diff options
| -rw-r--r-- | include/hardware/audio.h | 10 | ||||
| -rw-r--r-- | include/hardware/audio_effect.h | 26 | ||||
| -rw-r--r-- | modules/Android.mk | 2 | ||||
| -rw-r--r-- | modules/audio/audio_hw.c | 26 | ||||
| -rw-r--r-- | modules/audio_remote_submix/Android.mk | 29 | ||||
| -rw-r--r-- | modules/audio_remote_submix/audio_hw.cpp | 724 | ||||
| -rw-r--r-- | modules/usbaudio/audio_hw.c | 8 |
7 files changed, 791 insertions, 34 deletions
diff --git a/include/hardware/audio.h b/include/hardware/audio.h index 26e9ea9..3a0962e 100644 --- a/include/hardware/audio.h +++ b/include/hardware/audio.h @@ -53,7 +53,8 @@ __BEGIN_DECLS */ #define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0) #define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0) -#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_1_0 +#define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0) +#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_2_0 /** * List of known audio HAL modules. This is the base name of the audio HAL @@ -65,6 +66,7 @@ __BEGIN_DECLS #define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary" #define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp" #define AUDIO_HARDWARE_MODULE_ID_USB "usb" +#define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix" /**************************************/ @@ -328,6 +330,12 @@ struct audio_hw_device { * each audio_hw_device implementation. * * Return value is a bitmask of 1 or more values of audio_devices_t + * + * NOTE: audio HAL implementations starting with + * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function. + * All supported devices should be listed in audio_policy.conf + * file and the audio policy manager must choose the appropriate + * audio module based on information in this file. */ uint32_t (*get_supported_devices)(const struct audio_hw_device *dev); diff --git a/include/hardware/audio_effect.h b/include/hardware/audio_effect.h index 4037bbb..46e323d 100644 --- a/include/hardware/audio_effect.h +++ b/include/hardware/audio_effect.h @@ -142,6 +142,10 @@ typedef struct effect_descriptor_s { // | | | 1 requires audio mode updates // | | | 2..3 reserved // +---------------------------+-----------+----------------------------------- +// | Audio source indication | 20..21 | 0 none +// | | | 1 requires audio source updates +// | | | 2..3 reserved +// +---------------------------+-----------+----------------------------------- // Insert mode #define EFFECT_FLAG_TYPE_SHIFT 0 @@ -216,6 +220,13 @@ typedef struct effect_descriptor_s { #define EFFECT_FLAG_AUDIO_MODE_IND (1 << EFFECT_FLAG_AUDIO_MODE_SHIFT) #define EFFECT_FLAG_AUDIO_MODE_NONE (0 << EFFECT_FLAG_AUDIO_MODE_SHIFT) +// Audio source indication +#define EFFECT_FLAG_AUDIO_SOURCE_SHIFT (EFFECT_FLAG_AUDIO_MODE_SHIFT + EFFECT_FLAG_AUDIO_MODE_SIZE) +#define EFFECT_FLAG_AUDIO_SOURCE_SIZE 2 +#define EFFECT_FLAG_AUDIO_SOURCE_MASK (((1 << EFFECT_FLAG_AUDIO_SOURCE_SIZE) -1) \ + << EFFECT_FLAG_AUDIO_SOURCE_SHIFT) +#define EFFECT_FLAG_AUDIO_SOURCE_IND (1 << EFFECT_FLAG_AUDIO_SOURCE_SHIFT) +#define EFFECT_FLAG_AUDIO_SOURCE_NONE (0 << EFFECT_FLAG_AUDIO_SOURCE_SHIFT) #define EFFECT_MAKE_API_VERSION(M, m) (((M)<<16) | ((m) & 0xFFFF)) #define EFFECT_API_VERSION_MAJOR(v) ((v)>>16) @@ -413,6 +424,7 @@ enum effect_command_e { EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS,// get all supported configurations for a feature. EFFECT_CMD_GET_FEATURE_CONFIG, // get current feature configuration EFFECT_CMD_SET_FEATURE_CONFIG, // set current feature configuration + EFFECT_CMD_SET_AUDIO_SOURCE, // set the audio source (see audio.h, audio_source_t) EFFECT_CMD_FIRST_PROPRIETARY = 0x10000 // first proprietary command code }; @@ -705,6 +717,20 @@ enum effect_command_e { // size: sizeof(uint32_t) // data: status //================================================================================================== +// command: EFFECT_CMD_SET_AUDIO_SOURCE +//-------------------------------------------------------------------------------------------------- +// description: +// Set the audio source the capture path is configured for (Camcorder, voice recognition...). +// See audio.h, audio_source_t for values. +//-------------------------------------------------------------------------------------------------- +// command format: +// size: sizeof(uint32_t) +// data: uint32_t +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: 0 +// data: N/A +//================================================================================================== // command: EFFECT_CMD_FIRST_PROPRIETARY //-------------------------------------------------------------------------------------------------- // description: diff --git a/modules/Android.mk b/modules/Android.mk index 871b984..3f410c1 100644 --- a/modules/Android.mk +++ b/modules/Android.mk @@ -1,2 +1,2 @@ -hardware_modules := gralloc hwcomposer audio nfc local_time power usbaudio +hardware_modules := gralloc hwcomposer audio nfc local_time power usbaudio audio_remote_submix include $(call all-named-subdir-makefiles,$(hardware_modules)) diff --git a/modules/audio/audio_hw.c b/modules/audio/audio_hw.c index e4fb711..3051519 100644 --- a/modules/audio/audio_hw.c +++ b/modules/audio/audio_hw.c @@ -379,29 +379,6 @@ static int adev_close(hw_device_t *device) return 0; } -static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev) -{ - return (/* OUT */ - AUDIO_DEVICE_OUT_EARPIECE | - AUDIO_DEVICE_OUT_SPEAKER | - AUDIO_DEVICE_OUT_WIRED_HEADSET | - AUDIO_DEVICE_OUT_WIRED_HEADPHONE | - AUDIO_DEVICE_OUT_AUX_DIGITAL | - AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET | - AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET | - AUDIO_DEVICE_OUT_ALL_SCO | - AUDIO_DEVICE_OUT_DEFAULT | - /* IN */ - AUDIO_DEVICE_IN_COMMUNICATION | - AUDIO_DEVICE_IN_AMBIENT | - AUDIO_DEVICE_IN_BUILTIN_MIC | - AUDIO_DEVICE_IN_WIRED_HEADSET | - AUDIO_DEVICE_IN_AUX_DIGITAL | - AUDIO_DEVICE_IN_BACK_MIC | - AUDIO_DEVICE_IN_ALL_SCO | - AUDIO_DEVICE_IN_DEFAULT); -} - static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device) { @@ -416,11 +393,10 @@ static int adev_open(const hw_module_t* module, const char* name, return -ENOMEM; adev->device.common.tag = HARDWARE_DEVICE_TAG; - adev->device.common.version = AUDIO_DEVICE_API_VERSION_1_0; + 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.get_supported_devices = adev_get_supported_devices; 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; diff --git a/modules/audio_remote_submix/Android.mk b/modules/audio_remote_submix/Android.mk new file mode 100644 index 0000000..735215e --- /dev/null +++ b/modules/audio_remote_submix/Android.mk @@ -0,0 +1,29 @@ +# 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. + +LOCAL_PATH := $(call my-dir) + +include $(CLEAR_VARS) + +LOCAL_MODULE := audio.r_submix.default +LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw +LOCAL_SRC_FILES := \ + audio_hw.cpp +LOCAL_C_INCLUDES += \ + frameworks/av/include/ \ + frameworks/native/include/ +LOCAL_SHARED_LIBRARIES := liblog libcutils libutils libnbaio +LOCAL_MODULE_TAGS := optional +include $(BUILD_SHARED_LIBRARY) + diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp new file mode 100644 index 0000000..2468309 --- /dev/null +++ b/modules/audio_remote_submix/audio_hw.cpp @@ -0,0 +1,724 @@ +/* + * 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/Pipe.h> +#include <media/nbaio/PipeReader.h> +#include <media/AudioBufferProvider.h> + +extern "C" { + +namespace android { + +#define MAX_PIPE_DEPTH_IN_FRAMES (1024*4) +#define MAX_READ_ATTEMPTS 10 +#define READ_ATTEMPT_SLEEP_MS 10 // 10ms 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; + submix_config config; + // Pipe variables: they handle the ring buffer that "pipes" audio: + // - from the submix virtual audio output == what needs to be played by + // the 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. + // An 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<Pipe> rsxSink; + sp<PipeReader> rsxSource; + + 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; +}; + + +/* 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) +{ + // REMOTE_SUBMIX is a proxy / virtual audio device, so the notion of standby doesn't apply here + 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) +{ + 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 = 0; + struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream); + + pthread_mutex_lock(&out->dev->lock); + + Pipe* sink = out->dev->rsxSink.get(); + if (sink != NULL) { + out->dev->rsxSink->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); + + const size_t frames = bytes / audio_stream_frame_size(&stream->common); + written = sink->write(buffer, frames); + if (written < 0) { + if (written == (ssize_t)NEGOTIATE) { + ALOGE("out_write() write to pipe returned NEGOTIATE"); + written = 0; + } else { + // write() returned UNDERRUN or WOULD_BLOCK, retry + written = sink->write(buffer, frames); + } + } + + pthread_mutex_lock(&out->dev->lock); + + out->dev->rsxSink->decStrong(buffer); + + pthread_mutex_unlock(&out->dev->lock); + + if (written > 0) { + // fake timing for audio output, we can't return right after pushing the data in the pipe + // TODO who's doing the flow control here? the wifi display link, or the audio HAL? + usleep(written * 1000000 / out_get_sample_rate(&stream->common)); + return written * audio_stream_frame_size(&stream->common);; + } else { + // error occurred, fake timing + usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); + ALOGE("out_write error=%16lx", written); + return 0; + } +} + +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) +{ + // REMOTE_SUBMIX is a proxy / virtual audio device, so the notion of standby doesn't apply here + 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) +{ + ssize_t frames_read = -1977; + const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); + const size_t frame_size = audio_stream_frame_size(&stream->common); + + pthread_mutex_lock(&in->dev->lock); + + PipeReader* source = in->dev->rsxSource.get(); + if (source != NULL) { + in->dev->rsxSource->incStrong(in); + } else { + 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); + + int attempts = MAX_READ_ATTEMPTS; + size_t remaining_frames = bytes / frame_size; + char* buff = (char*)buffer; + while (attempts > 0) { + frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS); + if (frames_read > 0) { + //ALOGV("in_read frames=%ld size=%u", remaining_frames, frame_size); + remaining_frames -= frames_read; + buff += frames_read * frame_size; + if (remaining_frames == 0) { + // TODO simplify code by breaking out of loop + + pthread_mutex_lock(&in->dev->lock); + + in->dev->rsxSource->decStrong(in); + + pthread_mutex_unlock(&in->dev->lock); + + return bytes; + } + } else if (frames_read == 0) { + // TODO sleep should be tied to how much data is expected + usleep(READ_ATTEMPT_SLEEP_MS*1000); + attempts--; + } else { // frames_read is an error code + if (frames_read != (ssize_t)OVERRUN) { + attempts--; + } + // else OVERRUN: error has been signaled, ok to read, do not decrement counter + } + } + + pthread_mutex_lock(&in->dev->lock); + + in->dev->rsxSource->decStrong(in); + + pthread_mutex_unlock(&in->dev->lock); + + // TODO how to handle partial reads? + + if (frames_read < 0) { + ALOGE("in_read error=%16lx", frames_read); + } + return 0; +} + +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 Pipe, not a MonoPipe with optional blocking set to true, so audio frames + // entering a full sink will overwrite the contents of the pipe. + Pipe* sink = new Pipe(MAX_PIPE_DEPTH_IN_FRAMES, format); + ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + PipeReader* source = new PipeReader(*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; + + 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); + + 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 uint32_t adev_get_supported_devices(const struct audio_hw_device *dev) +{ + ALOGI("adev_get_supported_devices() returns %08x", + AUDIO_DEVICE_OUT_REMOTE_SUBMIX |AUDIO_DEVICE_IN_REMOTE_SUBMIX); + return (/* OUT */ + AUDIO_DEVICE_OUT_REMOTE_SUBMIX | + /* IN */ + AUDIO_DEVICE_IN_REMOTE_SUBMIX); +} + +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_1_0; + rsxadev->device.common.module = (struct hw_module_t *) module; + rsxadev->device.common.close = adev_close; + + rsxadev->device.get_supported_devices = adev_get_supported_devices; + 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; + + *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" diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c index 9283016..f33c343 100644 --- a/modules/usbaudio/audio_hw.c +++ b/modules/usbaudio/audio_hw.c @@ -379,11 +379,6 @@ static int adev_close(hw_device_t *device) return 0; } -static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev) -{ - return AUDIO_DEVICE_OUT_ALL_USB; -} - static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device) { @@ -398,11 +393,10 @@ static int adev_open(const hw_module_t* module, const char* name, return -ENOMEM; adev->hw_device.common.tag = HARDWARE_DEVICE_TAG; - adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_1_0; + adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0; adev->hw_device.common.module = (struct hw_module_t *) module; adev->hw_device.common.close = adev_close; - adev->hw_device.get_supported_devices = adev_get_supported_devices; adev->hw_device.init_check = adev_init_check; adev->hw_device.set_voice_volume = adev_set_voice_volume; adev->hw_device.set_master_volume = adev_set_master_volume; |