/* * Copyright (C) 2011 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 "audio_hw_default" //#define LOG_NDEBUG 0 #include <errno.h> #include <malloc.h> #include <pthread.h> #include <stdint.h> #include <stdlib.h> #include <string.h> #include <time.h> #include <unistd.h> #include <log/log.h> #include <hardware/audio.h> #include <hardware/hardware.h> #include <system/audio.h> #define STUB_DEFAULT_SAMPLE_RATE 48000 #define STUB_DEFAULT_AUDIO_FORMAT AUDIO_FORMAT_PCM_16_BIT #define STUB_INPUT_BUFFER_MILLISECONDS 20 #define STUB_INPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_IN_STEREO #define STUB_OUTPUT_BUFFER_MILLISECONDS 10 #define STUB_OUTPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_OUT_STEREO struct stub_audio_device { struct audio_hw_device device; }; struct stub_stream_out { struct audio_stream_out stream; int64_t last_write_time_us; uint32_t sample_rate; audio_channel_mask_t channel_mask; audio_format_t format; size_t frame_count; }; struct stub_stream_in { struct audio_stream_in stream; int64_t last_read_time_us; uint32_t sample_rate; audio_channel_mask_t channel_mask; audio_format_t format; size_t frame_count; }; static uint32_t out_get_sample_rate(const struct audio_stream *stream) { const struct stub_stream_out *out = (const struct stub_stream_out *)stream; ALOGV("out_get_sample_rate: %u", out->sample_rate); return out->sample_rate; } static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) { struct stub_stream_out *out = (struct stub_stream_out *)stream; ALOGV("out_set_sample_rate: %d", rate); out->sample_rate = rate; return 0; } static size_t out_get_buffer_size(const struct audio_stream *stream) { const struct stub_stream_out *out = (const struct stub_stream_out *)stream; size_t buffer_size = out->frame_count * audio_stream_out_frame_size(&out->stream); ALOGV("out_get_buffer_size: %zu", buffer_size); return buffer_size; } static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) { const struct stub_stream_out *out = (const struct stub_stream_out *)stream; ALOGV("out_get_channels: %x", out->channel_mask); return out->channel_mask; } static audio_format_t out_get_format(const struct audio_stream *stream) { const struct stub_stream_out *out = (const struct stub_stream_out *)stream; ALOGV("out_get_format: %d", out->format); return out->format; } static int out_set_format(struct audio_stream *stream, audio_format_t format) { struct stub_stream_out *out = (struct stub_stream_out *)stream; ALOGV("out_set_format: %d", format); out->format = format; return 0; } static int out_standby(struct audio_stream *stream) { ALOGV("out_standby"); // out->last_write_time_us = 0; unnecessary as a stale write time has same effect return 0; } static int out_dump(const struct audio_stream *stream, int fd) { ALOGV("out_dump"); return 0; } static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) { ALOGV("out_set_parameters"); return 0; } static char * out_get_parameters(const struct audio_stream *stream, const char *keys) { ALOGV("out_get_parameters"); return strdup(""); } static uint32_t out_get_latency(const struct audio_stream_out *stream) { ALOGV("out_get_latency"); return STUB_OUTPUT_BUFFER_MILLISECONDS; } static int out_set_volume(struct audio_stream_out *stream, float left, float right) { ALOGV("out_set_volume: Left:%f Right:%f", left, right); return 0; } static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes) { ALOGV("out_write: bytes: %zu", bytes); /* XXX: fake timing for audio output */ struct stub_stream_out *out = (struct stub_stream_out *)stream; struct timespec t = { .tv_sec = 0, .tv_nsec = 0 }; clock_gettime(CLOCK_MONOTONIC, &t); const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000; const int64_t elapsed_time_since_last_write = now - out->last_write_time_us; int64_t sleep_time = bytes * 1000000LL / audio_stream_out_frame_size(stream) / out_get_sample_rate(&stream->common) - elapsed_time_since_last_write; if (sleep_time > 0) { usleep(sleep_time); } else { // we don't sleep when we exit standby (this is typical for a real alsa buffer). sleep_time = 0; } out->last_write_time_us = now + sleep_time; // last_write_time_us is an approximation of when the (simulated) alsa // buffer is believed completely full. The usleep above waits for more space // in the buffer, but by the end of the sleep the buffer is considered // topped-off. // // On the subsequent out_write(), we measure the elapsed time spent in // the mixer. This is subtracted from the sleep estimate based on frames, // thereby accounting for drain in the alsa buffer during mixing. // This is a crude approximation; we don't handle underruns precisely. return bytes; } static int out_get_render_position(const struct audio_stream_out *stream, uint32_t *dsp_frames) { *dsp_frames = 0; ALOGV("out_get_render_position: dsp_frames: %p", dsp_frames); return -EINVAL; } static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { ALOGV("out_add_audio_effect: %p", effect); return 0; } static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { ALOGV("out_remove_audio_effect: %p", effect); return 0; } static int out_get_next_write_timestamp(const struct audio_stream_out *stream, int64_t *timestamp) { *timestamp = 0; ALOGV("out_get_next_write_timestamp: %ld", (long int)(*timestamp)); return -EINVAL; } /** audio_stream_in implementation **/ static uint32_t in_get_sample_rate(const struct audio_stream *stream) { const struct stub_stream_in *in = (const struct stub_stream_in *)stream; ALOGV("in_get_sample_rate: %u", in->sample_rate); return in->sample_rate; } static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) { struct stub_stream_in *in = (struct stub_stream_in *)stream; ALOGV("in_set_sample_rate: %u", rate); in->sample_rate = rate; return 0; } static size_t in_get_buffer_size(const struct audio_stream *stream) { const struct stub_stream_in *in = (const struct stub_stream_in *)stream; size_t buffer_size = in->frame_count * audio_stream_in_frame_size(&in->stream); ALOGV("in_get_buffer_size: %zu", buffer_size); return buffer_size; } static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) { const struct stub_stream_in *in = (const struct stub_stream_in *)stream; ALOGV("in_get_channels: %x", in->channel_mask); return in->channel_mask; } static audio_format_t in_get_format(const struct audio_stream *stream) { const struct stub_stream_in *in = (const struct stub_stream_in *)stream; ALOGV("in_get_format: %d", in->format); return in->format; } static int in_set_format(struct audio_stream *stream, audio_format_t format) { struct stub_stream_in *in = (struct stub_stream_in *)stream; ALOGV("in_set_format: %d", format); in->format = format; return 0; } static int in_standby(struct audio_stream *stream) { struct stub_stream_in *in = (struct stub_stream_in *)stream; in->last_read_time_us = 0; 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 %zu", bytes); /* XXX: fake timing for audio input */ struct stub_stream_in *in = (struct stub_stream_in *)stream; struct timespec t = { .tv_sec = 0, .tv_nsec = 0 }; clock_gettime(CLOCK_MONOTONIC, &t); const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000; // we do a full sleep when exiting standby. const bool standby = in->last_read_time_us == 0; const int64_t elapsed_time_since_last_read = standby ? 0 : now - in->last_read_time_us; int64_t sleep_time = bytes * 1000000LL / audio_stream_in_frame_size(stream) / in_get_sample_rate(&stream->common) - elapsed_time_since_last_read; if (sleep_time > 0) { usleep(sleep_time); } else { sleep_time = 0; } in->last_read_time_us = now + sleep_time; // last_read_time_us is an approximation of when the (simulated) alsa // buffer is drained by the read, and is empty. // // On the subsequent in_read(), we measure the elapsed time spent in // the recording thread. This is subtracted from the sleep estimate based on frames, // thereby accounting for fill in the alsa buffer during the interim. memset(buffer, 0, 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 size_t samples_per_milliseconds(size_t milliseconds, uint32_t sample_rate, size_t channel_count) { return milliseconds * sample_rate * channel_count / 1000; } 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) { ALOGV("adev_open_output_stream..."); *stream_out = NULL; struct stub_stream_out *out = (struct stub_stream_out *)calloc(1, sizeof(struct stub_stream_out)); if (!out) return -ENOMEM; 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->sample_rate = config->sample_rate; if (out->sample_rate == 0) out->sample_rate = STUB_DEFAULT_SAMPLE_RATE; out->channel_mask = config->channel_mask; if (out->channel_mask == AUDIO_CHANNEL_NONE) out->channel_mask = STUB_OUTPUT_DEFAULT_CHANNEL_MASK; out->format = config->format; if (out->format == AUDIO_FORMAT_DEFAULT) out->format = STUB_DEFAULT_AUDIO_FORMAT; out->frame_count = samples_per_milliseconds( STUB_OUTPUT_BUFFER_MILLISECONDS, out->sample_rate, 1); ALOGV("adev_open_output_stream: sample_rate: %u, channels: %x, format: %d," " frames: %zu", out->sample_rate, out->channel_mask, out->format, out->frame_count); *stream_out = &out->stream; return 0; } static void adev_close_output_stream(struct audio_hw_device *dev, struct audio_stream_out *stream) { ALOGV("adev_close_output_stream..."); free(stream); } static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) { ALOGV("adev_set_parameters"); return -ENOSYS; } static char * adev_get_parameters(const struct audio_hw_device *dev, const char *keys) { ALOGV("adev_get_parameters"); return strdup(""); } static int adev_init_check(const struct audio_hw_device *dev) { ALOGV("adev_init_check"); return 0; } static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) { ALOGV("adev_set_voice_volume: %f", volume); return -ENOSYS; } static int adev_set_master_volume(struct audio_hw_device *dev, float volume) { ALOGV("adev_set_master_volume: %f", volume); return -ENOSYS; } static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) { ALOGV("adev_get_master_volume: %f", *volume); return -ENOSYS; } static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) { ALOGV("adev_set_master_mute: %d", muted); return -ENOSYS; } static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) { ALOGV("adev_get_master_mute: %d", *muted); return -ENOSYS; } static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) { ALOGV("adev_set_mode: %d", mode); return 0; } static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) { ALOGV("adev_set_mic_mute: %d",state); return -ENOSYS; } static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) { ALOGV("adev_get_mic_mute"); return -ENOSYS; } static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, const struct audio_config *config) { size_t buffer_size = samples_per_milliseconds( STUB_INPUT_BUFFER_MILLISECONDS, config->sample_rate, audio_channel_count_from_in_mask( config->channel_mask)); if (!audio_has_proportional_frames(config->format)) { // Since the audio data is not proportional choose an arbitrary size for // the buffer. buffer_size *= 4; } else { buffer_size *= audio_bytes_per_sample(config->format); } ALOGV("adev_get_input_buffer_size: %zu", buffer_size); return buffer_size; } 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) { ALOGV("adev_open_input_stream..."); *stream_in = NULL; struct stub_stream_in *in = (struct stub_stream_in *)calloc(1, sizeof(struct stub_stream_in)); if (!in) return -ENOMEM; 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->sample_rate = config->sample_rate; if (in->sample_rate == 0) in->sample_rate = STUB_DEFAULT_SAMPLE_RATE; in->channel_mask = config->channel_mask; if (in->channel_mask == AUDIO_CHANNEL_NONE) in->channel_mask = STUB_INPUT_DEFAULT_CHANNEL_MASK; in->format = config->format; if (in->format == AUDIO_FORMAT_DEFAULT) in->format = STUB_DEFAULT_AUDIO_FORMAT; in->frame_count = samples_per_milliseconds( STUB_INPUT_BUFFER_MILLISECONDS, in->sample_rate, 1); ALOGV("adev_open_input_stream: sample_rate: %u, channels: %x, format: %d," "frames: %zu", in->sample_rate, in->channel_mask, in->format, in->frame_count); *stream_in = &in->stream; return 0; } static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *in) { ALOGV("adev_close_input_stream..."); return; } static int adev_dump(const audio_hw_device_t *device, int fd) { ALOGV("adev_dump"); return 0; } static int adev_close(hw_device_t *device) { ALOGV("adev_close"); free(device); return 0; } static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device) { ALOGV("adev_open: %s", name); struct stub_audio_device *adev; if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL; adev = calloc(1, sizeof(struct stub_audio_device)); if (!adev) return -ENOMEM; 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; *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 = "Default audio HW HAL", .author = "The Android Open Source Project", .methods = &hal_module_methods, }, };