/* * 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 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #define LOG_STREAMS_TO_FILES 0 #if LOG_STREAMS_TO_FILES #include #include #include #endif // LOG_STREAMS_TO_FILES extern "C" { namespace android { // Set to 1 to enable extremely verbose logging in this module. #define SUBMIX_VERBOSE_LOGGING 0 #if SUBMIX_VERBOSE_LOGGING #define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__) #define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__) #else #define SUBMIX_ALOGV(...) #define SUBMIX_ALOGE(...) #endif // SUBMIX_VERBOSE_LOGGING // NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe(). #define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*4) // Value used to divide the MonoPipe() buffer into segments that are written to the source and // read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer // the minimum latency is the MonoPipe buffer size divided by this value. #define DEFAULT_PIPE_PERIOD_COUNT 4 // 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_SAMPLE_RATE_HZ 48000 // default sample rate // See NBAIO_Format frameworks/av/include/media/nbaio/NBAIO.h. #define DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT // A legacy user of this device does not close the input stream when it shuts down, which // results in the application opening a new input stream before closing the old input stream // handle it was previously using. Setting this value to 1 allows multiple clients to open // multiple input streams from this device. If this option is enabled, each input stream returned // is *the same stream* which means that readers will race to read data from these streams. #define ENABLE_LEGACY_INPUT_OPEN 1 // Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled. #define ENABLE_CHANNEL_CONVERSION 1 // Whether resampling is enabled. #define ENABLE_RESAMPLING 1 #if LOG_STREAMS_TO_FILES // Folder to save stream log files to. #define LOG_STREAM_FOLDER "/data/misc/media" // Log filenames for input and output streams. #define LOG_STREAM_OUT_FILENAME LOG_STREAM_FOLDER "/r_submix_out.raw" #define LOG_STREAM_IN_FILENAME LOG_STREAM_FOLDER "/r_submix_in.raw" // File permissions for stream log files. #define LOG_STREAM_FILE_PERMISSIONS (S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH) #endif // LOG_STREAMS_TO_FILES // limit for number of read error log entries to avoid spamming the logs #define MAX_READ_ERROR_LOGS 5 // Common limits macros. #ifndef min #define min(a, b) ((a) < (b) ? (a) : (b)) #endif // min #ifndef max #define max(a, b) ((a) > (b) ? (a) : (b)) #endif // max // Set *result_variable_ptr to true if value_to_find is present in the array array_to_search, // otherwise set *result_variable_ptr to false. #define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \ { \ size_t i; \ *(result_variable_ptr) = false; \ for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \ if ((value_to_find) == (array_to_search)[i]) { \ *(result_variable_ptr) = true; \ break; \ } \ } \ } // Configuration of the submix pipe. struct submix_config { // Channel mask field in this data structure is set to either input_channel_mask or // output_channel_mask depending upon the last stream to be opened on this device. struct audio_config common; // Input stream and output stream channel masks. This is required since input and output // channel bitfields are not equivalent. audio_channel_mask_t input_channel_mask; audio_channel_mask_t output_channel_mask; #if ENABLE_RESAMPLING // Input stream and output stream sample rates. uint32_t input_sample_rate; uint32_t output_sample_rate; #endif // ENABLE_RESAMPLING size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe. size_t buffer_size_frames; // Size of the audio pipe in frames. // Maximum number of frames buffered by the input and output streams. size_t buffer_period_size_frames; }; #define MAX_ROUTES 10 typedef struct route_config { struct submix_config config; char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; // 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 rsxSink; sp rsxSource; // Pointers to the current input and output stream instances. rsxSink and rsxSource are // destroyed if both and input and output streams are destroyed. struct submix_stream_out *output; struct submix_stream_in *input; #if ENABLE_RESAMPLING // Buffer used as temporary storage for resampled data prior to returning data to the output // stream. int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; #endif // ENABLE_RESAMPLING } route_config_t; struct submix_audio_device { struct audio_hw_device device; route_config_t routes[MAX_ROUTES]; // Device lock, also used to protect access to submix_audio_device from the input and output // streams. pthread_mutex_t lock; }; struct submix_stream_out { struct audio_stream_out stream; struct submix_audio_device *dev; int route_handle; bool output_standby; uint64_t frames_written; uint64_t frames_written_since_standby; #if LOG_STREAMS_TO_FILES int log_fd; #endif // LOG_STREAMS_TO_FILES }; struct submix_stream_in { struct audio_stream_in stream; struct submix_audio_device *dev; int route_handle; bool input_standby; bool output_standby_rec_thr; // 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 uint64_t read_counter_frames; #if ENABLE_LEGACY_INPUT_OPEN // Number of references to this input stream. volatile int32_t ref_count; #endif // ENABLE_LEGACY_INPUT_OPEN #if LOG_STREAMS_TO_FILES int log_fd; #endif // LOG_STREAMS_TO_FILES volatile int16_t read_error_count; }; // Determine whether the specified sample rate is supported by the submix module. static bool sample_rate_supported(const uint32_t sample_rate) { // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp. static const unsigned int supported_sample_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, }; bool return_value; SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value); return return_value; } // Determine whether the specified sample rate is supported, if it is return the specified sample // rate, otherwise return the default sample rate for the submix module. static uint32_t get_supported_sample_rate(uint32_t sample_rate) { return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ; } // Determine whether the specified channel in mask is supported by the submix module. static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask) { // Set of channel in masks supported by Format_from_SR_C() // frameworks/av/media/libnbaio/NAIO.cpp. static const audio_channel_mask_t supported_channel_in_masks[] = { AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO, }; bool return_value; SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value); return return_value; } // Determine whether the specified channel in mask is supported, if it is return the specified // channel in mask, otherwise return the default channel in mask for the submix module. static audio_channel_mask_t get_supported_channel_in_mask( const audio_channel_mask_t channel_in_mask) { return channel_in_mask_supported(channel_in_mask) ? channel_in_mask : static_cast(AUDIO_CHANNEL_IN_STEREO); } // Determine whether the specified channel out mask is supported by the submix module. static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask) { // Set of channel out masks supported by Format_from_SR_C() // frameworks/av/media/libnbaio/NAIO.cpp. static const audio_channel_mask_t supported_channel_out_masks[] = { AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO, }; bool return_value; SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value); return return_value; } // Determine whether the specified channel out mask is supported, if it is return the specified // channel out mask, otherwise return the default channel out mask for the submix module. static audio_channel_mask_t get_supported_channel_out_mask( const audio_channel_mask_t channel_out_mask) { return channel_out_mask_supported(channel_out_mask) ? channel_out_mask : static_cast(AUDIO_CHANNEL_OUT_STEREO); } // Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the // structure. static struct submix_stream_out * audio_stream_out_get_submix_stream_out( struct audio_stream_out * const stream) { ALOG_ASSERT(stream); return reinterpret_cast(reinterpret_cast(stream) - offsetof(struct submix_stream_out, stream)); } // Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure. static struct submix_stream_out * audio_stream_get_submix_stream_out( struct audio_stream * const stream) { ALOG_ASSERT(stream); return audio_stream_out_get_submix_stream_out( reinterpret_cast(stream)); } // Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the // structure. static struct submix_stream_in * audio_stream_in_get_submix_stream_in( struct audio_stream_in * const stream) { ALOG_ASSERT(stream); return reinterpret_cast(reinterpret_cast(stream) - offsetof(struct submix_stream_in, stream)); } // Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure. static struct submix_stream_in * audio_stream_get_submix_stream_in( struct audio_stream * const stream) { ALOG_ASSERT(stream); return audio_stream_in_get_submix_stream_in( reinterpret_cast(stream)); } // Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within // the structure. static struct submix_audio_device * audio_hw_device_get_submix_audio_device( struct audio_hw_device *device) { ALOG_ASSERT(device); return reinterpret_cast(reinterpret_cast(device) - offsetof(struct submix_audio_device, device)); } // Compare an audio_config with input channel mask and an audio_config with output channel mask // returning false if they do *not* match, true otherwise. static bool audio_config_compare(const audio_config * const input_config, const audio_config * const output_config) { #if !ENABLE_CHANNEL_CONVERSION const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask); const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask); if (input_channels != output_channels) { ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d", input_channels, output_channels); return false; } #endif // !ENABLE_CHANNEL_CONVERSION #if ENABLE_RESAMPLING if (input_config->sample_rate != output_config->sample_rate && audio_channel_count_from_in_mask(input_config->channel_mask) != 1) { #else if (input_config->sample_rate != output_config->sample_rate) { #endif // ENABLE_RESAMPLING ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul", input_config->sample_rate, output_config->sample_rate); return false; } if (input_config->format != output_config->format) { ALOGE("audio_config_compare() format mismatch %x vs. %x", input_config->format, output_config->format); return false; } // This purposely ignores offload_info as it's not required for the submix device. return true; } // If one doesn't exist, create a pipe for the submix audio device rsxadev of size // buffer_size_frames and optionally associate "in" or "out" with the submix audio device. // Must be called with lock held on the submix_audio_device static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev, const struct audio_config * const config, const size_t buffer_size_frames, const uint32_t buffer_period_count, struct submix_stream_in * const in, struct submix_stream_out * const out, const char *address, int route_idx) { ALOG_ASSERT(in || out); ALOG_ASSERT(route_idx > -1); ALOG_ASSERT(route_idx < MAX_ROUTES); ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx); // Save a reference to the specified input or output stream and the associated channel // mask. if (in) { in->route_handle = route_idx; rsxadev->routes[route_idx].input = in; rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask; #if ENABLE_RESAMPLING rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate; // If the output isn't configured yet, set the output sample rate to the maximum supported // sample rate such that the smallest possible input buffer is created, and put a default // value for channel count if (!rsxadev->routes[route_idx].output) { rsxadev->routes[route_idx].config.output_sample_rate = 48000; rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO; } #endif // ENABLE_RESAMPLING } if (out) { out->route_handle = route_idx; rsxadev->routes[route_idx].output = out; rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask; #if ENABLE_RESAMPLING rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate; #endif // ENABLE_RESAMPLING } // Save the address strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN); ALOGD(" now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx); // If a pipe isn't associated with the device, create one. if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL) { struct submix_config * const device_config = &rsxadev->routes[route_idx].config; uint32_t channel_count; if (out) channel_count = audio_channel_count_from_out_mask(config->channel_mask); else channel_count = audio_channel_count_from_in_mask(config->channel_mask); #if ENABLE_CHANNEL_CONVERSION // If channel conversion is enabled, allocate enough space for the maximum number of // possible channels stored in the pipe for the situation when the number of channels in // the output stream don't match the number in the input stream. const uint32_t pipe_channel_count = max(channel_count, 2); #else const uint32_t pipe_channel_count = channel_count; #endif // ENABLE_CHANNEL_CONVERSION const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count, config->format); const NBAIO_Format offers[1] = {format}; size_t numCounterOffers = 0; // Create a MonoPipe with optional blocking set to true. MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/); // Negotiation between the source and sink cannot fail as the device open operation // creates both ends of the pipe using the same audio format. 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); ALOGV("submix_audio_device_create_pipe_l(): created pipe"); // Save references to the source and sink. ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL); ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL); rsxadev->routes[route_idx].rsxSink = sink; rsxadev->routes[route_idx].rsxSource = source; // Store the sanitized audio format in the device so that it's possible to determine // the format of the pipe source when opening the input device. memcpy(&device_config->common, config, sizeof(device_config->common)); device_config->buffer_size_frames = sink->maxFrames(); device_config->buffer_period_size_frames = device_config->buffer_size_frames / buffer_period_count; if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream); if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream); #if ENABLE_CHANNEL_CONVERSION // Calculate the pipe frame size based upon the number of channels. device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) / channel_count; #endif // ENABLE_CHANNEL_CONVERSION SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, " "period size %zd", device_config->pipe_frame_size, device_config->buffer_size_frames, device_config->buffer_period_size_frames); } } // Release references to the sink and source. Input and output threads may maintain references // to these objects via StrongPointer (sp and sp) which they can use // before they shutdown. // Must be called with lock held on the submix_audio_device static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev, int route_idx) { ALOG_ASSERT(route_idx > -1); ALOG_ASSERT(route_idx < MAX_ROUTES); ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx, rsxadev->routes[route_idx].address); if (rsxadev->routes[route_idx].rsxSink != 0) { rsxadev->routes[route_idx].rsxSink.clear(); rsxadev->routes[route_idx].rsxSink = 0; } if (rsxadev->routes[route_idx].rsxSource != 0) { rsxadev->routes[route_idx].rsxSource.clear(); rsxadev->routes[route_idx].rsxSource = 0; } memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN); #ifdef ENABLE_RESAMPLING memset(rsxadev->routes[route_idx].resampler_buffer, 0, sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); #endif } // Remove references to the specified input and output streams. When the device no longer // references input and output streams destroy the associated pipe. // Must be called with lock held on the submix_audio_device static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev, const struct submix_stream_in * const in, const struct submix_stream_out * const out) { MonoPipe* sink; ALOGV("submix_audio_device_destroy_pipe_l()"); int route_idx = -1; if (in != NULL) { #if ENABLE_LEGACY_INPUT_OPEN const_cast(in)->ref_count--; route_idx = in->route_handle; ALOG_ASSERT(rsxadev->routes[route_idx].input == in); if (in->ref_count == 0) { rsxadev->routes[route_idx].input = NULL; } ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count); #else rsxadev->input = NULL; #endif // ENABLE_LEGACY_INPUT_OPEN } if (out != NULL) { route_idx = out->route_handle; ALOG_ASSERT(rsxadev->routes[route_idx].output == out); rsxadev->routes[route_idx].output = NULL; } if (route_idx != -1 && rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) { submix_audio_device_release_pipe_l(rsxadev, route_idx); ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed"); } } // Sanitize the user specified audio config for a submix input / output stream. static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format) { config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) : get_supported_channel_out_mask(config->channel_mask); config->sample_rate = get_supported_sample_rate(config->sample_rate); config->format = DEFAULT_FORMAT; } // Verify a submix input or output stream can be opened. // Must be called with lock held on the submix_audio_device static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev, int route_idx, const struct audio_config * const config, const bool opening_input) { bool input_open; bool output_open; audio_config pipe_config; // Query the device for the current audio config and whether input and output streams are open. output_open = rsxadev->routes[route_idx].output != NULL; input_open = rsxadev->routes[route_idx].input != NULL; memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config)); // If the stream is already open, don't open it again. if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" : "Output"); return false; } SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x " "%s_channel_mask=%x", config->sample_rate, config->format, opening_input ? "in" : "out", config->channel_mask); // If either stream is open, verify the existing audio config the pipe matches the user // specified config. if (input_open || output_open) { const audio_config * const input_config = opening_input ? config : &pipe_config; const audio_config * const output_config = opening_input ? &pipe_config : config; // Get the channel mask of the open device. pipe_config.channel_mask = opening_input ? rsxadev->routes[route_idx].config.output_channel_mask : rsxadev->routes[route_idx].config.input_channel_mask; if (!audio_config_compare(input_config, output_config)) { ALOGE("submix_open_validate_l(): Unsupported format."); return false; } } return true; } // Must be called with lock held on the submix_audio_device static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev, const char* address, /*in*/ int *idx /*out*/) { // Do we already have a route for this address int route_idx = -1; int route_empty_idx = -1; // index of an empty route slot that can be used if needed for (int i=0 ; i < MAX_ROUTES ; i++) { if (strcmp(rsxadev->routes[i].address, "") == 0) { route_empty_idx = i; } if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { route_idx = i; break; } } if ((route_idx == -1) && (route_empty_idx == -1)) { ALOGE("Cannot create new route for address %s, max number of routes reached", address); return -ENOMEM; } if (route_idx == -1) { route_idx = route_empty_idx; } *idx = route_idx; return OK; } // Calculate the maximum size of the pipe buffer in frames for the specified stream. static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, const struct submix_config *config, const size_t pipe_frames, const size_t stream_frame_size) { const size_t pipe_frame_size = config->pipe_frame_size; const size_t max_frame_size = max(stream_frame_size, pipe_frame_size); return (pipe_frames * config->pipe_frame_size) / max_frame_size; } /* audio HAL functions */ static uint32_t out_get_sample_rate(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast(stream)); #if ENABLE_RESAMPLING const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate; #else const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate; #endif // ENABLE_RESAMPLING SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s", out_rate, out->dev->routes[out->route_handle].address); return out_rate; } static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) { struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); #if ENABLE_RESAMPLING // The sample rate of the stream can't be changed once it's set since this would change the // output buffer size and hence break playback to the shared pipe. if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) { ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from " "%u to %u for addr %s", out->dev->routes[out->route_handle].config.output_sample_rate, rate, out->dev->routes[out->route_handle].address); return -ENOSYS; } #endif // ENABLE_RESAMPLING if (!sample_rate_supported(rate)) { ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); return -ENOSYS; } SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); out->dev->routes[out->route_handle].config.common.sample_rate = rate; return 0; } static size_t out_get_buffer_size(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast(stream)); const struct submix_config * const config = &out->dev->routes[out->route_handle].config; const size_t stream_frame_size = audio_stream_out_frame_size((const struct audio_stream_out *)stream); const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( stream, config, config->buffer_period_size_frames, stream_frame_size); const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, buffer_size_frames); return buffer_size_bytes; } static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast(stream)); uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask; SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); return channel_mask; } static audio_format_t out_get_format(const struct audio_stream *stream) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( const_cast(stream)); const audio_format_t format = out->dev->routes[out->route_handle].config.common.format; SUBMIX_ALOGV("out_get_format() returns %x", format); return format; } static int out_set_format(struct audio_stream *stream, audio_format_t format) { const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); if (format != out->dev->routes[out->route_handle].config.common.format) { ALOGE("out_set_format(format=%x) format unsupported", format); return -ENOSYS; } SUBMIX_ALOGV("out_set_format(format=%x)", format); return 0; } static int out_standby(struct audio_stream *stream) { ALOGI("out_standby()"); struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); struct submix_audio_device * const rsxadev = out->dev; pthread_mutex_lock(&rsxadev->lock); out->output_standby = true; out->frames_written_since_standby = 0; pthread_mutex_unlock(&rsxadev->lock); return 0; } static int out_dump(const struct audio_stream *stream, int fd) { (void)stream; (void)fd; return 0; } static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) { int exiting = -1; AudioParameter parms = AudioParameter(String8(kvpairs)); SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", 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)) { struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_out(stream)->dev; pthread_mutex_lock(&rsxadev->lock); { // using the sink sp sink = rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle] .rsxSink; if (sink == NULL) { pthread_mutex_unlock(&rsxadev->lock); return 0; } ALOGD("out_set_parameters(): shutting down MonoPipe sink"); sink->shutdown(true); } // done using the sink pthread_mutex_unlock(&rsxadev->lock); } return 0; } static char * out_get_parameters(const struct audio_stream *stream, const char *keys) { (void)stream; (void)keys; return strdup(""); } static uint32_t out_get_latency(const struct audio_stream_out *stream) { const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( const_cast(stream)); const struct submix_config * const config = &out->dev->routes[out->route_handle].config; const size_t stream_frame_size = audio_stream_out_frame_size(stream); const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( &stream->common, config, config->buffer_size_frames, stream_frame_size); const uint32_t sample_rate = out_get_sample_rate(&stream->common); const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate; SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u", latency_ms, buffer_size_frames, sample_rate); return latency_ms; } static int out_set_volume(struct audio_stream_out *stream, float left, float right) { (void)stream; (void)left; (void)right; return -ENOSYS; } static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes) { SUBMIX_ALOGV("out_write(bytes=%zd)", bytes); ssize_t written_frames = 0; const size_t frame_size = audio_stream_out_frame_size(stream); struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); struct submix_audio_device * const rsxadev = out->dev; const size_t frames = bytes / frame_size; pthread_mutex_lock(&rsxadev->lock); out->output_standby = false; sp sink = rsxadev->routes[out->route_handle].rsxSink; if (sink != NULL) { if (sink->isShutdown()) { sink.clear(); pthread_mutex_unlock(&rsxadev->lock); SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write."); // 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; } } else { pthread_mutex_unlock(&rsxadev->lock); ALOGE("out_write without a pipe!"); ALOG_ASSERT("out_write without a pipe!"); return 0; } // If the write to the sink would block when no input stream is present, flush enough frames // from the pipe to make space to write the most recent data. { const size_t availableToWrite = sink->availableToWrite(); sp source = rsxadev->routes[out->route_handle].rsxSource; if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) { static uint8_t flush_buffer[64]; const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; size_t frames_to_flush_from_source = frames - availableToWrite; SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking", frames_to_flush_from_source); while (frames_to_flush_from_source) { const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); frames_to_flush_from_source -= flush_size; // read does not block source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS); } } } pthread_mutex_unlock(&rsxadev->lock); written_frames = sink->write(buffer, frames); #if LOG_STREAMS_TO_FILES if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size); #endif // LOG_STREAMS_TO_FILES if (written_frames < 0) { if (written_frames == (ssize_t)NEGOTIATE) { ALOGE("out_write() write to pipe returned NEGOTIATE"); pthread_mutex_lock(&rsxadev->lock); sink.clear(); pthread_mutex_unlock(&rsxadev->lock); written_frames = 0; return 0; } else { // write() returned UNDERRUN or WOULD_BLOCK, retry ALOGE("out_write() write to pipe returned unexpected %zd", written_frames); written_frames = sink->write(buffer, frames); } } pthread_mutex_lock(&rsxadev->lock); sink.clear(); if (written_frames > 0) { out->frames_written_since_standby += written_frames; out->frames_written += written_frames; } pthread_mutex_unlock(&rsxadev->lock); if (written_frames < 0) { ALOGE("out_write() failed writing to pipe with %zd", written_frames); return 0; } const ssize_t written_bytes = written_frames * frame_size; SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames); return written_bytes; } static int out_get_presentation_position(const struct audio_stream_out *stream, uint64_t *frames, struct timespec *timestamp) { if (stream == NULL || frames == NULL || timestamp == NULL) { return -EINVAL; } const submix_stream_out *out = audio_stream_out_get_submix_stream_out( const_cast(stream)); struct submix_audio_device * const rsxadev = out->dev; int ret = -EWOULDBLOCK; pthread_mutex_lock(&rsxadev->lock); const ssize_t frames_in_pipe = rsxadev->routes[out->route_handle].rsxSource->availableToRead(); if (CC_UNLIKELY(frames_in_pipe < 0)) { *frames = out->frames_written; ret = 0; } else if (out->frames_written >= (uint64_t)frames_in_pipe) { *frames = out->frames_written - frames_in_pipe; ret = 0; } pthread_mutex_unlock(&rsxadev->lock); if (ret == 0) { clock_gettime(CLOCK_MONOTONIC, timestamp); } SUBMIX_ALOGV("out_get_presentation_position() got frames=%llu timestamp sec=%llu", frames ? *frames : -1, timestamp ? timestamp->tv_sec : -1); return ret; } static int out_get_render_position(const struct audio_stream_out *stream, uint32_t *dsp_frames) { if (stream == NULL || dsp_frames == NULL) { return -EINVAL; } const submix_stream_out *out = audio_stream_out_get_submix_stream_out( const_cast(stream)); struct submix_audio_device * const rsxadev = out->dev; pthread_mutex_lock(&rsxadev->lock); const ssize_t frames_in_pipe = rsxadev->routes[out->route_handle].rsxSource->availableToRead(); if (CC_UNLIKELY(frames_in_pipe < 0)) { *dsp_frames = (uint32_t)out->frames_written_since_standby; } else { *dsp_frames = out->frames_written_since_standby > (uint64_t) frames_in_pipe ? (uint32_t)(out->frames_written_since_standby - frames_in_pipe) : 0; } pthread_mutex_unlock(&rsxadev->lock); return 0; } static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { (void)stream; (void)effect; return 0; } static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { (void)stream; (void)effect; return 0; } static int out_get_next_write_timestamp(const struct audio_stream_out *stream, int64_t *timestamp) { (void)stream; (void)timestamp; return -EINVAL; } /** audio_stream_in implementation **/ static uint32_t in_get_sample_rate(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast(stream)); #if ENABLE_RESAMPLING const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate; #else const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate; #endif // ENABLE_RESAMPLING SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); return rate; } static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); #if ENABLE_RESAMPLING // The sample rate of the stream can't be changed once it's set since this would change the // input buffer size and hence break recording from the shared pipe. if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) { ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from " "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate); return -ENOSYS; } #endif // ENABLE_RESAMPLING if (!sample_rate_supported(rate)) { ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); return -ENOSYS; } in->dev->routes[in->route_handle].config.common.sample_rate = rate; SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); return 0; } static size_t in_get_buffer_size(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast(stream)); const struct submix_config * const config = &in->dev->routes[in->route_handle].config; const size_t stream_frame_size = audio_stream_in_frame_size((const struct audio_stream_in *)stream); size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( stream, config, config->buffer_period_size_frames, stream_frame_size); #if ENABLE_RESAMPLING // Scale the size of the buffer based upon the maximum number of frames that could be returned // given the ratio of output to input sample rate. buffer_size_frames = (size_t)(((float)buffer_size_frames * (float)config->input_sample_rate) / (float)config->output_sample_rate); #endif // ENABLE_RESAMPLING const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, buffer_size_frames); return buffer_size_bytes; } static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast(stream)); const audio_channel_mask_t channel_mask = in->dev->routes[in->route_handle].config.input_channel_mask; SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); return channel_mask; } static audio_format_t in_get_format(const struct audio_stream *stream) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( const_cast(stream)); const audio_format_t format = in->dev->routes[in->route_handle].config.common.format; SUBMIX_ALOGV("in_get_format() returns %x", format); return format; } static int in_set_format(struct audio_stream *stream, audio_format_t format) { const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); if (format != in->dev->routes[in->route_handle].config.common.format) { ALOGE("in_set_format(format=%x) format unsupported", format); return -ENOSYS; } SUBMIX_ALOGV("in_set_format(format=%x)", format); return 0; } static int in_standby(struct audio_stream *stream) { ALOGI("in_standby()"); struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); struct submix_audio_device * const rsxadev = in->dev; pthread_mutex_lock(&rsxadev->lock); in->input_standby = true; pthread_mutex_unlock(&rsxadev->lock); return 0; } static int in_dump(const struct audio_stream *stream, int fd) { (void)stream; (void)fd; return 0; } static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) { (void)stream; (void)kvpairs; return 0; } static char * in_get_parameters(const struct audio_stream *stream, const char *keys) { (void)stream; (void)keys; return strdup(""); } static int in_set_gain(struct audio_stream_in *stream, float gain) { (void)stream; (void)gain; return 0; } static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes) { struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); struct submix_audio_device * const rsxadev = in->dev; struct audio_config *format; const size_t frame_size = audio_stream_in_frame_size(stream); const size_t frames_to_read = bytes / frame_size; SUBMIX_ALOGV("in_read bytes=%zu", bytes); pthread_mutex_lock(&rsxadev->lock); const bool output_standby = rsxadev->routes[in->route_handle].output == NULL ? true : rsxadev->routes[in->route_handle].output->output_standby; const bool output_standby_transition = (in->output_standby_rec_thr != output_standby); in->output_standby_rec_thr = output_standby; if (in->input_standby || output_standby_transition) { in->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; size_t remaining_frames = frames_to_read; { // about to read from audio source sp source = rsxadev->routes[in->route_handle].rsxSource; if (source == NULL) { in->read_error_count++;// ok if it rolls over ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS, "no audio pipe yet we're trying to read! (not all errors will be logged)"); pthread_mutex_unlock(&rsxadev->lock); usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common)); memset(buffer, 0, bytes); return bytes; } pthread_mutex_unlock(&rsxadev->lock); // read the data from the pipe (it's non blocking) int attempts = 0; char* buff = (char*)buffer; #if ENABLE_CHANNEL_CONVERSION // Determine whether channel conversion is required. const uint32_t input_channels = audio_channel_count_from_in_mask( rsxadev->routes[in->route_handle].config.input_channel_mask); const uint32_t output_channels = audio_channel_count_from_out_mask( rsxadev->routes[in->route_handle].config.output_channel_mask); if (input_channels != output_channels) { SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " "input channels", output_channels, input_channels); // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == AUDIO_FORMAT_PCM_16_BIT); ALOG_ASSERT((input_channels == 1 && output_channels == 2) || (input_channels == 2 && output_channels == 1)); } #endif // ENABLE_CHANNEL_CONVERSION #if ENABLE_RESAMPLING const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); const uint32_t output_sample_rate = rsxadev->routes[in->route_handle].config.output_sample_rate; const size_t resampler_buffer_size_frames = sizeof(rsxadev->routes[in->route_handle].resampler_buffer) / sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]); float resampler_ratio = 1.0f; // Determine whether resampling is required. if (input_sample_rate != output_sample_rate) { resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; // Only support 16-bit PCM mono resampling. // NOTE: Resampling is performed after the channel conversion step. ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == AUDIO_FORMAT_PCM_16_BIT); ALOG_ASSERT(audio_channel_count_from_in_mask( rsxadev->routes[in->route_handle].config.input_channel_mask) == 1); } #endif // ENABLE_RESAMPLING while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { ssize_t frames_read = -1977; size_t read_frames = remaining_frames; #if ENABLE_RESAMPLING char* const saved_buff = buff; if (resampler_ratio != 1.0f) { // Calculate the number of frames from the pipe that need to be read to generate // the data for the input stream read. const size_t frames_required_for_resampler = (size_t)( (float)read_frames * (float)resampler_ratio); read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); // Read into the resampler buffer. buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; } #endif // ENABLE_RESAMPLING #if ENABLE_CHANNEL_CONVERSION if (output_channels == 1 && input_channels == 2) { // Need to read half the requested frames since the converted output // data will take twice the space (mono->stereo). read_frames /= 2; } #endif // ENABLE_CHANNEL_CONVERSION SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); frames_read = source->read(buff, read_frames, AudioBufferProvider::kInvalidPTS); SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); #if ENABLE_CHANNEL_CONVERSION // Perform in-place channel conversion. // NOTE: In the following "input stream" refers to the data returned by this function // and "output stream" refers to the data read from the pipe. if (input_channels != output_channels && frames_read > 0) { int16_t *data = (int16_t*)buff; if (output_channels == 2 && input_channels == 1) { // Offset into the output stream data in samples. ssize_t output_stream_offset = 0; for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; input_stream_frame++, output_stream_offset += 2) { // Average the content from both channels. data[input_stream_frame] = ((int32_t)data[output_stream_offset] + (int32_t)data[output_stream_offset + 1]) / 2; } } else if (output_channels == 1 && input_channels == 2) { // Offset into the input stream data in samples. ssize_t input_stream_offset = (frames_read - 1) * 2; for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; output_stream_frame--, input_stream_offset -= 2) { const short sample = data[output_stream_frame]; data[input_stream_offset] = sample; data[input_stream_offset + 1] = sample; } } } #endif // ENABLE_CHANNEL_CONVERSION #if ENABLE_RESAMPLING if (resampler_ratio != 1.0f) { SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); const int16_t * const data = (int16_t*)buff; int16_t * const resampled_buffer = (int16_t*)saved_buff; // Resample with *no* filtering - if the data from the ouptut stream was really // sampled at a different rate this will result in very nasty aliasing. const float output_stream_frames = (float)frames_read; size_t input_stream_frame = 0; for (float output_stream_frame = 0.0f; output_stream_frame < output_stream_frames && input_stream_frame < remaining_frames; output_stream_frame += resampler_ratio, input_stream_frame++) { resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; } ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); frames_read = input_stream_frame; buff = saved_buff; } #endif // ENABLE_RESAMPLING if (frames_read > 0) { #if LOG_STREAMS_TO_FILES if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); #endif // LOG_STREAMS_TO_FILES remaining_frames -= frames_read; buff += frames_read * frame_size; SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu", attempts, frames_read, remaining_frames); } else { attempts++; SUBMIX_ALOGE(" in_read read returned %zd", frames_read); usleep(READ_ATTEMPT_SLEEP_MS * 1000); } } // done using the source pthread_mutex_lock(&rsxadev->lock); source.clear(); pthread_mutex_unlock(&rsxadev->lock); } if (remaining_frames > 0) { const size_t remaining_bytes = remaining_frames * frame_size; SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames); memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes); } // 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); SUBMIX_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); } } SUBMIX_ALOGV("in_read returns %zu", bytes); return bytes; } static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) { (void)stream; return 0; } static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { (void)stream; (void)effect; return 0; } static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { (void)stream; (void)effect; return 0; } static int adev_open_output_stream(struct audio_hw_device *dev, audio_io_handle_t handle, audio_devices_t devices, audio_output_flags_t flags, struct audio_config *config, struct audio_stream_out **stream_out, const char *address) { struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); ALOGD("adev_open_output_stream(address=%s)", address); struct submix_stream_out *out; bool force_pipe_creation = false; (void)handle; (void)devices; (void)flags; *stream_out = NULL; // Make sure it's possible to open the device given the current audio config. submix_sanitize_config(config, false); int route_idx = -1; pthread_mutex_lock(&rsxadev->lock); status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); if (res != OK) { ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); pthread_mutex_unlock(&rsxadev->lock); return res; } if (!submix_open_validate_l(rsxadev, route_idx, config, false)) { ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address); pthread_mutex_unlock(&rsxadev->lock); return -EINVAL; } out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); if (!out) { pthread_mutex_unlock(&rsxadev->lock); return -ENOMEM; } // Initialize the function pointer tables (v-tables). out->stream.common.get_sample_rate = out_get_sample_rate; out->stream.common.set_sample_rate = out_set_sample_rate; out->stream.common.get_buffer_size = out_get_buffer_size; out->stream.common.get_channels = out_get_channels; out->stream.common.get_format = out_get_format; out->stream.common.set_format = out_set_format; out->stream.common.standby = out_standby; out->stream.common.dump = out_dump; out->stream.common.set_parameters = out_set_parameters; out->stream.common.get_parameters = out_get_parameters; out->stream.common.add_audio_effect = out_add_audio_effect; out->stream.common.remove_audio_effect = out_remove_audio_effect; out->stream.get_latency = out_get_latency; out->stream.set_volume = out_set_volume; out->stream.write = out_write; out->stream.get_render_position = out_get_render_position; out->stream.get_next_write_timestamp = out_get_next_write_timestamp; out->stream.get_presentation_position = out_get_presentation_position; #if ENABLE_RESAMPLING // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits // writes correctly. force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate != config->sample_rate; #endif // ENABLE_RESAMPLING // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so // that it's recreated. if ((rsxadev->routes[route_idx].rsxSink != NULL && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) { submix_audio_device_release_pipe_l(rsxadev, route_idx); } // Store a pointer to the device from the output stream. out->dev = rsxadev; // Initialize the pipe. ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx); submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx); #if LOG_STREAMS_TO_FILES out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, LOG_STREAM_FILE_PERMISSIONS); ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s", strerror(errno)); ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd); #endif // LOG_STREAMS_TO_FILES // Return the output stream. *stream_out = &out->stream; pthread_mutex_unlock(&rsxadev->lock); return 0; } static void adev_close_output_stream(struct audio_hw_device *dev, struct audio_stream_out *stream) { struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( const_cast(dev)); struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); pthread_mutex_lock(&rsxadev->lock); ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address); submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out); #if LOG_STREAMS_TO_FILES if (out->log_fd >= 0) close(out->log_fd); #endif // LOG_STREAMS_TO_FILES pthread_mutex_unlock(&rsxadev->lock); free(out); } static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) { (void)dev; (void)kvpairs; return -ENOSYS; } static char * adev_get_parameters(const struct audio_hw_device *dev, const char *keys) { (void)dev; (void)keys; return strdup("");; } static int adev_init_check(const struct audio_hw_device *dev) { ALOGI("adev_init_check()"); (void)dev; return 0; } static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) { (void)dev; (void)volume; return -ENOSYS; } static int adev_set_master_volume(struct audio_hw_device *dev, float volume) { (void)dev; (void)volume; return -ENOSYS; } static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) { (void)dev; (void)volume; return -ENOSYS; } static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) { (void)dev; (void)muted; return -ENOSYS; } static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) { (void)dev; (void)muted; return -ENOSYS; } static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) { (void)dev; (void)mode; return 0; } static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) { (void)dev; (void)state; return -ENOSYS; } static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) { (void)dev; (void)state; return -ENOSYS; } static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, const struct audio_config *config) { if (audio_is_linear_pcm(config->format)) { size_t max_buffer_period_size_frames = 0; struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( const_cast(dev)); // look for the largest buffer period size for (int i = 0 ; i < MAX_ROUTES ; i++) { if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames) { max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames; } } const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * audio_bytes_per_sample(config->format); const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes; SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames", buffer_size, buffer_period_size_frames); return buffer_size; } return 0; } 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, audio_source_t source __unused) { struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); struct submix_stream_in *in; ALOGD("adev_open_input_stream(addr=%s)", address); (void)handle; (void)devices; *stream_in = NULL; // Do we already have a route for this address int route_idx = -1; pthread_mutex_lock(&rsxadev->lock); status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); if (res != OK) { ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); pthread_mutex_unlock(&rsxadev->lock); return res; } // Make sure it's possible to open the device given the current audio config. submix_sanitize_config(config, true); if (!submix_open_validate_l(rsxadev, route_idx, config, true)) { ALOGE("adev_open_input_stream(): Unable to open input stream."); pthread_mutex_unlock(&rsxadev->lock); return -EINVAL; } #if ENABLE_LEGACY_INPUT_OPEN in = rsxadev->routes[route_idx].input; if (in) { in->ref_count++; sp sink = rsxadev->routes[route_idx].rsxSink; ALOG_ASSERT(sink != NULL); // If the sink has been shutdown, delete the pipe. if (sink != NULL) { if (sink->isShutdown()) { ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d", in->ref_count); submix_audio_device_release_pipe_l(rsxadev, in->route_handle); } else { ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count); } } else { ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count); } } #else in = NULL; #endif // ENABLE_LEGACY_INPUT_OPEN if (!in) { in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); if (!in) return -ENOMEM; in->ref_count = 1; // Initialize the function pointer tables (v-tables). in->stream.common.get_sample_rate = in_get_sample_rate; in->stream.common.set_sample_rate = in_set_sample_rate; in->stream.common.get_buffer_size = in_get_buffer_size; in->stream.common.get_channels = in_get_channels; in->stream.common.get_format = in_get_format; in->stream.common.set_format = in_set_format; in->stream.common.standby = in_standby; in->stream.common.dump = in_dump; in->stream.common.set_parameters = in_set_parameters; in->stream.common.get_parameters = in_get_parameters; in->stream.common.add_audio_effect = in_add_audio_effect; in->stream.common.remove_audio_effect = in_remove_audio_effect; in->stream.set_gain = in_set_gain; in->stream.read = in_read; in->stream.get_input_frames_lost = in_get_input_frames_lost; in->dev = rsxadev; #if LOG_STREAMS_TO_FILES in->log_fd = -1; #endif } // Initialize the input stream. in->read_counter_frames = 0; in->input_standby = true; if (rsxadev->routes[route_idx].output != NULL) { in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby; } else { in->output_standby_rec_thr = true; } in->read_error_count = 0; // Initialize the pipe. ALOGV("adev_open_input_stream(): about to create pipe"); submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx); #if LOG_STREAMS_TO_FILES if (in->log_fd >= 0) close(in->log_fd); in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, LOG_STREAM_FILE_PERMISSIONS); ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", strerror(errno)); ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd); #endif // LOG_STREAMS_TO_FILES // Return the input stream. *stream_in = &in->stream; pthread_mutex_unlock(&rsxadev->lock); return 0; } static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream) { struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); ALOGD("adev_close_input_stream()"); pthread_mutex_lock(&rsxadev->lock); submix_audio_device_destroy_pipe_l(rsxadev, in, NULL); #if LOG_STREAMS_TO_FILES if (in->log_fd >= 0) close(in->log_fd); #endif // LOG_STREAMS_TO_FILES #if ENABLE_LEGACY_INPUT_OPEN if (in->ref_count == 0) free(in); #else free(in); #endif // ENABLE_LEGACY_INPUT_OPEN pthread_mutex_unlock(&rsxadev->lock); } static int adev_dump(const audio_hw_device_t *device, int fd) { const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device); reinterpret_cast( reinterpret_cast(device) - offsetof(struct submix_audio_device, device)); char msg[100]; int n = sprintf(msg, "\nReroute submix audio module:\n"); write(fd, &msg, n); for (int i=0 ; i < MAX_ROUTES ; i++) { n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i, rsxadev->routes[i].config.input_sample_rate, rsxadev->routes[i].config.output_sample_rate, rsxadev->routes[i].address); write(fd, &msg, n); } 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; for (int i=0 ; i < MAX_ROUTES ; i++) { memset(&rsxadev->routes[i], 0, sizeof(route_config)); strcpy(rsxadev->routes[i].address, ""); } *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"