unsigned int bt_port;
bool mic_mute;
bool in_call;
audio_mode_t mode;
};
struct j6_stream_in {
struct audio_stream_in stream;
struct j6_audio_device *dev;
struct pcm_config config;
struct pcm *pcm;
struct buffer_remix *remix; /* adapt hw chan count to client */
struct resampler_itfe *resampler;
struct resampler_buffer_provider buf_provider;
int16_t *buffer;
size_t frames_in;
size_t hw_frame_size;
unsigned int requested_rate;
unsigned int requested_channels;
int read_status;
pthread_mutex_t lock;
bool standby;
};
struct j6_stream_out {
struct audio_stream_out stream;
struct j6_audio_device *dev;
struct pcm_config config;
struct pcm *pcm;
struct timespec last;
pthread_mutex_t lock;
bool standby;
int64_t written; /* total frames written, not cleared when entering standby */
};
static const char *supported_cards[] = {
"dra7evm",
"VayuEVM",
"DRA7xxEVM",
};
#define SUPPORTED_IN_DEVICES (AUDIO_DEVICE_IN_BUILTIN_MIC | \
AUDIO_DEVICE_IN_WIRED_HEADSET | \
AUDIO_DEVICE_IN_DEFAULT)
#define SUPPORTED_OUT_DEVICES (AUDIO_DEVICE_OUT_SPEAKER | \
AUDIO_DEVICE_OUT_WIRED_HEADSET | \
AUDIO_DEVICE_OUT_WIRED_HEADPHONE | \
AUDIO_DEVICE_OUT_DEFAULT)
#define CAPTURE_SAMPLE_RATE 44100
#define CAPTURE_PERIOD_SIZE 960
#define CAPTURE_PERIOD_COUNT 4
#define CAPTURE_BUFFER_SIZE (CAPTURE_PERIOD_SIZE * CAPTURE_PERIOD_COUNT)
#define PLAYBACK_SAMPLE_RATE 44100
#define PLAYBACK_PERIOD_SIZE 960
#define PLAYBACK_PERIOD_COUNT 4
#define PLAYBACK_BUFFER_SIZE (PLAYBACK_PERIOD_SIZE * PLAYBACK_PERIOD_COUNT)
#define BT_SAMPLE_RATE 8000
#define BT_PERIOD_SIZE 160
#define BT_PERIOD_COUNT 4
#define BT_BUFFER_SIZE (BT_PERIOD_SIZE * BT_PERIOD_COUNT)
struct pcm_config pcm_config_capture = {
.channels = 2,
.rate = CAPTURE_SAMPLE_RATE,
.format = PCM_FORMAT_S16_LE,
.period_size = CAPTURE_PERIOD_SIZE,
.period_count = CAPTURE_PERIOD_COUNT,
.start_threshold = 1,
.stop_threshold = CAPTURE_BUFFER_SIZE,
};
struct pcm_config pcm_config_playback = {
.channels = 2,
.rate = PLAYBACK_SAMPLE_RATE,
.format = PCM_FORMAT_S16_LE,
.period_size = PLAYBACK_PERIOD_SIZE,
.period_count = PLAYBACK_PERIOD_COUNT,
.start_threshold = PLAYBACK_BUFFER_SIZE / 2,
.stop_threshold = PLAYBACK_BUFFER_SIZE,
.avail_min = PLAYBACK_PERIOD_SIZE,
};
struct pcm_config pcm_config_bt_in = {
.channels = 2,
.rate = BT_SAMPLE_RATE,
.format = PCM_FORMAT_S16_LE,
.period_size = BT_PERIOD_SIZE,
.period_count = BT_PERIOD_COUNT,
.start_threshold = 1,
.stop_threshold = BT_BUFFER_SIZE,
};
struct pcm_config pcm_config_bt_out = {
.channels = 2,
.rate = BT_SAMPLE_RATE,
.format = PCM_FORMAT_S16_LE,
.period_size = BT_PERIOD_SIZE,
.period_count = BT_PERIOD_COUNT,
.start_threshold = BT_BUFFER_SIZE / 2,
.stop_threshold = BT_BUFFER_SIZE,
.avail_min = BT_PERIOD_SIZE,
};
static int find_supported_card(void)
{
char name[256] = "";
int card = 0;
int found = 0;
unsigned int i;
#ifdef OMAP_ENHANCEMENT
do {
/* returns an error after last valid card */
int ret = mixer_get_card_name(card, name, sizeof(name));
if (ret)
break;
for (i = 0; i < ARRAY_SIZE(supported_cards); ++i) {
if (supported_cards[i] && !strcmp(name, supported_cards[i])) {
ALOGV("Supported card '%s' found at %d", name, card);
found = 1;
break;
}
}
} while (!found && (card++ < MAX_CARD_COUNT));
#endif
/* Use default card number if not found */
if (!found)
card = 0;
return card;
}
static void do_out_standby(struct j6_stream_out *out);
/* must be called with device lock held */
static void select_input_device(struct j6_audio_device *adev)
{
if (adev->in_device & ~SUPPORTED_IN_DEVICES)
ALOGW("select_input_device() device not supported, will use default device");
}
/* must be called with device lock held */
static void select_output_device(struct j6_audio_device *adev)
{
if (adev->out_device & ~SUPPORTED_OUT_DEVICES)
ALOGW("select_output_device() device(s) not supported, will use default devices");
}
static size_t get_input_buffer_size(uint32_t sample_rate, int format, int channel_count)
{
size_t size;
/*
* take resampling into account and return the closest majoring
* multiple of 16 frames, as audioflinger expects audio buffers to
* be a multiple of 16 frames
*/
size = (pcm_config_capture.period_size * sample_rate) / pcm_config_capture.rate;
size = ((size + 15) / 16) * 16;
return size * channel_count * audio_bytes_per_sample(format);
}
/*
* Implementation of buffer_remix::remix_func that removes
* channels in place without doing any other processing. The
* extra channels are truncated.
*/
static void remove_channels_from_buf(struct buffer_remix *data, void *buf, size_t frames)
{
size_t samp_size, in_frame, out_frame;
size_t N, c;
char *s, *d;
ALOGVV("remove_channels_from_buf() remix=%p buf=%p frames=%u",
data, buf, frames);
if (frames == 0)
return;
samp_size = data->sample_size;
in_frame = data->in_chans * samp_size;
out_frame = data->out_chans * samp_size;
if (out_frame >= in_frame) {
ALOGE("BUG: remove_channels_from_buf() can not add channels to a buffer.\n");
return;
}
N = frames - 1;
d = (char*)buf + out_frame;
s = (char*)buf + in_frame;
/* take the first several channels and truncate the rest */
while (N--) {
for (c = 0; c < out_frame; ++c)
d[c] = s[c];
d += out_frame;
s += in_frame;
}
}
static int setup_stereo_to_mono_input_remix(struct j6_stream_in *in)
{
ALOGV("setup_stereo_to_mono_input_remix() stream=%p", in);
struct buffer_remix *br = (struct buffer_remix *)malloc(sizeof(struct buffer_remix));
if (!br)
return -ENOMEM;
br->remix_func = remove_channels_from_buf;
br->sample_size = sizeof(int16_t);
br->in_chans = 2;
br->out_chans = 1;
in->remix = br;
return 0;
}
/*
* Implementation of buffer_remix::remix_func that duplicates the first
* channel into the rest of channels in the frame without doing any other
* processing. It assumes data in 16-bits, but it's not explicitly checked
*/
static void mono_remix(struct buffer_remix *data, void *buf, size_t frames)
{
int16_t *buffer = (int16_t*)buf;
size_t i;
ALOGVV("mono_remix() remix=%p buf=%p frames=%u", data, buf, frames);
if (frames == 0)
return;
/* duplicate first channel into the rest of channels in the frame */
while (frames--) {
for (i = 1; i < data->out_chans; i++)
buffer[i] = buffer[0];
buffer += data->out_chans;
}
}
static int setup_mono_input_remix(struct j6_voice_stream *stream)
{
ALOGV("setup_mono_input_remix() %s stream", stream->name);
struct buffer_remix *br = (struct buffer_remix *)malloc(sizeof(struct buffer_remix));
if (!br)
return -ENOMEM;
br->remix_func = mono_remix;
br->sample_size = sizeof(int16_t);
br->in_chans = stream->in_config.channels;
br->out_chans = stream->out_config.channels;
stream->remix = br;
return 0;
}
static int voice_get_next_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
struct j6_voice_stream *stream;
int ret;
if (buffer_provider == NULL || buffer == NULL) {
ALOGE("voice_get_next_buffer() invalid buffer/provider");
return -EINVAL;
}
stream = (struct j6_voice_stream *)((char *)buffer_provider -
offsetof(struct j6_voice_stream, buf_provider));
if (stream->pcm_in == NULL) {
buffer->raw = NULL;
buffer->frame_count = 0;
return -ENODEV;
}
if (buffer->frame_count > stream->in_frames) {
ALOGW("voice_get_next_buffer() %s unexpected frame count %u, "
"buffer was allocated for %u frames",
stream->name, buffer->frame_count, stream->in_frames);
buffer->frame_count = stream->in_frames;
}
ret = pcm_read(stream->pcm_in, stream->in_buffer,
buffer->frame_count * stream->frame_size);
if (ret) {
ALOGE("voice_get_next_buffer() failed to read %s: %s",
stream->name, pcm_get_error(stream->pcm_in));
buffer->raw = NULL;
buffer->frame_count = 0;
return ret;
}
buffer->i16 = stream->in_buffer;
return ret;
}
static void voice_release_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
}
static void *voice_thread_func(void *arg)
{
struct j6_voice_stream *stream = (struct j6_voice_stream *)arg;
struct j6_audio_device *adev = stream->dev;
struct timespec now;
size_t frames = stream->out_frames;
uint32_t periods = 0;
uint32_t avail;
bool in_steady = false;
bool out_steady = false;
int ret = 0;
pcm_start(stream->pcm_in);
memset(stream->out_buffer, 0, stream->out_frames * stream->frame_size);
while (adev->in_call) {
if (out_steady) {
if (in_steady) {
stream->resampler->resample_from_provider(stream->resampler,
stream->out_buffer,
&frames);
} else {
ret = pcm_get_htimestamp(stream->pcm_in, &avail, &now);
if (!ret && (avail > 0)) {
in_steady = true;
continue;
}
}
} else if (++periods == stream->out_config.period_count) {
out_steady = true;
}
if (stream->remix)
stream->remix->remix_func(stream->remix, stream->out_buffer, frames);
ret = pcm_write(stream->pcm_out, stream->out_buffer,
frames * stream->frame_size);
if (ret) {
ALOGE("voice_thread_func() failed to write %s: %s",
stream->name, pcm_get_error(stream->pcm_out));
usleep((frames * 1000000) / stream->out_config.rate);
}
}
return (void*)ret;
}
static void voice_stream_exit(struct j6_voice_stream *stream)
{
if (stream->resampler) {
release_resampler(stream->resampler);
stream->resampler = NULL;
}
if (stream->pcm_out) {
pcm_close(stream->pcm_out);
stream->pcm_out = NULL;
}
if (stream->pcm_in) {
pcm_close(stream->pcm_in);
stream->pcm_in = NULL;
}
if (stream->in_buffer) {
free(stream->in_buffer);
stream->in_buffer = NULL;
stream->in_frames = 0;
}
if (stream->out_buffer) {
free(stream->out_buffer);
stream->out_buffer = NULL;
stream->out_frames = 0;
}
if (stream->remix) {
free(stream->remix);
stream->remix = NULL;
}
if (stream->name) {
free(stream->name);
stream->name = NULL;
}
}
static int voice_stream_init(struct j6_voice_stream *stream,
unsigned int in_port,
unsigned int out_port,
bool needs_mono_remix)
{
struct j6_audio_device *adev = stream->dev;
int ret;
stream->buf_provider.get_next_buffer = voice_get_next_buffer;
stream->buf_provider.release_buffer = voice_release_buffer;
ret = create_resampler(stream->in_config.rate,
stream->out_config.rate,
2,
RESAMPLER_QUALITY_DEFAULT,
&stream->buf_provider,
&stream->resampler);
if (ret) {
ALOGE("voice_stream_init() failed to create %s resampler %d", stream->name, ret);
return ret;
}
stream->pcm_in = pcm_open(adev->card, in_port, PCM_IN, &stream->in_config);
stream->pcm_out = pcm_open(adev->card, out_port, PCM_OUT, &stream->out_config);
if (!pcm_is_ready(stream->pcm_in) || !pcm_is_ready(stream->pcm_out)) {
ALOGE("voice_stream_init() failed to open pcm %s devices", stream->name);
voice_stream_exit(stream);
return -ENODEV;
}
stream->frame_size = pcm_frames_to_bytes(stream->pcm_in, 1);
/* out_buffer will store the resampled data */
stream->out_frames = stream->out_config.period_size;
stream->out_buffer = malloc(stream->out_frames * stream->frame_size);
/* in_buffer will store the frames recorded from the PCM device */
stream->in_frames = (stream->out_frames * stream->in_config.rate) / stream->out_config.rate +
RESAMPLER_HEADROOM_FRAMES;
stream->in_buffer = malloc(stream->in_frames * stream->frame_size);
if (!stream->in_buffer || !stream->out_buffer) {
ALOGE("voice_stream_init() failed to allocate %s buffers", stream->name);
voice_stream_exit(stream);
return -ENOMEM;
}
if (needs_mono_remix) {
ret = setup_mono_input_remix(stream);
if (ret) {
ALOGE("voice_stream_init() failed to setup mono remix %d", ret);
voice_stream_exit(stream);
return ret;
}
} else {
stream->remix = NULL;
}
return 0;
}
static int enter_voice_call(struct j6_audio_device *adev)
{
struct j6_voice *voice = &adev->voice;
int ret;
ALOGI("enter_voice_call() entering bluetooth voice call");
audio_route_apply_path(adev->route, "BT SCO Master");
audio_route_update_mixer(adev->route);
/* Let the primary output switch to a dummy sink */
if (adev->out)
do_out_standby(adev->out);
/* Uplink: Mic (44.1kHz) -> BT (8kHz) */
voice->ul.name = strdup("UL");
voice->ul.in_config = pcm_config_capture;
voice->ul.out_config = pcm_config_bt_out;
voice->ul.dev = adev;
ret = voice_stream_init(&voice->ul, adev->in_port, adev->bt_port, false);
if (ret) {
ALOGE("enter_voice_call() failed to init uplink %d", ret);
goto err_ul_init;
}
/* Downlink: BT (8kHz) -> HP/Spk (44.1kHz) */
voice->dl.name = strdup("DL");
voice->dl.in_config = pcm_config_bt_in;
voice->dl.out_config = pcm_config_playback;
voice->dl.dev = adev;
ret = voice_stream_init(&voice->dl, adev->bt_port, adev->out_port, true);
if (ret) {
ALOGE("enter_voice_call() failed to init downlink %d", ret);
goto err_dl_init;
}
adev->in_call = true;
/* Create uplink thread: Mic -> BT */
ret = pthread_create(&voice->ul.thread, NULL, voice_thread_func, &voice->ul);
if (ret) {
ALOGE("enter_voice_call() failed to create uplink thread %d", ret);
adev->in_call = false;
goto err_ul_thread;
}
/* Create downlink thread: BT -> HP/Spk */
ret = pthread_create(&voice->dl.thread, NULL, voice_thread_func, &voice->dl);
if (ret) {
ALOGE("enter_voice_call() failed to create downlink thread %d", ret);
adev->in_call = false;
goto err_dl_thread;
}
return 0;
err_dl_thread:
pthread_join(voice->ul.thread, NULL);
err_ul_thread:
voice_stream_exit(&voice->ul);
err_dl_init:
voice_stream_exit(&voice->dl);
err_ul_init:
audio_route_reset_path(adev->route, "BT SCO Master");
audio_route_update_mixer(adev->route);
return ret;
}
static void leave_voice_call(struct j6_audio_device *adev)
{
struct j6_voice *voice = &adev->voice;
struct j6_voice_stream *ul = &voice->ul;
struct j6_voice_stream *dl = &voice->dl;
void *ret;
ALOGI("leave_voice_call() leaving bluetooth voice call");
adev->in_call = false;
/*
* The PCM ports used for Bluetooth are slaves and they can lose the
* BCLK and FSYNC while still active. That leads to blocking read() and
* write() calls, which is prevented by switching the clock source to
* an internal one and explicitly stopping both ports for the new source
* to take effect at kernel level
*/
audio_route_reset_path(adev->route, "BT SCO Master");
audio_route_update_mixer(adev->route);
if (ul->pcm_out)
pcm_stop(ul->pcm_out);
if (dl->pcm_in)
pcm_stop(dl->pcm_in);
pthread_join(voice->dl.thread, &ret);
pthread_join(voice->ul.thread, &ret);
voice_stream_exit(&voice->dl);
voice_stream_exit(&voice->ul);
/* Let the primary output switch back to its ALSA PCM device */
if (adev->out)
do_out_standby(adev->out);
}
static uint32_t time_diff(struct timespec t1, struct timespec t0)
{
struct timespec temp;
if ((t1.tv_nsec - t0.tv_nsec) < 0) {
temp.tv_sec = t1.tv_sec - t0.tv_sec-1;
temp.tv_nsec = 1000000000UL + t1.tv_nsec - t0.tv_nsec;
} else {
temp.tv_sec = t1.tv_sec - t0.tv_sec;
temp.tv_nsec = t1.tv_nsec - t0.tv_nsec;
}
return (temp.tv_sec * 1000000UL + temp.tv_nsec / 1000);
}
/* audio HAL functions */
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
uint32_t rate = PLAYBACK_SAMPLE_RATE;
ALOGVV("out_get_sample_rate() stream=%p rate=%u", stream, rate);
return rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGVV("out_set_sample_rate() stream=%p rate=%u", stream, rate);
return 0;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
const struct audio_stream_out *s = (const struct audio_stream_out *)stream;
uint32_t frames = ((PLAYBACK_PERIOD_SIZE + 15) / 16) * 16;
size_t bytes = frames * audio_stream_out_frame_size(s);
ALOGVV("out_get_buffer_size() stream=%p frames=%u bytes=%u", stream, frames, bytes);
return bytes;
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
audio_channel_mask_t channels = AUDIO_CHANNEL_OUT_STEREO;
ALOGVV("out_get_channels() stream=%p channels=%u", stream, popcount(channels));
return channels;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
ALOGVV("out_set_format() stream=%p format=0x%08x (%u bits/sample)",
stream, format, audio_bytes_per_sample(format) << 3);
return format;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
ALOGVV("out_set_format() stream=%p format=0x%08x (%u bits/sample)",
stream, format, audio_bytes_per_sample(format) << 3);
if (format != AUDIO_FORMAT_PCM_16_BIT) {
return -ENOSYS;
} else {
return 0;
}
}
/* must be called with locks held */
static void do_out_standby(struct j6_stream_out *out)
{
struct j6_audio_device *adev = out->dev;
if (!out->standby) {
if (adev->mode != AUDIO_MODE_IN_CALL) {
ALOGI("do_out_standby() close card %u port %u", adev->card, adev->out_port);
pcm_close(out->pcm);
out->pcm = NULL;
} else {
ALOGI("do_out_standby() close dummy card");
}
out->standby = true;
}
}
static int out_standby(struct audio_stream *stream)
{
struct j6_stream_out *out = (struct j6_stream_out *)(stream);
struct j6_audio_device *adev = out->dev;
ALOGV("out_standby() stream=%p", out);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
do_out_standby(out);
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
return 0;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct j6_stream_out *out = (struct j6_stream_out *)(stream);
struct j6_audio_device *adev = out->dev;
struct str_parms *parms;
char value[32];
int ret;
uint32_t val = 0;
ALOGV("out_set_parameters() stream=%p parameter='%s'", out, kvpairs);
parms = str_parms_create_str(kvpairs);
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 (val != 0) {
if ((adev->out_device & AUDIO_DEVICE_OUT_ALL) != val)
do_out_standby(out);
/* set the active output device */
adev->out_device = val;
select_output_device(adev);
}
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
}
return 0;
}
static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
{
return strdup("");
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
const struct j6_stream_out *out = (const struct j6_stream_out *)(stream);
uint32_t frames = PLAYBACK_BUFFER_SIZE;
uint32_t latency = (frames * 1000) / PLAYBACK_SAMPLE_RATE;
ALOGVV("out_get_latency() stream=%p latency=%u msecs", out, latency);
return latency;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
return -ENOSYS;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
const struct audio_stream_out *s = (const struct audio_stream_out *)stream;
struct j6_stream_out *out = (struct j6_stream_out *)(stream);
struct j6_audio_device *adev = out->dev;
struct timespec now;
const size_t frame_size = audio_stream_out_frame_size(s);
const size_t frames = bytes / frame_size;
uint32_t rate = out->config.rate;
uint32_t write_usecs = frames * 1000000 / rate;
uint32_t diff_usecs;
int ret = 0;
ALOGVV("out_write() stream=%p buffer=%p size=%u/%u time=%u usecs",
out, buffer, frames, rate, write_usecs);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
if (!adev->in_call) {
select_output_device(adev);
ALOGI("out_write() open card %u port %u", adev->card, adev->out_port);
out->pcm = pcm_open(adev->card, adev->out_port, PCM_OUT, &out->config);
if (!pcm_is_ready(out->pcm)) {
ALOGE("out_write() failed to open pcm out: %s", pcm_get_error(out->pcm));
pcm_close(out->pcm);
out->pcm = NULL;
ret = -ENODEV;
}
} else {
ALOGI("out_write() open dummy port");
clock_gettime(CLOCK_REALTIME, &out->last);
}
if (ret) {
usleep(write_usecs); /* limits the rate of error messages */
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
return ret;
}
out->standby = false;
}
pthread_mutex_unlock(&adev->lock);
if (!adev->in_call) {
ret = pcm_write(out->pcm, buffer, bytes);
if (ret) {
ALOGE("out_write() failed to write audio data %d", ret);
usleep(write_usecs); /* limits the rate of error messages */
}
} else {
clock_gettime(CLOCK_REALTIME, &now);
diff_usecs = time_diff(now, out->last);
if (write_usecs > diff_usecs)
usleep(write_usecs - diff_usecs);
clock_gettime(CLOCK_REALTIME, &out->last);
}
out->written += frames;
pthread_mutex_unlock(&out->lock);
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
return -EINVAL;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
int64_t *timestamp)
{
return -EINVAL;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
struct j6_stream_out *out = (struct j6_stream_out *)(stream);
struct j6_audio_device *adev = out->dev;
int64_t signed_frames = -1;
size_t avail;
int ret = -1;
pthread_mutex_lock(&out->lock);
if (!adev->in_call) {
if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
signed_frames = out->written - pcm_get_buffer_size(out->pcm) + avail;
}
} else {
clock_gettime(CLOCK_REALTIME, timestamp);
signed_frames = out->written +
(time_diff(*timestamp, out->last) * out->config.rate) / 1000000;
}
/* It would be unusual for this value to be negative, but check just in case ... */
if (signed_frames >= 0) {
*frames = signed_frames;
ret = 0;
}
pthread_mutex_unlock(&out->lock);
return ret;
}
/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
const struct j6_stream_in *in = (const struct j6_stream_in *)(stream);
ALOGVV("in_get_sample_rate() stream=%p rate=%u", stream, in->requested_rate);
return in->requested_rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("in_set_sample_rate() stream=%p rate=%u", stream, rate);
return 0;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
const struct j6_stream_in *in = (const struct j6_stream_in *)(stream);
size_t bytes = get_input_buffer_size(in->requested_rate,
AUDIO_FORMAT_PCM_16_BIT,
in->requested_channels);
ALOGVV("in_get_buffer_size() stream=%p bytes=%u", in, bytes);
return bytes;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
const struct j6_stream_in *in = (const struct j6_stream_in *)(stream);
audio_channel_mask_t channels = audio_channel_in_mask_from_count(in->requested_channels);
ALOGVV("in_get_channels() stream=%p channels=%u", in, in->requested_channels);
return channels;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
ALOGVV("in_set_format() stream=%p format=0x%08x (%u bits/sample)",
stream, format, audio_bytes_per_sample(format) << 3);
return format;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
ALOGV("in_set_format() stream=%p format=0x%08x (%u bits/sample)",
stream, format, audio_bytes_per_sample(format) << 3);
if (format != AUDIO_FORMAT_PCM_16_BIT) {
return -ENOSYS;
} else {
return 0;
}
}
/* must be called with locks held */
static void do_in_standby(struct j6_stream_in *in)
{
struct j6_audio_device *adev = in->dev;
if (!in->standby) {
ALOGI("do_in_standby() close card %u port %u", adev->card, adev->out_port);
pcm_close(in->pcm);
in->pcm = NULL;
in->standby = true;
}
}
static int in_standby(struct audio_stream *stream)
{
struct j6_stream_in *in = (struct j6_stream_in *)(stream);
struct j6_audio_device *adev = in->dev;
ALOGV("in_standby() stream=%p", in);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&in->lock);
do_in_standby(in);
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&adev->lock);
return 0;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct j6_stream_in *in = (struct j6_stream_in *)(stream);
struct j6_audio_device *adev = in->dev;
struct str_parms *parms;
char value[32];
int ret;
uint32_t val = 0;
ALOGV("in_set_parameters() stream=%p parameter='%s'", stream, kvpairs);
parms = str_parms_create_str(kvpairs);
/* Nothing to do for AUDIO_PARAMETER_STREAM_INPUT_SOURCE, so it's ignored */
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(&in->lock);
if (val != 0) {
if ((adev->in_device & AUDIO_DEVICE_IN_ALL) != val)
do_in_standby(in);
/* set the active input device */
adev->in_device = val;
select_input_device(adev);
}
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&adev->lock);
}
return 0;
}
static char * in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
return strdup("");
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
return 0;
}
static int get_next_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
struct j6_stream_in *in;
struct buffer_remix *remix;
if (buffer_provider == NULL || buffer == NULL)
return -EINVAL;
in = (struct j6_stream_in *)((char *)buffer_provider -
offsetof(struct j6_stream_in, buf_provider));
if (in->pcm == NULL) {
buffer->raw = NULL;
buffer->frame_count = 0;
in->read_status = -ENODEV;
return -ENODEV;
}
if (in->frames_in == 0) {
in->read_status = pcm_read(in->pcm,
(void*)in->buffer,
buffer->frame_count * in->hw_frame_size);
if (in->read_status != 0) {
ALOGE("get_next_buffer() pcm_read error %d", in->read_status);
buffer->raw = NULL;
buffer->frame_count = 0;
return in->read_status;
}
in->frames_in = buffer->frame_count;
remix = in->remix;
if (remix)
remix->remix_func(remix, in->buffer, in->frames_in);
}
buffer->frame_count = (buffer->frame_count > in->frames_in) ?
in->frames_in : buffer->frame_count;
buffer->i16 = in->buffer;
return in->read_status;
}
static void release_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
struct j6_stream_in *in;
if (buffer_provider == NULL || buffer == NULL)
return;
in = (struct j6_stream_in *)((char *)buffer_provider -
offsetof(struct j6_stream_in, buf_provider));
in->frames_in -= buffer->frame_count;
}
/*
* read_frames() reads frames from kernel driver, down samples to capture rate
* if necessary and output the number of frames requested to the buffer specified
*/
static ssize_t read_frames(struct j6_stream_in *in, void *buffer, ssize_t frames)
{
const struct audio_stream_in *s = (const struct audio_stream_in *)in;
ssize_t frames_wr = 0;
size_t frame_size;
ALOGVV("read_frames() stream=%p frames=%u", in, frames);
if (in->remix)
frame_size = audio_stream_in_frame_size(s);
else
frame_size = in->hw_frame_size;
while (frames_wr < frames) {
size_t frames_rd = frames - frames_wr;
in->resampler->resample_from_provider(in->resampler,
(int16_t *)((char *)buffer + frames_wr * frame_size),
&frames_rd);
/* in->read_status is updated by getNextBuffer() also called by
* in->resampler->resample_from_provider() */
if (in->read_status != 0)
return in->read_status;
frames_wr += frames_rd;
}
return frames_wr;
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
const struct audio_stream_in *s = (const struct audio_stream_in *)stream;
struct j6_stream_in *in = (struct j6_stream_in *)(stream);
struct j6_audio_device *adev = in->dev;
const size_t frame_size = audio_stream_in_frame_size(stream);
const size_t frames = bytes / frame_size;
uint32_t rate = in_get_sample_rate(&stream->common);
uint32_t read_usecs = frames * 1000000 / rate;
int ret;
ALOGVV("in_read() stream=%p buffer=%p size=%u/%u time=%u usecs",
stream, buffer, frames, rate, read_usecs);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&in->lock);
if (in->standby) {
select_input_device(adev);
ALOGI("in_read() open card %u port %u", adev->card, adev->in_port);
in->pcm = pcm_open(adev->card, adev->in_port, PCM_IN, &in->config);
if (!pcm_is_ready(in->pcm)) {
ALOGE("in_read() failed to open pcm in: %s", pcm_get_error(in->pcm));
pcm_close(in->pcm);
in->pcm = NULL;
usleep(read_usecs); /* limits the rate of error messages */
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&adev->lock);
return -ENODEV;
}
/* if no supported sample rate is available, use the resampler */
if (in->resampler) {
in->resampler->reset(in->resampler);
in->frames_in = 0;
}
in->standby = false;
}
pthread_mutex_unlock(&adev->lock);
if (in->resampler || in->remix)
ret = read_frames(in, buffer, frames);
else
ret = pcm_read(in->pcm, buffer, bytes);
if (ret < 0) {
ALOGE("in_read() failed to read audio data %d", ret);
usleep(read_usecs); /* limits the rate of error messages */
memset(buffer, 0, bytes);
} else if (adev->mic_mute) {
memset(buffer, 0, bytes);
}
pthread_mutex_unlock(&in->lock);
return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
ALOGVV("in_get_input_frames_lost() stream=%p frames=%u", stream, 0);
return 0;
}
static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
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)
{
struct j6_audio_device *adev = (struct j6_audio_device *)dev;
struct j6_stream_out *out;
out = (struct j6_stream_out *)malloc(sizeof(struct j6_stream_out));
if (!out)
return -ENOMEM;
ALOGV("adev_open_output_stream() stream=%p rate=%u channels=%u "
"format=0x%08x flags=0x%08x",
out, config->sample_rate, popcount(config->channel_mask),
config->format, flags);
pthread_mutex_init(&out->lock, NULL);
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
out->stream.common.get_channels = out_get_channels;
out->stream.common.get_format = out_get_format;
out->stream.common.set_format = out_set_format;
out->stream.common.standby = out_standby;
out->stream.common.dump = out_dump;
out->stream.common.set_parameters = out_set_parameters;
out->stream.common.get_parameters = out_get_parameters;
out->stream.common.add_audio_effect = out_add_audio_effect;
out->stream.common.remove_audio_effect = out_remove_audio_effect;
out->stream.get_latency = out_get_latency;
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->dev = adev;
out->standby = true;
out->config = pcm_config_playback;
out->written = 0;
adev->out = out;
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);
*stream_out = &out->stream;
return 0;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
struct j6_audio_device *adev = (struct j6_audio_device *)dev;
struct j6_stream_out *out = (struct j6_stream_out *)(stream);
ALOGV("adev_close_output_stream() stream=%p", out);
out_standby(&stream->common);
out->dev = NULL;
adev->out = NULL;
free(stream);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
return -ENOSYS;
}
static char * adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
return strdup("");;
}
static int adev_init_check(const struct audio_hw_device *dev)
{
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
return -ENOSYS;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
return -ENOSYS;
}
static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
{
return -ENOSYS;
}
static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
{
return -ENOSYS;
}
static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
{
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
struct j6_audio_device *adev = (struct j6_audio_device *)dev;
struct j6_stream_out *out = adev->out;
int ret = 0;
ALOGV("adev_set_mode() mode=0x%08x", mode);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (adev->mode == mode) {
ALOGV("adev_set_mode() already in mode=0x%08x", mode);
goto out;
}
if (mode == AUDIO_MODE_IN_CALL) {
ret = enter_voice_call(adev);
if (ret) {
ALOGE("adev_set_mode() failed to initialize voice call %d", ret);
goto out;
}
} else if (adev->mode == AUDIO_MODE_IN_CALL) {
leave_voice_call(adev);
}
adev->mode = mode;
out:
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
return ret;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
struct j6_audio_device *adev = (struct j6_audio_device *)dev;
ALOGV("adev_set_mic_mute() state=%s", state ? "mute" : "unmute");
adev->mic_mute = state;
return 0;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
const struct j6_audio_device *adev = (const struct j6_audio_device *)dev;
*state = adev->mic_mute;
ALOGV("adev_get_mic_mute() state=%s", *state ? "mute" : "unmute");
return 0;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
size_t bytes = get_input_buffer_size(config->sample_rate,
config->format,
popcount(config->channel_mask));
ALOGVV("adev_in_get_buffer_size() bytes=%u", bytes);
return bytes;
}
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 j6_audio_device *adev = (struct j6_audio_device *)dev;
struct j6_stream_in *in;
int ret;
in = (struct j6_stream_in *)malloc(sizeof(struct j6_stream_in));
if (!in)
return -ENOMEM;
ALOGV("adev_open_input_stream() stream=%p rate=%u channels=%u format=0x%08x",
in, config->sample_rate, popcount(config->channel_mask), config->format);
pthread_mutex_init(&in->lock, NULL);
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
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.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.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
in->dev = adev;
in->standby = true;
in->config = pcm_config_capture;
in->requested_rate = config->sample_rate;
in->requested_channels = popcount(config->channel_mask);
in->hw_frame_size = in->config.channels * sizeof(int16_t);
in->remix = NULL;
in->resampler = NULL;
in->buffer = NULL;
adev->in = in;
/* in-place stereo-to-mono remix since capture stream is stereo */
if (in->requested_channels == 1) {
ALOGV("adev_open_input_stream() stereo-to-mono remix needed");
/*
* buffer size is already enough to allow stereo-to-mono remix
* and resample if needed
*/
in->buffer = malloc(2 * in->config.period_size * in->hw_frame_size);
if (!in->buffer) {
ret = -ENOMEM;
goto err1;
}
ret = setup_stereo_to_mono_input_remix(in);
if (ret) {
ALOGE("adev_open_input_stream() failed to setup remix %d", ret);
goto err2;
}
}
if (in->requested_rate != in->config.rate) {
ALOGV("adev_open_input_stream() resample needed, req=%uHz got=%uHz",
in->requested_rate, in->config.rate);
in->buf_provider.get_next_buffer = get_next_buffer;
in->buf_provider.release_buffer = release_buffer;
ret = create_resampler(in->config.rate,
in->requested_rate,
in->requested_channels,
RESAMPLER_QUALITY_DEFAULT,
&in->buf_provider,
&in->resampler);
if (ret) {
ALOGE("adev_open_input_stream() failed to create resampler %d", ret);
goto err3;
}
}
*stream_in = &in->stream;
return 0;
err3:
free(in->remix);
err2:
free(in->buffer);
err1:
free(in);
return ret;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream)
{
struct j6_audio_device *adev = (struct j6_audio_device *)dev;
struct j6_stream_in *in = (struct j6_stream_in *)(stream);
ALOGV("adev_close_input_stream() stream=%p", stream);
in_standby(&stream->common);
if (in->resampler)
release_resampler(in->resampler);
in->resampler = NULL;
if (in->remix)
free(in->remix);
in->remix = NULL;
in->dev = NULL;
adev->in = NULL;
free(in->buffer);
free(in);
}
static int adev_dump(const audio_hw_device_t *device, int fd)
{
return 0;
}
static int adev_close(hw_device_t *device)
{
struct j6_audio_device *adev = (struct j6_audio_device *)device;
ALOGI("adev_close()");
audio_route_free(adev->route);
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct j6_audio_device *adev;
ALOGI("adev_open() %s", name);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
adev = (struct j6_audio_device*)malloc(sizeof(struct j6_audio_device));
if (!adev)
return -ENOMEM;
pthread_mutex_init(&adev->lock, NULL);
adev->device.common.tag = HARDWARE_DEVICE_TAG;
adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->device.common.module = (struct hw_module_t *) module;
adev->device.common.close = adev_close;
adev->device.init_check = adev_init_check;
adev->device.set_voice_volume = adev_set_voice_volume;
adev->device.set_master_volume = adev_set_master_volume;
adev->device.get_master_volume = adev_get_master_volume;
adev->device.set_master_mute = adev_set_master_mute;
adev->device.get_master_mute = adev_get_master_mute;
adev->device.set_mode = adev_set_mode;
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;
adev->device.get_parameters = adev_get_parameters;
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->in_device = AUDIO_DEVICE_IN_BUILTIN_MIC;
adev->out_device = AUDIO_DEVICE_OUT_SPEAKER;
adev->card = find_supported_card();
adev->in_port = 0;
adev->out_port = 0;
adev->bt_port = 2;
adev->mic_mute = false;
adev->in_call = false;
adev->mode = AUDIO_MODE_NORMAL;
adev->route = audio_route_init(adev->card, NULL);
if (!adev->route) {
ALOGE("Unable to initialize audio routes");
free(adev);
return -EINVAL;
}
*device = &adev->device.common;
return 0;
}
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
struct audio_module HAL_MODULE_INFO_SYM = {
.common = {
.tag = HARDWARE_MODULE_TAG,
.module_api_version = AUDIO_MODULE_API_VERSION_0_1,
.hal_api_version = HARDWARE_HAL_API_VERSION,
.id = AUDIO_HARDWARE_MODULE_ID,
.name = "Jacinto6 Audio HAL",
.author = "Texas Instruments Inc.",
.methods = &hal_module_methods,
},
};
