size_t LibmediaPlayback::FillBuffer(void* data, size_t size) {
CHECK(data);
if (in_file_) {
int frames_read = 0;
int frame_size = audio_bytes_per_sample(audio_format_) * num_channels_;
int sample_size = audio_bytes_per_sample(audio_format_);
if (sample_size == 1) {
void* temp = malloc(size * sizeof(short));
frames_read = sf_readf_short(in_file_, reinterpret_cast<short*>(temp),
size / frame_size);
int num_samples = frames_read * frame_size / sample_size;
memcpy_to_u8_from_i16(reinterpret_cast<uint8_t*>(data),
reinterpret_cast<int16_t*>(temp), num_samples);
free(temp);
} else if (sample_size == 2) {
frames_read = sf_readf_short(in_file_, reinterpret_cast<short*>(data),
size / frame_size);
} else if (sample_size == 4) {
frames_read = sf_readf_int(in_file_, reinterpret_cast<int*>(data),
size / frame_size);
} else {
LOG(ERROR) << "Could not handle file with sample size = " << sample_size;
return 0;
}
size = frame_size * frames_read;
} else {
size = (size < sine_data_buffer_->size()) ? size :
sine_data_buffer_->size();
memcpy(data, sine_data_buffer_->data(), size);
}
return size;
}
size_t AudioRecord::frameSize() const
{
if (inputSource() == AUDIO_SOURCE_VOICE_COMMUNICATION) {
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
} else {
return channelCount()*sizeof(int16_t);
}
} else {
if (format() ==AUDIO_FORMAT_AMR_NB) {
return channelCount() * AMR_FRAMESIZE; // Full rate framesize
} else if (format() == AUDIO_FORMAT_EVRC) {
return channelCount() * EVRC_FRAMESIZE; // Full rate framesize
} else if (format() == AUDIO_FORMAT_QCELP) {
return channelCount() * QCELP_FRAMESIZE; // Full rate framesize
} else if (format() == AUDIO_FORMAT_AAC) {
// Not actual framsize but for variable frame rate AAC encoding,
// buffer size is treated as a frame size
return AAC_FRAMESIZE;
} else if(format() == AUDIO_FORMAT_AMR_WB) {
return channelCount() * AMR_WB_FRAMESIZE;
}
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
} else {
return sizeof(uint8_t);
}
}
}
bool AudioPort::isBetterFormatMatch(audio_format_t newFormat,
audio_format_t currentFormat,
audio_format_t targetFormat)
{
if (newFormat == currentFormat) {
return false;
}
if (currentFormat == AUDIO_FORMAT_INVALID) {
return true;
}
if (newFormat == targetFormat) {
return true;
}
int currentDiffBytes = (int)audio_bytes_per_sample(targetFormat) -
audio_bytes_per_sample(currentFormat);
int newDiffBytes = (int)audio_bytes_per_sample(targetFormat) -
audio_bytes_per_sample(newFormat);
if (abs(newDiffBytes) < abs(currentDiffBytes)) {
return true;
} else if (abs(newDiffBytes) == abs(currentDiffBytes)) {
return (newDiffBytes >= 0);
}
return false;
}
ReformatBufferProvider::ReformatBufferProvider(int32_t channelCount,
audio_format_t inputFormat, audio_format_t outputFormat,
size_t bufferFrameCount) :
CopyBufferProvider(
channelCount * audio_bytes_per_sample(inputFormat),
channelCount * audio_bytes_per_sample(outputFormat),
bufferFrameCount),
mChannelCount(channelCount),
mInputFormat(inputFormat),
mOutputFormat(outputFormat)
{
ALOGV("ReformatBufferProvider(%p)(%u, %#x, %#x)",
this, channelCount, inputFormat, outputFormat);
}
// static
status_t AudioRecord::getMinFrameCount(
int* frameCount,
uint32_t sampleRate,
int format,
int channelCount)
{
size_t size = 0;
if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size)
!= NO_ERROR) {
LOGE("AudioSystem could not query the input buffer size.");
return NO_INIT;
}
if (size == 0) {
LOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d",
sampleRate, format, channelCount);
return BAD_VALUE;
}
// We double the size of input buffer for ping pong use of record buffer.
size <<= 1;
if (audio_is_linear_pcm(format)) {
size /= channelCount * audio_bytes_per_sample(format);
}
*frameCount = size;
return NO_ERROR;
}
// static
status_t AudioRecord::getMinFrameCount(
size_t* frameCount,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask)
{
if (frameCount == NULL) {
return BAD_VALUE;
}
size_t size;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
if (status != NO_ERROR) {
ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
"channelMask %#x; status %d", sampleRate, format, channelMask, status);
return status;
}
// handle non-linear-pcm formats and update frameCount
if (!audio_is_linear_pcm(format)) {
*frameCount = (size * 2) / sizeof(uint8_t);
return NO_ERROR;
}
// We double the size of input buffer for ping pong use of record buffer.
// Assumes audio_is_linear_pcm(format)
if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) *
audio_bytes_per_sample(format))) == 0) {
ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
return NO_ERROR;
}
static size_t audio_hw_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
struct tinyalsa_audio_device *device;
struct tinyalsa_mixer_io_props *mixer_props;
size_t size;
int channel_count = popcount(config->channel_mask);
ALOGD("%s(%p, %d, %d, %d)++", __func__, dev, config->sample_rate, config->format, channel_count);
if(dev == NULL)
return -1;
device = (struct tinyalsa_audio_device *) dev;
if(device->mixer == NULL)
return -1;
mixer_props = tinyalsa_mixer_get_input_props(device->mixer);
if(mixer_props == NULL)
return -1;
// Default value
if(mixer_props->rate == 0)
mixer_props->rate = 44100;
size = (mixer_props->period_size * config->sample_rate) / mixer_props->rate;
size = ((size + 15) / 16) * 16;
size = size * channel_count * audio_bytes_per_sample(config->format);
ALOGD("%s(%p, %d, %d, %d)--", __func__, dev, config->sample_rate, config->format, channel_count);
return size;
}
int AudioTrack::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
} else {
return sizeof(uint8_t);
}
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
{
int ret;
struct stream_out *out = (struct stream_out *)stream;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
ret = start_output_stream(out);
if (ret != 0) {
pthread_mutex_unlock(&out->dev->lock);
goto err;
}
out->standby = false;
}
pthread_mutex_unlock(&out->dev->lock);
alsa_device_proxy* proxy = &out->proxy;
const void * write_buff = buffer;
int num_write_buff_bytes = bytes;
const int num_device_channels = proxy_get_channel_count(proxy); /* what we told alsa */
const int num_req_channels = out->hal_channel_count; /* what we told AudioFlinger */
if (num_device_channels != num_req_channels) {
/* allocate buffer */
const size_t required_conversion_buffer_size =
bytes * num_device_channels / num_req_channels;
if (required_conversion_buffer_size > out->conversion_buffer_size) {
out->conversion_buffer_size = required_conversion_buffer_size;
out->conversion_buffer = realloc(out->conversion_buffer,
out->conversion_buffer_size);
}
/* convert data */
const audio_format_t audio_format = out_get_format(&(out->stream.common));
const unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
num_write_buff_bytes =
adjust_channels(write_buff, num_req_channels,
out->conversion_buffer, num_device_channels,
sample_size_in_bytes, num_write_buff_bytes);
write_buff = out->conversion_buffer;
}
if (write_buff != NULL && num_write_buff_bytes != 0) {
proxy_write(&out->proxy, write_buff, num_write_buff_bytes);
}
pthread_mutex_unlock(&out->lock);
return bytes;
err:
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common));
}
return bytes;
}
RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask,
audio_channel_mask_t outputChannelMask, audio_format_t format,
size_t bufferFrameCount) :
CopyBufferProvider(
audio_bytes_per_sample(format)
* audio_channel_count_from_out_mask(inputChannelMask),
audio_bytes_per_sample(format)
* audio_channel_count_from_out_mask(outputChannelMask),
bufferFrameCount),
mFormat(format),
mSampleSize(audio_bytes_per_sample(format)),
mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)),
mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask))
{
ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
this, format, inputChannelMask, outputChannelMask,
mInputChannels, mOutputChannels);
(void) memcpy_by_index_array_initialization_from_channel_mask(
mIdxAry, ARRAY_SIZE(mIdxAry), outputChannelMask, inputChannelMask);
}
void PowerLog::log(const void *buffer, size_t frames, int64_t nowNs)
{
std::lock_guard<std::mutex> guard(mLock);
const size_t bytes_per_sample = audio_bytes_per_sample(mFormat);
while (frames > 0) {
// check partial computation
size_t required = mFramesPerEntry - mCurrentFrames;
size_t process = std::min(required, frames);
if (mCurrentTime == 0) {
mCurrentTime = nowNs;
}
mCurrentEnergy +=
audio_utils_compute_energy_mono(buffer, mFormat, process * mChannelCount);
mCurrentFrames += process;
ALOGV("nowNs:%lld, required:%zu, process:%zu, mCurrentEnergy:%f, mCurrentFrames:%zu",
(long long)nowNs, required, process, mCurrentEnergy, mCurrentFrames);
if (process < required) {
return;
}
// We store the data as normalized energy per sample. The energy sequence is
// zero terminated. Consecutive zeroes are ignored.
if (mCurrentEnergy == 0.f) {
if (mConsecutiveZeroes++ == 0) {
mEntries[mIdx++] = std::make_pair(nowNs, 0.f);
// zero terminate the signal sequence.
}
} else {
mConsecutiveZeroes = 0;
mEntries[mIdx++] = std::make_pair(mCurrentTime, mCurrentEnergy);
ALOGV("writing %lld %f", (long long)mCurrentTime, mCurrentEnergy);
}
if (mIdx >= mEntries.size()) {
mIdx -= mEntries.size();
}
mCurrentTime = 0;
mCurrentEnergy = 0;
mCurrentFrames = 0;
frames -= process;
buffer = (const uint8_t *)buffer + mCurrentFrames * mChannelCount * bytes_per_sample;
}
}
// static
status_t AudioRecord::getMinFrameCount(
size_t* frameCount,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask)
{
if (frameCount == NULL) {
return BAD_VALUE;
}
// default to 0 in case of error
*frameCount = 0;
size_t size = 0;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
if (status != NO_ERROR) {
ALOGE("AudioSystem could not query the input buffer size; status %d", status);
return NO_INIT;
}
if (size == 0) {
ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
// We double the size of input buffer for ping pong use of record buffer.
size <<= 1;
uint32_t channelCount = popcount(channelMask);
#ifdef QCOM_HARDWARE
if (audio_is_linear_pcm(format))
#endif
size /= channelCount * audio_bytes_per_sample(format);
#ifdef QCOM_HARDWARE
else
size /= sizeof(uint8_t);
#endif
*frameCount = size;
return NO_ERROR;
}
// Play audio from a WAV file.
//
// Parameters:
// out_stream: A pointer to the output audio stream.
// in_file: A pointer to a SNDFILE object.
// config: A pointer to struct that contains audio configuration data.
//
// Returns: An int which has a non-negative number on success.
int PlayFile(audio_stream_out_t* out_stream, SNDFILE* in_file,
audio_config_t* config) {
size_t buffer_size = out_stream->common.get_buffer_size(&out_stream->common);
size_t kFrameSize =
audio_bytes_per_sample(kAudioPlaybackFormat) *
audio_channel_count_from_out_mask(config->channel_mask);
short* data = new short[buffer_size / kFrameSize];
int rc = 0;
sf_count_t frames_read = 1;
while (frames_read != 0) {
size_t bytes_wanted =
out_stream->common.get_buffer_size(&out_stream->common);
frames_read = sf_readf_short(in_file, data, bytes_wanted / kFrameSize);
rc = out_stream->write(out_stream, data, frames_read * kFrameSize);
if (rc < 0) {
LOG(ERROR) << "Writing data to hal failed. (" << strerror(rc) << ")";
break;
}
}
return rc;
}
TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) :
mChannelCount(channelCount),
mFormat(format),
mSampleRate(sampleRate),
mFrameSize(channelCount * audio_bytes_per_sample(format)),
mLocalBufferFrameCount(0),
mLocalBufferData(NULL),
mRemaining(0),
mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),
mFallbackFailErrorShown(false),
mAudioPlaybackRateValid(false)
{
LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,
"TimestretchBufferProvider can't allocate Sonic stream");
setPlaybackRate(playbackRate);
ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f %d %d)",
this, channelCount, format, sampleRate, playbackRate.mSpeed,
playbackRate.mPitch, playbackRate.mStretchMode, playbackRate.mFallbackMode);
mBuffer.frameCount = 0;
}
void memcpy_by_audio_format(void *dst, audio_format_t dst_format,
const void *src, audio_format_t src_format, size_t count)
{
/* default cases for error falls through to fatal log below. */
if (dst_format == src_format) {
switch (dst_format) {
case AUDIO_FORMAT_PCM_16_BIT:
case AUDIO_FORMAT_PCM_FLOAT:
case AUDIO_FORMAT_PCM_8_BIT:
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
case AUDIO_FORMAT_PCM_32_BIT:
case AUDIO_FORMAT_PCM_8_24_BIT:
memcpy(dst, src, count * audio_bytes_per_sample(dst_format));
return;
default:
break;
}
}
switch (dst_format) {
case AUDIO_FORMAT_PCM_16_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_i16_from_float((int16_t*)dst, (float*)src, count);
return;
case AUDIO_FORMAT_PCM_8_BIT:
memcpy_to_i16_from_u8((int16_t*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
memcpy_to_i16_from_p24((int16_t*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_32_BIT:
memcpy_to_i16_from_i32((int16_t*)dst, (int32_t*)src, count);
return;
case AUDIO_FORMAT_PCM_8_24_BIT:
memcpy_to_i16_from_q8_23((int16_t*)dst, (int32_t*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_FLOAT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_float_from_i16((float*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_8_BIT:
memcpy_to_float_from_u8((float*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
memcpy_to_float_from_p24((float*)dst, (uint8_t*)src, count);
return;
case AUDIO_FORMAT_PCM_32_BIT:
memcpy_to_float_from_i32((float*)dst, (int32_t*)src, count);
return;
case AUDIO_FORMAT_PCM_8_24_BIT:
memcpy_to_float_from_q8_23((float*)dst, (int32_t*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_8_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_u8_from_i16((uint8_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_u8_from_float((uint8_t*)dst, (float*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_p24_from_i16((uint8_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_p24_from_float((uint8_t*)dst, (float*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_32_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_i32_from_i16((int32_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_i32_from_float((int32_t*)dst, (float*)src, count);
return;
default:
break;
}
break;
case AUDIO_FORMAT_PCM_8_24_BIT:
switch (src_format) {
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_q8_23_from_i16((int32_t*)dst, (int16_t*)src, count);
return;
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_q8_23_from_float_with_clamp((int32_t*)dst, (float*)src, count);
return;
case AUDIO_FORMAT_PCM_24_BIT_PACKED: {
memcpy_to_q8_23_from_p24((int32_t *)dst, (uint8_t *)src, count);
return;
}
default:
break;
}
break;
default:
break;
}
LOG_ALWAYS_FATAL("invalid src format %#x for dst format %#x",
src_format, dst_format);
}
status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
callback_t cbf,
void* user,
uint32_t notificationFrames,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
audio_input_flags_t flags)
{
ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"notificationFrames %u, sessionId %d, transferType %d, flags %#x",
inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
sessionId, transferType, flags);
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mTransfer = transferType;
AutoMutex lock(mLock);
// invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
// handle default values first.
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
mInputSource = inputSource;
if (sampleRate == 0) {
ALOGE("Invalid sample rate %u", sampleRate);
return BAD_VALUE;
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %#x", format);
return BAD_VALUE;
}
// Temporary restriction: AudioFlinger currently supports 16-bit PCM only
if (format != AUDIO_FORMAT_PCM_16_BIT) {
ALOGE("Format %#x is not supported", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
mChannelCount = channelCount;
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
// mFrameCount is initialized in openRecord_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
// mNotificationFramesAct is initialized in openRecord_l
if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = AudioSystem::newAudioUniqueId();
} else {
mSessionId = sessionId;
}
ALOGV("set(): mSessionId %d", mSessionId);
mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
}
// create the IAudioRecord
status_t status = openRecord_l(0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioRecordThread != 0) {
mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mActive = false;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId, -1);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
return NO_ERROR;
}
/* TODO mutex stuff here (see out_write) */
static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
{
size_t num_read_buff_bytes = 0;
void * read_buff = buffer;
void * out_buff = buffer;
int ret = 0;
struct stream_in * in = (struct stream_in *)stream;
lock_input_stream(in);
if (in->standby) {
pthread_mutex_lock(&in->dev->lock);
ret = start_input_stream(in);
pthread_mutex_unlock(&in->dev->lock);
if (ret != 0) {
goto err;
}
in->standby = false;
}
alsa_device_profile * profile = in->profile;
/*
* OK, we need to figure out how much data to read to be able to output the requested
* number of bytes in the HAL format (16-bit, stereo).
*/
num_read_buff_bytes = bytes;
int num_device_channels = proxy_get_channel_count(&in->proxy); /* what we told Alsa */
int num_req_channels = in->hal_channel_count; /* what we told AudioFlinger */
if (num_device_channels != num_req_channels) {
num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels;
}
/* Setup/Realloc the conversion buffer (if necessary). */
if (num_read_buff_bytes != bytes) {
if (num_read_buff_bytes > in->conversion_buffer_size) {
/*TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
(and do these conversions themselves) */
in->conversion_buffer_size = num_read_buff_bytes;
in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
}
read_buff = in->conversion_buffer;
}
ret = proxy_read(&in->proxy, read_buff, num_read_buff_bytes);
if (ret == 0) {
if (num_device_channels != num_req_channels) {
// ALOGV("chans dev:%d req:%d", num_device_channels, num_req_channels);
out_buff = buffer;
/* Num Channels conversion */
if (num_device_channels != num_req_channels) {
audio_format_t audio_format = in_get_format(&(in->stream.common));
unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
num_read_buff_bytes =
adjust_channels(read_buff, num_device_channels,
out_buff, num_req_channels,
sample_size_in_bytes, num_read_buff_bytes);
}
}
/* no need to acquire in->dev->lock to read mic_muted here as we don't change its state */
if (num_read_buff_bytes > 0 && in->dev->mic_muted)
memset(buffer, 0, num_read_buff_bytes);
} else {
num_read_buff_bytes = 0; // reset the value after headset is unplugged
}
err:
pthread_mutex_unlock(&in->lock);
return num_read_buff_bytes;
}
status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCountInt,
callback_t cbf,
void* user,
int notificationFrames,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
audio_input_flags_t flags)
{
ALOGV("sampleRate %u, channelMask %#x, format %d", sampleRate, channelMask, format);
ALOGV("inputSource %d", inputSource);
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mTransfer = transferType;
// FIXME "int" here is legacy and will be replaced by size_t later
if (frameCountInt < 0) {
ALOGE("Invalid frame count %d", frameCountInt);
return BAD_VALUE;
}
size_t frameCount = frameCountInt;
ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
frameCount);
AutoMutex lock(mLock);
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
mInputSource = inputSource;
if (sampleRate == 0) {
ALOGE("Invalid sample rate %u", sampleRate);
return BAD_VALUE;
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %d", format);
return BAD_VALUE;
}
#if defined(QCOM_HARDWARE) && !defined(QCOM_DIRECTTRACK)
if (format != AUDIO_FORMAT_PCM_16_BIT &&
!audio_is_compress_voip_format(format) &&
!audio_is_compress_capture_format(format)) {
#else
#ifndef QCOM_DIRECTTRACK
// Temporary restriction: AudioFlinger currently supports 16-bit PCM only
if (format != AUDIO_FORMAT_PCM_16_BIT) {
#endif
#endif
#ifndef QCOM_DIRECTTRACK
ALOGE("Format %d is not supported", format);
return BAD_VALUE;
}
#endif
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = popcount(channelMask);
mChannelCount = channelCount;
#ifdef QCOM_DIRECTTRACK
mFrameSize = frameSize();
size_t inputBuffSizeInBytes = -1;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &inputBuffSizeInBytes);
if (status != NO_ERROR) {
ALOGE("AudioSystem could not query the input buffer size; status %d", status);
return NO_INIT;
}
if (inputBuffSizeInBytes == 0) {
ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
int minFrameCount = (inputBuffSizeInBytes * 2)/mFrameSize;
#else
// Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
#ifdef QCOM_HARDWARE
if (audio_is_linear_pcm(format))
mFrameSize = channelCount * audio_bytes_per_sample(format);
else
mFrameSize = sizeof(uint8_t);
#else
mFrameSize = channelCount * audio_bytes_per_sample(format);
#endif
// validate framecount
size_t minFrameCount = 0;
status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
sampleRate, format, channelMask);
if (status != NO_ERROR) {
ALOGE("getMinFrameCount() failed; status %d", status);
return status;
}
#endif
ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
return BAD_VALUE;
}
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
if (sessionId == 0 ) {
mSessionId = AudioSystem::newAudioSessionId();
} else {
mSessionId = sessionId;
}
ALOGV("set(): mSessionId %d", mSessionId);
mFlags = flags;
// create the IAudioRecord
status = openRecord_l(0 /*epoch*/);
if (status) {
return status;
}
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
}
mStatus = NO_ERROR;
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount_;
mActive = false;
mCbf = cbf;
mRefreshRemaining = true;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
return NO_ERROR;
}
#ifdef QCOM_DIRECTTRACK
audio_source_t AudioRecord::inputSource() const
{
return mInputSource;
}
#endif
// -------------------------------------------------------------------------
status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
{
ALOGV("start, sync event %d trigger session %d", event, triggerSession);
AutoMutex lock(mLock);
if (mActive) {
return NO_ERROR;
}
// reset current position as seen by client to 0
mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
mNewPosition = mProxy->getPosition() + mUpdatePeriod;
int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
status_t status = NO_ERROR;
if (!(flags & CBLK_INVALID)) {
ALOGV("mAudioRecord->start()");
status = mAudioRecord->start(event, triggerSession);
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
}
}
if (flags & CBLK_INVALID) {
status = restoreRecord_l("start");
}
if (status != NO_ERROR) {
ALOGE("start() status %d", status);
} else {
mActive = true;
sp<AudioRecordThread> t = mAudioRecordThread;
if (t != 0) {
t->resume();
} else {
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
get_sched_policy(0, &mPreviousSchedulingGroup);
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
}
return status;
}
status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
callback_t cbf,
void* user,
uint32_t notificationFrames,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
audio_input_flags_t flags,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes)
{
ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"notificationFrames %u, sessionId %d, transferType %d, flags %#x, opPackageName %s "
"uid %d, pid %d",
inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
sessionId, transferType, flags, String8(mOpPackageName).string(), uid, pid);
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mTransfer = transferType;
// invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
if (pAttributes == NULL) {
memset(&mAttributes, 0, sizeof(audio_attributes_t));
mAttributes.source = inputSource;
} else {
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
ALOGV("Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",
mAttributes.source, mAttributes.flags, mAttributes.tags);
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
// AudioFlinger capture only supports linear PCM
if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
ALOGE("Format %#x is not linear pcm", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
mChannelCount = channelCount;
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
// mFrameCount is initialized in openRecord_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
// mNotificationFramesAct is initialized in openRecord_l
if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
} else {
mSessionId = sessionId;
}
ALOGV("set(): mSessionId %d", mSessionId);
int callingpid = IPCThreadState::self()->getCallingPid();
int mypid = getpid();
if (uid == -1 || (callingpid != mypid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingpid != mypid)) {
mClientPid = callingpid;
} else {
mClientPid = pid;
}
mOrigFlags = mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioRecord
status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);
if (status != NO_ERROR) {
if (mAudioRecordThread != 0) {
mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000 * mFrameCount) / mSampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId, -1);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
mFramesRead = 0;
mFramesReadServerOffset = 0;
return NO_ERROR;
}
DownmixerBufferProvider::DownmixerBufferProvider(
audio_channel_mask_t inputChannelMask,
audio_channel_mask_t outputChannelMask, audio_format_t format,
uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) :
CopyBufferProvider(
audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
bufferFrameCount) // set bufferFrameCount to 0 to do in-place
{
ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d)",
this, inputChannelMask, outputChannelMask, format,
sampleRate, sessionId);
if (!sIsMultichannelCapable
|| EffectCreate(&sDwnmFxDesc.uuid,
sessionId,
SESSION_ID_INVALID_AND_IGNORED,
&mDownmixHandle) != 0) {
ALOGE("DownmixerBufferProvider() error creating downmixer effect");
mDownmixHandle = NULL;
return;
}
// channel input configuration will be overridden per-track
mDownmixConfig.inputCfg.channels = inputChannelMask; // FIXME: Should be bits
mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits
mDownmixConfig.inputCfg.format = format;
mDownmixConfig.outputCfg.format = format;
mDownmixConfig.inputCfg.samplingRate = sampleRate;
mDownmixConfig.outputCfg.samplingRate = sampleRate;
mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
// input and output buffer provider, and frame count will not be used as the downmix effect
// process() function is called directly (see DownmixerBufferProvider::getNextBuffer())
mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask;
int cmdStatus;
uint32_t replySize = sizeof(int);
// Configure downmixer
status_t status = (*mDownmixHandle)->command(mDownmixHandle,
EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
&mDownmixConfig /*pCmdData*/,
&replySize, &cmdStatus /*pReplyData*/);
if (status != 0 || cmdStatus != 0) {
ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer",
status, cmdStatus);
EffectRelease(mDownmixHandle);
mDownmixHandle = NULL;
return;
}
// Enable downmixer
replySize = sizeof(int);
status = (*mDownmixHandle)->command(mDownmixHandle,
EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
&replySize, &cmdStatus /*pReplyData*/);
if (status != 0 || cmdStatus != 0) {
ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer",
status, cmdStatus);
EffectRelease(mDownmixHandle);
mDownmixHandle = NULL;
return;
}
// Set downmix type
// parameter size rounded for padding on 32bit boundary
const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int);
const int downmixParamSize =
sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t);
effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize);
CHECK(param != NULL);
param->psize = sizeof(downmix_params_t);
const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE;
memcpy(param->data, &downmixParam, param->psize);
const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD;
param->vsize = sizeof(downmix_type_t);
memcpy(param->data + psizePadded, &downmixType, param->vsize);
replySize = sizeof(int);
status = (*mDownmixHandle)->command(mDownmixHandle,
EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */,
param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/);
free(param);
if (status != 0 || cmdStatus != 0) {
ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type",
status, cmdStatus);
EffectRelease(mDownmixHandle);
mDownmixHandle = NULL;
return;
}
ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType);
}
