diff options
Diffstat (limited to 'modules/audio_remote_submix/audio_hw.cpp')
| -rwxr-xr-x | modules/audio_remote_submix/audio_hw.cpp | 839 |
1 files changed, 839 insertions, 0 deletions
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp new file mode 100755 index 0000000..3756274 --- /dev/null +++ b/modules/audio_remote_submix/audio_hw.cpp @@ -0,0 +1,839 @@ +/* + * Copyright (C) 2012 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * 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 "r_submix" +//#define LOG_NDEBUG 0 + +#include <errno.h> +#include <pthread.h> +#include <stdint.h> +#include <sys/time.h> +#include <stdlib.h> + +#include <cutils/log.h> +#include <cutils/str_parms.h> +#include <cutils/properties.h> + +#include <hardware/hardware.h> +#include <system/audio.h> +#include <hardware/audio.h> + +#include <media/nbaio/MonoPipe.h> +#include <media/nbaio/MonoPipeReader.h> +#include <media/AudioBufferProvider.h> + +#include <utils/String8.h> +#include <media/AudioParameter.h> + +extern "C" { + +namespace android { + +#define MAX_PIPE_DEPTH_IN_FRAMES (1024*8) +// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to +// the duration of a record buffer at the current record sample rate (of the device, not of +// the recording itself). Here we have: +// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms +#define MAX_READ_ATTEMPTS 3 +#define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty +#define DEFAULT_RATE_HZ 48000 // default sample rate + +struct submix_config { + audio_format_t format; + audio_channel_mask_t channel_mask; + unsigned int rate; // sample rate for the device + unsigned int period_size; // size of the audio pipe is period_size * period_count in frames + unsigned int period_count; +}; + +struct submix_audio_device { + struct audio_hw_device device; + bool output_standby; + bool input_standby; + submix_config config; + // Pipe variables: they handle the ring buffer that "pipes" audio: + // - from the submix virtual audio output == what needs to be played + // remotely, seen as an output for AudioFlinger + // - to the virtual audio source == what is captured by the component + // which "records" the submix / virtual audio source, and handles it as needed. + // A usecase example is one where the component capturing the audio is then sending it over + // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a + // TV with Wifi Display capabilities), or to a wireless audio player. + sp<MonoPipe> rsxSink; + sp<MonoPipeReader> rsxSource; + + // device lock, also used to protect access to the audio pipe + pthread_mutex_t lock; +}; + +struct submix_stream_out { + struct audio_stream_out stream; + struct submix_audio_device *dev; +}; + +struct submix_stream_in { + struct audio_stream_in stream; + struct submix_audio_device *dev; + bool output_standby; // output standby state as seen from record thread + + // wall clock when recording starts + struct timespec record_start_time; + // how many frames have been requested to be read + int64_t read_counter_frames; +}; + + +/* audio HAL functions */ + +static uint32_t out_get_sample_rate(const struct audio_stream *stream) +{ + const struct submix_stream_out *out = + reinterpret_cast<const struct submix_stream_out *>(stream); + uint32_t out_rate = out->dev->config.rate; + //ALOGV("out_get_sample_rate() returns %u", out_rate); + return out_rate; +} + +static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) +{ + if ((rate != 44100) && (rate != 48000)) { + ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); + return -ENOSYS; + } + struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream); + //ALOGV("out_set_sample_rate(rate=%u)", rate); + out->dev->config.rate = rate; + return 0; +} + +static size_t out_get_buffer_size(const struct audio_stream *stream) +{ + const struct submix_stream_out *out = + reinterpret_cast<const struct submix_stream_out *>(stream); + const struct submix_config& config_out = out->dev->config; + size_t buffer_size = config_out.period_size * popcount(config_out.channel_mask) + * sizeof(int16_t); // only PCM 16bit + //ALOGV("out_get_buffer_size() returns %u, period size=%u", + // buffer_size, config_out.period_size); + return buffer_size; +} + +static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) +{ + const struct submix_stream_out *out = + reinterpret_cast<const struct submix_stream_out *>(stream); + uint32_t channels = out->dev->config.channel_mask; + //ALOGV("out_get_channels() returns %08x", channels); + return channels; +} + +static audio_format_t out_get_format(const struct audio_stream *stream) +{ + return AUDIO_FORMAT_PCM_16_BIT; +} + +static int out_set_format(struct audio_stream *stream, audio_format_t format) +{ + if (format != AUDIO_FORMAT_PCM_16_BIT) { + return -ENOSYS; + } else { + return 0; + } +} + +static int out_standby(struct audio_stream *stream) +{ + ALOGI("out_standby()"); + + const struct submix_stream_out *out = reinterpret_cast<const struct submix_stream_out *>(stream); + + pthread_mutex_lock(&out->dev->lock); + + out->dev->output_standby = true; + + pthread_mutex_unlock(&out->dev->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) +{ + int exiting = -1; + AudioParameter parms = AudioParameter(String8(kvpairs)); + // FIXME this is using hard-coded strings but in the future, this functionality will be + // converted to use audio HAL extensions required to support tunneling + if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { + const struct submix_stream_out *out = + reinterpret_cast<const struct submix_stream_out *>(stream); + + pthread_mutex_lock(&out->dev->lock); + + MonoPipe* sink = out->dev->rsxSink.get(); + if (sink != NULL) { + sink->incStrong(out); + } else { + pthread_mutex_unlock(&out->dev->lock); + return 0; + } + + ALOGI("shutdown"); + sink->shutdown(true); + + sink->decStrong(out); + + pthread_mutex_unlock(&out->dev->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 submix_stream_out *out = + reinterpret_cast<const struct submix_stream_out *>(stream); + const struct submix_config * config_out = &(out->dev->config); + uint32_t latency = (MAX_PIPE_DEPTH_IN_FRAMES * 1000) / config_out->rate; + ALOGV("out_get_latency() returns %u", 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) +{ + //ALOGV("out_write(bytes=%d)", bytes); + ssize_t written_frames = 0; + struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream); + + const size_t frame_size = audio_stream_frame_size(&stream->common); + const size_t frames = bytes / frame_size; + + pthread_mutex_lock(&out->dev->lock); + + out->dev->output_standby = false; + + MonoPipe* sink = out->dev->rsxSink.get(); + if (sink != NULL) { + if (sink->isShutdown()) { + pthread_mutex_unlock(&out->dev->lock); + // the pipe has already been shutdown, this buffer will be lost but we must + // simulate timing so we don't drain the output faster than realtime + usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); + return bytes; + } + sink->incStrong(buffer); + } else { + pthread_mutex_unlock(&out->dev->lock); + ALOGE("out_write without a pipe!"); + ALOG_ASSERT("out_write without a pipe!"); + return 0; + } + + pthread_mutex_unlock(&out->dev->lock); + + written_frames = sink->write(buffer, frames); + if (written_frames < 0) { + if (written_frames == (ssize_t)NEGOTIATE) { + ALOGE("out_write() write to pipe returned NEGOTIATE"); + + pthread_mutex_lock(&out->dev->lock); + sink->decStrong(buffer); + pthread_mutex_unlock(&out->dev->lock); + + written_frames = 0; + return 0; + } else { + // write() returned UNDERRUN or WOULD_BLOCK, retry + ALOGE("out_write() write to pipe returned unexpected %16lx", written_frames); + written_frames = sink->write(buffer, frames); + } + } + + pthread_mutex_lock(&out->dev->lock); + + sink->decStrong(buffer); + + pthread_mutex_unlock(&out->dev->lock); + + if (written_frames < 0) { + ALOGE("out_write() failed writing to pipe with %16lx", written_frames); + return 0; + } else { + ALOGV("out_write() wrote %lu bytes)", written_frames * frame_size); + return written_frames * frame_size; + } +} + +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; +} + +/** audio_stream_in implementation **/ +static uint32_t in_get_sample_rate(const struct audio_stream *stream) +{ + const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); + //ALOGV("in_get_sample_rate() returns %u", in->dev->config.rate); + return in->dev->config.rate; +} + +static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) +{ + return -ENOSYS; +} + +static size_t in_get_buffer_size(const struct audio_stream *stream) +{ + const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); + ALOGV("in_get_buffer_size() returns %u", + in->dev->config.period_size * audio_stream_frame_size(stream)); + return in->dev->config.period_size * audio_stream_frame_size(stream); +} + +static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) +{ + return AUDIO_CHANNEL_IN_STEREO; +} + +static audio_format_t in_get_format(const struct audio_stream *stream) +{ + return AUDIO_FORMAT_PCM_16_BIT; +} + +static int in_set_format(struct audio_stream *stream, audio_format_t format) +{ + if (format != AUDIO_FORMAT_PCM_16_BIT) { + return -ENOSYS; + } else { + return 0; + } +} + +static int in_standby(struct audio_stream *stream) +{ + ALOGI("in_standby()"); + const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); + + pthread_mutex_lock(&in->dev->lock); + + in->dev->input_standby = true; + + pthread_mutex_unlock(&in->dev->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) +{ + 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 ssize_t in_read(struct audio_stream_in *stream, void* buffer, + size_t bytes) +{ + //ALOGV("in_read bytes=%u", bytes); + ssize_t frames_read = -1977; + struct submix_stream_in *in = reinterpret_cast<struct submix_stream_in *>(stream); + const size_t frame_size = audio_stream_frame_size(&stream->common); + const size_t frames_to_read = bytes / frame_size; + + pthread_mutex_lock(&in->dev->lock); + + const bool output_standby_transition = (in->output_standby != in->dev->output_standby); + in->output_standby = in->dev->output_standby; + + if (in->dev->input_standby || output_standby_transition) { + in->dev->input_standby = false; + // keep track of when we exit input standby (== first read == start "real recording") + // or when we start recording silence, and reset projected time + int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); + if (rc == 0) { + in->read_counter_frames = 0; + } + } + + in->read_counter_frames += frames_to_read; + + MonoPipeReader* source = in->dev->rsxSource.get(); + if (source != NULL) { + source->incStrong(buffer); + } else { + ALOGE("no audio pipe yet we're trying to read!"); + pthread_mutex_unlock(&in->dev->lock); + usleep((bytes / frame_size) * 1000000 / in_get_sample_rate(&stream->common)); + memset(buffer, 0, bytes); + return bytes; + } + + pthread_mutex_unlock(&in->dev->lock); + + // read the data from the pipe (it's non blocking) + size_t remaining_frames = frames_to_read; + int attempts = 0; + char* buff = (char*)buffer; + while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { + attempts++; + frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS); + if (frames_read > 0) { + remaining_frames -= frames_read; + buff += frames_read * frame_size; + //ALOGV(" in_read (att=%d) got %ld frames, remaining=%u", + // attempts, frames_read, remaining_frames); + } else { + //ALOGE(" in_read read returned %ld", frames_read); + usleep(READ_ATTEMPT_SLEEP_MS * 1000); + } + } + + // done using the source + pthread_mutex_lock(&in->dev->lock); + + source->decStrong(buffer); + + pthread_mutex_unlock(&in->dev->lock); + + if (remaining_frames > 0) { + ALOGV(" remaining_frames = %d", remaining_frames); + memset(((char*)buffer)+ bytes - (remaining_frames * frame_size), 0, + remaining_frames * frame_size); + } + + // compute how much we need to sleep after reading the data by comparing the wall clock with + // the projected time at which we should return. + struct timespec time_after_read;// wall clock after reading from the pipe + struct timespec record_duration;// observed record duration + int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); + const uint32_t sample_rate = in_get_sample_rate(&stream->common); + if (rc == 0) { + // for how long have we been recording? + record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; + record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; + if (record_duration.tv_nsec < 0) { + record_duration.tv_sec--; + record_duration.tv_nsec += 1000000000; + } + + // read_counter_frames contains the number of frames that have been read since the beginning + // of recording (including this call): it's converted to usec and compared to how long we've + // been recording for, which gives us how long we must wait to sync the projected recording + // time, and the observed recording time + long projected_vs_observed_offset_us = + ((int64_t)(in->read_counter_frames + - (record_duration.tv_sec*sample_rate))) + * 1000000 / sample_rate + - (record_duration.tv_nsec / 1000); + + ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", + record_duration.tv_sec, record_duration.tv_nsec/1000000, + projected_vs_observed_offset_us); + if (projected_vs_observed_offset_us > 0) { + usleep(projected_vs_observed_offset_us); + } + } + + + ALOGV("in_read returns %d", bytes); + return bytes; + +} + +static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) +{ + 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) +{ + ALOGV("adev_open_output_stream()"); + struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; + struct submix_stream_out *out; + int ret; + + out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); + if (!out) { + ret = -ENOMEM; + goto err_open; + } + + pthread_mutex_lock(&rsxadev->lock); + + 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; + + config->channel_mask = AUDIO_CHANNEL_OUT_STEREO; + rsxadev->config.channel_mask = config->channel_mask; + + if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) { + config->sample_rate = DEFAULT_RATE_HZ; + } + rsxadev->config.rate = config->sample_rate; + + config->format = AUDIO_FORMAT_PCM_16_BIT; + rsxadev->config.format = config->format; + + rsxadev->config.period_size = 1024; + rsxadev->config.period_count = 4; + out->dev = rsxadev; + + *stream_out = &out->stream; + + // initialize pipe + { + ALOGV(" initializing pipe"); + const NBAIO_Format format = + config->sample_rate == 48000 ? Format_SR48_C2_I16 : Format_SR44_1_C2_I16; + const NBAIO_Format offers[1] = {format}; + size_t numCounterOffers = 0; + // creating a MonoPipe with optional blocking set to true. + MonoPipe* sink = new MonoPipe(MAX_PIPE_DEPTH_IN_FRAMES, format, true/*writeCanBlock*/); + ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + MonoPipeReader* source = new MonoPipeReader(sink); + numCounterOffers = 0; + index = source->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + rsxadev->rsxSink = sink; + rsxadev->rsxSource = source; + } + + pthread_mutex_unlock(&rsxadev->lock); + + return 0; + +err_open: + *stream_out = NULL; + return ret; +} + +static void adev_close_output_stream(struct audio_hw_device *dev, + struct audio_stream_out *stream) +{ + ALOGV("adev_close_output_stream()"); + struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; + + pthread_mutex_lock(&rsxadev->lock); + + rsxadev->rsxSink.clear(); + rsxadev->rsxSource.clear(); + free(stream); + + pthread_mutex_unlock(&rsxadev->lock); +} + +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) +{ + ALOGI("adev_init_check()"); + 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) +{ + return 0; +} + +static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) +{ + return -ENOSYS; +} + +static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) +{ + return -ENOSYS; +} + +static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, + const struct audio_config *config) +{ + //### TODO correlate this with pipe parameters + return 4096; +} + +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) +{ + ALOGI("adev_open_input_stream()"); + + struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; + struct submix_stream_in *in; + int ret; + + in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); + if (!in) { + ret = -ENOMEM; + goto err_open; + } + + pthread_mutex_lock(&rsxadev->lock); + + 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; + + config->channel_mask = AUDIO_CHANNEL_IN_STEREO; + rsxadev->config.channel_mask = config->channel_mask; + + if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) { + config->sample_rate = DEFAULT_RATE_HZ; + } + rsxadev->config.rate = config->sample_rate; + + config->format = AUDIO_FORMAT_PCM_16_BIT; + rsxadev->config.format = config->format; + + rsxadev->config.period_size = 1024; + rsxadev->config.period_count = 4; + + *stream_in = &in->stream; + + in->dev = rsxadev; + + in->read_counter_frames = 0; + in->output_standby = rsxadev->output_standby; + + pthread_mutex_unlock(&rsxadev->lock); + + return 0; + +err_open: + *stream_in = NULL; + return ret; +} + +static void adev_close_input_stream(struct audio_hw_device *dev, + struct audio_stream_in *stream) +{ + ALOGV("adev_close_input_stream()"); + struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; + + pthread_mutex_lock(&rsxadev->lock); + + MonoPipe* sink = rsxadev->rsxSink.get(); + if (sink != NULL) { + ALOGI("shutdown"); + sink->shutdown(true); + } + + free(stream); + + pthread_mutex_unlock(&rsxadev->lock); +} + +static int adev_dump(const audio_hw_device_t *device, int fd) +{ + return 0; +} + +static int adev_close(hw_device_t *device) +{ + ALOGI("adev_close()"); + free(device); + return 0; +} + +static int adev_open(const hw_module_t* module, const char* name, + hw_device_t** device) +{ + ALOGI("adev_open(name=%s)", name); + struct submix_audio_device *rsxadev; + + if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) + return -EINVAL; + + rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); + if (!rsxadev) + return -ENOMEM; + + rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; + rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; + rsxadev->device.common.module = (struct hw_module_t *) module; + rsxadev->device.common.close = adev_close; + + rsxadev->device.init_check = adev_init_check; + rsxadev->device.set_voice_volume = adev_set_voice_volume; + rsxadev->device.set_master_volume = adev_set_master_volume; + rsxadev->device.get_master_volume = adev_get_master_volume; + rsxadev->device.set_master_mute = adev_set_master_mute; + rsxadev->device.get_master_mute = adev_get_master_mute; + rsxadev->device.set_mode = adev_set_mode; + rsxadev->device.set_mic_mute = adev_set_mic_mute; + rsxadev->device.get_mic_mute = adev_get_mic_mute; + rsxadev->device.set_parameters = adev_set_parameters; + rsxadev->device.get_parameters = adev_get_parameters; + rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; + rsxadev->device.open_output_stream = adev_open_output_stream; + rsxadev->device.close_output_stream = adev_close_output_stream; + rsxadev->device.open_input_stream = adev_open_input_stream; + rsxadev->device.close_input_stream = adev_close_input_stream; + rsxadev->device.dump = adev_dump; + + rsxadev->input_standby = true; + rsxadev->output_standby = true; + + *device = &rsxadev->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 */ "Wifi Display audio HAL", + /* author */ "The Android Open Source Project", + /* methods */ &hal_module_methods, + /* dso */ NULL, + /* reserved */ { 0 }, + }, +}; + +} //namespace android + +} //extern "C" |