Home | History | Annotate | Download | only in audio_remote_submix
      1 /*
      2  * Copyright (C) 2012 The Android Open Source Project
      3  *
      4  * Licensed under the Apache License, Version 2.0 (the "License");
      5  * you may not use this file except in compliance with the License.
      6  * You may obtain a copy of the License at
      7  *
      8  *      http://www.apache.org/licenses/LICENSE-2.0
      9  *
     10  * Unless required by applicable law or agreed to in writing, software
     11  * distributed under the License is distributed on an "AS IS" BASIS,
     12  * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
     13  * See the License for the specific language governing permissions and
     14  * limitations under the License.
     15  */
     16 
     17 #define LOG_TAG "r_submix"
     18 //#define LOG_NDEBUG 0
     19 
     20 #include <errno.h>
     21 #include <pthread.h>
     22 #include <stdint.h>
     23 #include <stdlib.h>
     24 #include <sys/param.h>
     25 #include <sys/time.h>
     26 #include <sys/limits.h>
     27 
     28 #include <cutils/compiler.h>
     29 #include <cutils/log.h>
     30 #include <cutils/properties.h>
     31 #include <cutils/str_parms.h>
     32 
     33 #include <hardware/audio.h>
     34 #include <hardware/hardware.h>
     35 #include <system/audio.h>
     36 
     37 #include <media/AudioParameter.h>
     38 #include <media/AudioBufferProvider.h>
     39 #include <media/nbaio/MonoPipe.h>
     40 #include <media/nbaio/MonoPipeReader.h>
     41 
     42 #include <utils/String8.h>
     43 
     44 #define LOG_STREAMS_TO_FILES 0
     45 #if LOG_STREAMS_TO_FILES
     46 #include <fcntl.h>
     47 #include <stdio.h>
     48 #include <sys/stat.h>
     49 #endif // LOG_STREAMS_TO_FILES
     50 
     51 extern "C" {
     52 
     53 namespace android {
     54 
     55 // Set to 1 to enable extremely verbose logging in this module.
     56 #define SUBMIX_VERBOSE_LOGGING 0
     57 #if SUBMIX_VERBOSE_LOGGING
     58 #define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__)
     59 #define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__)
     60 #else
     61 #define SUBMIX_ALOGV(...)
     62 #define SUBMIX_ALOGE(...)
     63 #endif // SUBMIX_VERBOSE_LOGGING
     64 
     65 // NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe().
     66 #define DEFAULT_PIPE_SIZE_IN_FRAMES  (1024*4)
     67 // Value used to divide the MonoPipe() buffer into segments that are written to the source and
     68 // read from the sink.  The maximum latency of the device is the size of the MonoPipe's buffer
     69 // the minimum latency is the MonoPipe buffer size divided by this value.
     70 #define DEFAULT_PIPE_PERIOD_COUNT    4
     71 // The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to
     72 //   the duration of a record buffer at the current record sample rate (of the device, not of
     73 //   the recording itself). Here we have:
     74 //      3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms
     75 #define MAX_READ_ATTEMPTS            3
     76 #define READ_ATTEMPT_SLEEP_MS        5 // 5ms between two read attempts when pipe is empty
     77 #define DEFAULT_SAMPLE_RATE_HZ       48000 // default sample rate
     78 // See NBAIO_Format frameworks/av/include/media/nbaio/NBAIO.h.
     79 #define DEFAULT_FORMAT               AUDIO_FORMAT_PCM_16_BIT
     80 // A legacy user of this device does not close the input stream when it shuts down, which
     81 // results in the application opening a new input stream before closing the old input stream
     82 // handle it was previously using.  Setting this value to 1 allows multiple clients to open
     83 // multiple input streams from this device.  If this option is enabled, each input stream returned
     84 // is *the same stream* which means that readers will race to read data from these streams.
     85 #define ENABLE_LEGACY_INPUT_OPEN     1
     86 // Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled.
     87 #define ENABLE_CHANNEL_CONVERSION    1
     88 // Whether resampling is enabled.
     89 #define ENABLE_RESAMPLING            1
     90 #if LOG_STREAMS_TO_FILES
     91 // Folder to save stream log files to.
     92 #define LOG_STREAM_FOLDER "/data/misc/audioserver"
     93 // Log filenames for input and output streams.
     94 #define LOG_STREAM_OUT_FILENAME LOG_STREAM_FOLDER "/r_submix_out.raw"
     95 #define LOG_STREAM_IN_FILENAME LOG_STREAM_FOLDER "/r_submix_in.raw"
     96 // File permissions for stream log files.
     97 #define LOG_STREAM_FILE_PERMISSIONS (S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH)
     98 #endif // LOG_STREAMS_TO_FILES
     99 // limit for number of read error log entries to avoid spamming the logs
    100 #define MAX_READ_ERROR_LOGS 5
    101 
    102 // Common limits macros.
    103 #ifndef min
    104 #define min(a, b) ((a) < (b) ? (a) : (b))
    105 #endif // min
    106 #ifndef max
    107 #define max(a, b) ((a) > (b) ? (a) : (b))
    108 #endif // max
    109 
    110 // Set *result_variable_ptr to true if value_to_find is present in the array array_to_search,
    111 // otherwise set *result_variable_ptr to false.
    112 #define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \
    113     { \
    114         size_t i; \
    115         *(result_variable_ptr) = false; \
    116         for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \
    117           if ((value_to_find) == (array_to_search)[i]) { \
    118                 *(result_variable_ptr) = true; \
    119                 break; \
    120             } \
    121         } \
    122     }
    123 
    124 // Configuration of the submix pipe.
    125 struct submix_config {
    126     // Channel mask field in this data structure is set to either input_channel_mask or
    127     // output_channel_mask depending upon the last stream to be opened on this device.
    128     struct audio_config common;
    129     // Input stream and output stream channel masks.  This is required since input and output
    130     // channel bitfields are not equivalent.
    131     audio_channel_mask_t input_channel_mask;
    132     audio_channel_mask_t output_channel_mask;
    133 #if ENABLE_RESAMPLING
    134     // Input stream and output stream sample rates.
    135     uint32_t input_sample_rate;
    136     uint32_t output_sample_rate;
    137 #endif // ENABLE_RESAMPLING
    138     size_t pipe_frame_size;  // Number of bytes in each audio frame in the pipe.
    139     size_t buffer_size_frames; // Size of the audio pipe in frames.
    140     // Maximum number of frames buffered by the input and output streams.
    141     size_t buffer_period_size_frames;
    142 };
    143 
    144 #define MAX_ROUTES 10
    145 typedef struct route_config {
    146     struct submix_config config;
    147     char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
    148     // Pipe variables: they handle the ring buffer that "pipes" audio:
    149     //  - from the submix virtual audio output == what needs to be played
    150     //    remotely, seen as an output for AudioFlinger
    151     //  - to the virtual audio source == what is captured by the component
    152     //    which "records" the submix / virtual audio source, and handles it as needed.
    153     // A usecase example is one where the component capturing the audio is then sending it over
    154     // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a
    155     // TV with Wifi Display capabilities), or to a wireless audio player.
    156     sp<MonoPipe> rsxSink;
    157     sp<MonoPipeReader> rsxSource;
    158     // Pointers to the current input and output stream instances.  rsxSink and rsxSource are
    159     // destroyed if both and input and output streams are destroyed.
    160     struct submix_stream_out *output;
    161     struct submix_stream_in *input;
    162 #if ENABLE_RESAMPLING
    163     // Buffer used as temporary storage for resampled data prior to returning data to the output
    164     // stream.
    165     int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES];
    166 #endif // ENABLE_RESAMPLING
    167 } route_config_t;
    168 
    169 struct submix_audio_device {
    170     struct audio_hw_device device;
    171     route_config_t routes[MAX_ROUTES];
    172     // Device lock, also used to protect access to submix_audio_device from the input and output
    173     // streams.
    174     pthread_mutex_t lock;
    175 };
    176 
    177 struct submix_stream_out {
    178     struct audio_stream_out stream;
    179     struct submix_audio_device *dev;
    180     int route_handle;
    181     bool output_standby;
    182     uint64_t frames_written;
    183     uint64_t frames_written_since_standby;
    184 #if LOG_STREAMS_TO_FILES
    185     int log_fd;
    186 #endif // LOG_STREAMS_TO_FILES
    187 };
    188 
    189 struct submix_stream_in {
    190     struct audio_stream_in stream;
    191     struct submix_audio_device *dev;
    192     int route_handle;
    193     bool input_standby;
    194     bool output_standby_rec_thr; // output standby state as seen from record thread
    195     // wall clock when recording starts
    196     struct timespec record_start_time;
    197     // how many frames have been requested to be read
    198     uint64_t read_counter_frames;
    199 
    200 #if ENABLE_LEGACY_INPUT_OPEN
    201     // Number of references to this input stream.
    202     volatile int32_t ref_count;
    203 #endif // ENABLE_LEGACY_INPUT_OPEN
    204 #if LOG_STREAMS_TO_FILES
    205     int log_fd;
    206 #endif // LOG_STREAMS_TO_FILES
    207 
    208     volatile int16_t read_error_count;
    209 };
    210 
    211 // Determine whether the specified sample rate is supported by the submix module.
    212 static bool sample_rate_supported(const uint32_t sample_rate)
    213 {
    214     // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp.
    215     static const unsigned int supported_sample_rates[] = {
    216         8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
    217     };
    218     bool return_value;
    219     SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value);
    220     return return_value;
    221 }
    222 
    223 // Determine whether the specified sample rate is supported, if it is return the specified sample
    224 // rate, otherwise return the default sample rate for the submix module.
    225 static uint32_t get_supported_sample_rate(uint32_t sample_rate)
    226 {
    227   return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ;
    228 }
    229 
    230 // Determine whether the specified channel in mask is supported by the submix module.
    231 static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask)
    232 {
    233     // Set of channel in masks supported by Format_from_SR_C()
    234     // frameworks/av/media/libnbaio/NAIO.cpp.
    235     static const audio_channel_mask_t supported_channel_in_masks[] = {
    236         AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO,
    237     };
    238     bool return_value;
    239     SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value);
    240     return return_value;
    241 }
    242 
    243 // Determine whether the specified channel in mask is supported, if it is return the specified
    244 // channel in mask, otherwise return the default channel in mask for the submix module.
    245 static audio_channel_mask_t get_supported_channel_in_mask(
    246         const audio_channel_mask_t channel_in_mask)
    247 {
    248     return channel_in_mask_supported(channel_in_mask) ? channel_in_mask :
    249             static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO);
    250 }
    251 
    252 // Determine whether the specified channel out mask is supported by the submix module.
    253 static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask)
    254 {
    255     // Set of channel out masks supported by Format_from_SR_C()
    256     // frameworks/av/media/libnbaio/NAIO.cpp.
    257     static const audio_channel_mask_t supported_channel_out_masks[] = {
    258         AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO,
    259     };
    260     bool return_value;
    261     SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value);
    262     return return_value;
    263 }
    264 
    265 // Determine whether the specified channel out mask is supported, if it is return the specified
    266 // channel out mask, otherwise return the default channel out mask for the submix module.
    267 static audio_channel_mask_t get_supported_channel_out_mask(
    268         const audio_channel_mask_t channel_out_mask)
    269 {
    270     return channel_out_mask_supported(channel_out_mask) ? channel_out_mask :
    271         static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO);
    272 }
    273 
    274 // Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the
    275 // structure.
    276 static struct submix_stream_out * audio_stream_out_get_submix_stream_out(
    277         struct audio_stream_out * const stream)
    278 {
    279     ALOG_ASSERT(stream);
    280     return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) -
    281                 offsetof(struct submix_stream_out, stream));
    282 }
    283 
    284 // Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure.
    285 static struct submix_stream_out * audio_stream_get_submix_stream_out(
    286         struct audio_stream * const stream)
    287 {
    288     ALOG_ASSERT(stream);
    289     return audio_stream_out_get_submix_stream_out(
    290             reinterpret_cast<struct audio_stream_out *>(stream));
    291 }
    292 
    293 // Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the
    294 // structure.
    295 static struct submix_stream_in * audio_stream_in_get_submix_stream_in(
    296         struct audio_stream_in * const stream)
    297 {
    298     ALOG_ASSERT(stream);
    299     return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) -
    300             offsetof(struct submix_stream_in, stream));
    301 }
    302 
    303 // Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure.
    304 static struct submix_stream_in * audio_stream_get_submix_stream_in(
    305         struct audio_stream * const stream)
    306 {
    307     ALOG_ASSERT(stream);
    308     return audio_stream_in_get_submix_stream_in(
    309             reinterpret_cast<struct audio_stream_in *>(stream));
    310 }
    311 
    312 // Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within
    313 // the structure.
    314 static struct submix_audio_device * audio_hw_device_get_submix_audio_device(
    315         struct audio_hw_device *device)
    316 {
    317     ALOG_ASSERT(device);
    318     return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) -
    319         offsetof(struct submix_audio_device, device));
    320 }
    321 
    322 // Compare an audio_config with input channel mask and an audio_config with output channel mask
    323 // returning false if they do *not* match, true otherwise.
    324 static bool audio_config_compare(const audio_config * const input_config,
    325         const audio_config * const output_config)
    326 {
    327 #if !ENABLE_CHANNEL_CONVERSION
    328     const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask);
    329     const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask);
    330     if (input_channels != output_channels) {
    331         ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d",
    332               input_channels, output_channels);
    333         return false;
    334     }
    335 #endif // !ENABLE_CHANNEL_CONVERSION
    336 #if ENABLE_RESAMPLING
    337     if (input_config->sample_rate != output_config->sample_rate &&
    338             audio_channel_count_from_in_mask(input_config->channel_mask) != 1) {
    339 #else
    340     if (input_config->sample_rate != output_config->sample_rate) {
    341 #endif // ENABLE_RESAMPLING
    342         ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul",
    343               input_config->sample_rate, output_config->sample_rate);
    344         return false;
    345     }
    346     if (input_config->format != output_config->format) {
    347         ALOGE("audio_config_compare() format mismatch %x vs. %x",
    348               input_config->format, output_config->format);
    349         return false;
    350     }
    351     // This purposely ignores offload_info as it's not required for the submix device.
    352     return true;
    353 }
    354 
    355 // If one doesn't exist, create a pipe for the submix audio device rsxadev of size
    356 // buffer_size_frames and optionally associate "in" or "out" with the submix audio device.
    357 // Must be called with lock held on the submix_audio_device
    358 static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev,
    359                                             const struct audio_config * const config,
    360                                             const size_t buffer_size_frames,
    361                                             const uint32_t buffer_period_count,
    362                                             struct submix_stream_in * const in,
    363                                             struct submix_stream_out * const out,
    364                                             const char *address,
    365                                             int route_idx)
    366 {
    367     ALOG_ASSERT(in || out);
    368     ALOG_ASSERT(route_idx > -1);
    369     ALOG_ASSERT(route_idx < MAX_ROUTES);
    370     ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx);
    371 
    372     // Save a reference to the specified input or output stream and the associated channel
    373     // mask.
    374     if (in) {
    375         in->route_handle = route_idx;
    376         rsxadev->routes[route_idx].input = in;
    377         rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask;
    378 #if ENABLE_RESAMPLING
    379         rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate;
    380         // If the output isn't configured yet, set the output sample rate to the maximum supported
    381         // sample rate such that the smallest possible input buffer is created, and put a default
    382         // value for channel count
    383         if (!rsxadev->routes[route_idx].output) {
    384             rsxadev->routes[route_idx].config.output_sample_rate = 48000;
    385             rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO;
    386         }
    387 #endif // ENABLE_RESAMPLING
    388     }
    389     if (out) {
    390         out->route_handle = route_idx;
    391         rsxadev->routes[route_idx].output = out;
    392         rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask;
    393 #if ENABLE_RESAMPLING
    394         rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate;
    395 #endif // ENABLE_RESAMPLING
    396     }
    397     // Save the address
    398     strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN);
    399     ALOGD("  now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx);
    400     // If a pipe isn't associated with the device, create one.
    401     if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL)
    402     {
    403         struct submix_config * const device_config = &rsxadev->routes[route_idx].config;
    404         uint32_t channel_count;
    405         if (out)
    406             channel_count = audio_channel_count_from_out_mask(config->channel_mask);
    407         else
    408             channel_count = audio_channel_count_from_in_mask(config->channel_mask);
    409 #if ENABLE_CHANNEL_CONVERSION
    410         // If channel conversion is enabled, allocate enough space for the maximum number of
    411         // possible channels stored in the pipe for the situation when the number of channels in
    412         // the output stream don't match the number in the input stream.
    413         const uint32_t pipe_channel_count = max(channel_count, 2);
    414 #else
    415         const uint32_t pipe_channel_count = channel_count;
    416 #endif // ENABLE_CHANNEL_CONVERSION
    417         const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count,
    418             config->format);
    419         const NBAIO_Format offers[1] = {format};
    420         size_t numCounterOffers = 0;
    421         // Create a MonoPipe with optional blocking set to true.
    422         MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/);
    423         // Negotiation between the source and sink cannot fail as the device open operation
    424         // creates both ends of the pipe using the same audio format.
    425         ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers);
    426         ALOG_ASSERT(index == 0);
    427         MonoPipeReader* source = new MonoPipeReader(sink);
    428         numCounterOffers = 0;
    429         index = source->negotiate(offers, 1, NULL, numCounterOffers);
    430         ALOG_ASSERT(index == 0);
    431         ALOGV("submix_audio_device_create_pipe_l(): created pipe");
    432 
    433         // Save references to the source and sink.
    434         ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL);
    435         ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL);
    436         rsxadev->routes[route_idx].rsxSink = sink;
    437         rsxadev->routes[route_idx].rsxSource = source;
    438         // Store the sanitized audio format in the device so that it's possible to determine
    439         // the format of the pipe source when opening the input device.
    440         memcpy(&device_config->common, config, sizeof(device_config->common));
    441         device_config->buffer_size_frames = sink->maxFrames();
    442         device_config->buffer_period_size_frames = device_config->buffer_size_frames /
    443                 buffer_period_count;
    444         if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream);
    445         if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream);
    446 #if ENABLE_CHANNEL_CONVERSION
    447         // Calculate the pipe frame size based upon the number of channels.
    448         device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) /
    449                 channel_count;
    450 #endif // ENABLE_CHANNEL_CONVERSION
    451         SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, "
    452                      "period size %zd", device_config->pipe_frame_size,
    453                      device_config->buffer_size_frames, device_config->buffer_period_size_frames);
    454     }
    455 }
    456 
    457 // Release references to the sink and source.  Input and output threads may maintain references
    458 // to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use
    459 // before they shutdown.
    460 // Must be called with lock held on the submix_audio_device
    461 static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev,
    462         int route_idx)
    463 {
    464     ALOG_ASSERT(route_idx > -1);
    465     ALOG_ASSERT(route_idx < MAX_ROUTES);
    466     ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx,
    467             rsxadev->routes[route_idx].address);
    468     if (rsxadev->routes[route_idx].rsxSink != 0) {
    469         rsxadev->routes[route_idx].rsxSink.clear();
    470         rsxadev->routes[route_idx].rsxSink = 0;
    471     }
    472     if (rsxadev->routes[route_idx].rsxSource != 0) {
    473         rsxadev->routes[route_idx].rsxSource.clear();
    474         rsxadev->routes[route_idx].rsxSource = 0;
    475     }
    476     memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN);
    477 #ifdef ENABLE_RESAMPLING
    478     memset(rsxadev->routes[route_idx].resampler_buffer, 0,
    479             sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES);
    480 #endif
    481 }
    482 
    483 // Remove references to the specified input and output streams.  When the device no longer
    484 // references input and output streams destroy the associated pipe.
    485 // Must be called with lock held on the submix_audio_device
    486 static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev,
    487                                              const struct submix_stream_in * const in,
    488                                              const struct submix_stream_out * const out)
    489 {
    490     MonoPipe* sink;
    491     ALOGV("submix_audio_device_destroy_pipe_l()");
    492     int route_idx = -1;
    493     if (in != NULL) {
    494 #if ENABLE_LEGACY_INPUT_OPEN
    495         const_cast<struct submix_stream_in*>(in)->ref_count--;
    496         route_idx = in->route_handle;
    497         ALOG_ASSERT(rsxadev->routes[route_idx].input == in);
    498         if (in->ref_count == 0) {
    499             rsxadev->routes[route_idx].input = NULL;
    500         }
    501         ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count);
    502 #else
    503         rsxadev->input = NULL;
    504 #endif // ENABLE_LEGACY_INPUT_OPEN
    505     }
    506     if (out != NULL) {
    507         route_idx = out->route_handle;
    508         ALOG_ASSERT(rsxadev->routes[route_idx].output == out);
    509         rsxadev->routes[route_idx].output = NULL;
    510     }
    511     if (route_idx != -1 &&
    512             rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) {
    513         submix_audio_device_release_pipe_l(rsxadev, route_idx);
    514         ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed");
    515     }
    516 }
    517 
    518 // Sanitize the user specified audio config for a submix input / output stream.
    519 static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format)
    520 {
    521     config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) :
    522             get_supported_channel_out_mask(config->channel_mask);
    523     config->sample_rate = get_supported_sample_rate(config->sample_rate);
    524     config->format = DEFAULT_FORMAT;
    525 }
    526 
    527 // Verify a submix input or output stream can be opened.
    528 // Must be called with lock held on the submix_audio_device
    529 static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev,
    530                                  int route_idx,
    531                                  const struct audio_config * const config,
    532                                  const bool opening_input)
    533 {
    534     bool input_open;
    535     bool output_open;
    536     audio_config pipe_config;
    537 
    538     // Query the device for the current audio config and whether input and output streams are open.
    539     output_open = rsxadev->routes[route_idx].output != NULL;
    540     input_open = rsxadev->routes[route_idx].input != NULL;
    541     memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config));
    542 
    543     // If the stream is already open, don't open it again.
    544     if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) {
    545         ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" :
    546                 "Output");
    547         return false;
    548     }
    549 
    550     SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x "
    551                  "%s_channel_mask=%x", config->sample_rate, config->format,
    552                  opening_input ? "in" : "out", config->channel_mask);
    553 
    554     // If either stream is open, verify the existing audio config the pipe matches the user
    555     // specified config.
    556     if (input_open || output_open) {
    557         const audio_config * const input_config = opening_input ? config : &pipe_config;
    558         const audio_config * const output_config = opening_input ? &pipe_config : config;
    559         // Get the channel mask of the open device.
    560         pipe_config.channel_mask =
    561             opening_input ? rsxadev->routes[route_idx].config.output_channel_mask :
    562                 rsxadev->routes[route_idx].config.input_channel_mask;
    563         if (!audio_config_compare(input_config, output_config)) {
    564             ALOGE("submix_open_validate_l(): Unsupported format.");
    565             return false;
    566         }
    567     }
    568     return true;
    569 }
    570 
    571 // Must be called with lock held on the submix_audio_device
    572 static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev,
    573                                                  const char* address, /*in*/
    574                                                  int *idx /*out*/)
    575 {
    576     // Do we already have a route for this address
    577     int route_idx = -1;
    578     int route_empty_idx = -1; // index of an empty route slot that can be used if needed
    579     for (int i=0 ; i < MAX_ROUTES ; i++) {
    580         if (strcmp(rsxadev->routes[i].address, "") == 0) {
    581             route_empty_idx = i;
    582         }
    583         if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
    584             route_idx = i;
    585             break;
    586         }
    587     }
    588 
    589     if ((route_idx == -1) && (route_empty_idx == -1)) {
    590         ALOGE("Cannot create new route for address %s, max number of routes reached", address);
    591         return -ENOMEM;
    592     }
    593     if (route_idx == -1) {
    594         route_idx = route_empty_idx;
    595     }
    596     *idx = route_idx;
    597     return OK;
    598 }
    599 
    600 
    601 // Calculate the maximum size of the pipe buffer in frames for the specified stream.
    602 static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream,
    603                                                    const struct submix_config *config,
    604                                                    const size_t pipe_frames,
    605                                                    const size_t stream_frame_size)
    606 {
    607     const size_t pipe_frame_size = config->pipe_frame_size;
    608     const size_t max_frame_size = max(stream_frame_size, pipe_frame_size);
    609     return (pipe_frames * config->pipe_frame_size) / max_frame_size;
    610 }
    611 
    612 /* audio HAL functions */
    613 
    614 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
    615 {
    616     const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
    617             const_cast<struct audio_stream *>(stream));
    618 #if ENABLE_RESAMPLING
    619     const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate;
    620 #else
    621     const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate;
    622 #endif // ENABLE_RESAMPLING
    623     SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s",
    624             out_rate, out->dev->routes[out->route_handle].address);
    625     return out_rate;
    626 }
    627 
    628 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
    629 {
    630     struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
    631 #if ENABLE_RESAMPLING
    632     // The sample rate of the stream can't be changed once it's set since this would change the
    633     // output buffer size and hence break playback to the shared pipe.
    634     if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) {
    635         ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from "
    636               "%u to %u for addr %s",
    637               out->dev->routes[out->route_handle].config.output_sample_rate, rate,
    638               out->dev->routes[out->route_handle].address);
    639         return -ENOSYS;
    640     }
    641 #endif // ENABLE_RESAMPLING
    642     if (!sample_rate_supported(rate)) {
    643         ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
    644         return -ENOSYS;
    645     }
    646     SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate);
    647     out->dev->routes[out->route_handle].config.common.sample_rate = rate;
    648     return 0;
    649 }
    650 
    651 static size_t out_get_buffer_size(const struct audio_stream *stream)
    652 {
    653     const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
    654             const_cast<struct audio_stream *>(stream));
    655     const struct submix_config * const config = &out->dev->routes[out->route_handle].config;
    656     const size_t stream_frame_size =
    657                             audio_stream_out_frame_size((const struct audio_stream_out *)stream);
    658     const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
    659         stream, config, config->buffer_period_size_frames, stream_frame_size);
    660     const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
    661     SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames",
    662                  buffer_size_bytes, buffer_size_frames);
    663     return buffer_size_bytes;
    664 }
    665 
    666 static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
    667 {
    668     const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
    669             const_cast<struct audio_stream *>(stream));
    670     uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask;
    671     SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask);
    672     return channel_mask;
    673 }
    674 
    675 static audio_format_t out_get_format(const struct audio_stream *stream)
    676 {
    677     const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
    678             const_cast<struct audio_stream *>(stream));
    679     const audio_format_t format = out->dev->routes[out->route_handle].config.common.format;
    680     SUBMIX_ALOGV("out_get_format() returns %x", format);
    681     return format;
    682 }
    683 
    684 static int out_set_format(struct audio_stream *stream, audio_format_t format)
    685 {
    686     const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
    687     if (format != out->dev->routes[out->route_handle].config.common.format) {
    688         ALOGE("out_set_format(format=%x) format unsupported", format);
    689         return -ENOSYS;
    690     }
    691     SUBMIX_ALOGV("out_set_format(format=%x)", format);
    692     return 0;
    693 }
    694 
    695 static int out_standby(struct audio_stream *stream)
    696 {
    697     ALOGI("out_standby()");
    698     struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
    699     struct submix_audio_device * const rsxadev = out->dev;
    700 
    701     pthread_mutex_lock(&rsxadev->lock);
    702 
    703     out->output_standby = true;
    704     out->frames_written_since_standby = 0;
    705 
    706     pthread_mutex_unlock(&rsxadev->lock);
    707 
    708     return 0;
    709 }
    710 
    711 static int out_dump(const struct audio_stream *stream, int fd)
    712 {
    713     (void)stream;
    714     (void)fd;
    715     return 0;
    716 }
    717 
    718 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
    719 {
    720     int exiting = -1;
    721     AudioParameter parms = AudioParameter(String8(kvpairs));
    722     SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs);
    723 
    724     // FIXME this is using hard-coded strings but in the future, this functionality will be
    725     //       converted to use audio HAL extensions required to support tunneling
    726     if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) {
    727         struct submix_audio_device * const rsxadev =
    728                 audio_stream_get_submix_stream_out(stream)->dev;
    729         pthread_mutex_lock(&rsxadev->lock);
    730         { // using the sink
    731             sp<MonoPipe> sink =
    732                     rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle]
    733                                     .rsxSink;
    734             if (sink == NULL) {
    735                 pthread_mutex_unlock(&rsxadev->lock);
    736                 return 0;
    737             }
    738 
    739             ALOGD("out_set_parameters(): shutting down MonoPipe sink");
    740             sink->shutdown(true);
    741         } // done using the sink
    742         pthread_mutex_unlock(&rsxadev->lock);
    743     }
    744     return 0;
    745 }
    746 
    747 static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
    748 {
    749     (void)stream;
    750     (void)keys;
    751     return strdup("");
    752 }
    753 
    754 static uint32_t out_get_latency(const struct audio_stream_out *stream)
    755 {
    756     const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(
    757             const_cast<struct audio_stream_out *>(stream));
    758     const struct submix_config * const config = &out->dev->routes[out->route_handle].config;
    759     const size_t stream_frame_size =
    760                             audio_stream_out_frame_size(stream);
    761     const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
    762             &stream->common, config, config->buffer_size_frames, stream_frame_size);
    763     const uint32_t sample_rate = out_get_sample_rate(&stream->common);
    764     const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate;
    765     SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u",
    766                  latency_ms, buffer_size_frames, sample_rate);
    767     return latency_ms;
    768 }
    769 
    770 static int out_set_volume(struct audio_stream_out *stream, float left,
    771                           float right)
    772 {
    773     (void)stream;
    774     (void)left;
    775     (void)right;
    776     return -ENOSYS;
    777 }
    778 
    779 static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
    780                          size_t bytes)
    781 {
    782     SUBMIX_ALOGV("out_write(bytes=%zd)", bytes);
    783     ssize_t written_frames = 0;
    784     const size_t frame_size = audio_stream_out_frame_size(stream);
    785     struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream);
    786     struct submix_audio_device * const rsxadev = out->dev;
    787     const size_t frames = bytes / frame_size;
    788 
    789     pthread_mutex_lock(&rsxadev->lock);
    790 
    791     out->output_standby = false;
    792 
    793     sp<MonoPipe> sink = rsxadev->routes[out->route_handle].rsxSink;
    794     if (sink != NULL) {
    795         if (sink->isShutdown()) {
    796             sink.clear();
    797             pthread_mutex_unlock(&rsxadev->lock);
    798             SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write.");
    799             // the pipe has already been shutdown, this buffer will be lost but we must
    800             //   simulate timing so we don't drain the output faster than realtime
    801             usleep(frames * 1000000 / out_get_sample_rate(&stream->common));
    802             return bytes;
    803         }
    804     } else {
    805         pthread_mutex_unlock(&rsxadev->lock);
    806         ALOGE("out_write without a pipe!");
    807         ALOG_ASSERT("out_write without a pipe!");
    808         return 0;
    809     }
    810 
    811     // If the write to the sink would block when no input stream is present, flush enough frames
    812     // from the pipe to make space to write the most recent data.
    813     {
    814         const size_t availableToWrite = sink->availableToWrite();
    815         sp<MonoPipeReader> source = rsxadev->routes[out->route_handle].rsxSource;
    816         if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) {
    817             static uint8_t flush_buffer[64];
    818             const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size;
    819             size_t frames_to_flush_from_source = frames - availableToWrite;
    820             SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking",
    821                          frames_to_flush_from_source);
    822             while (frames_to_flush_from_source) {
    823                 const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames);
    824                 frames_to_flush_from_source -= flush_size;
    825                 // read does not block
    826                 source->read(flush_buffer, flush_size);
    827             }
    828         }
    829     }
    830 
    831     pthread_mutex_unlock(&rsxadev->lock);
    832 
    833     written_frames = sink->write(buffer, frames);
    834 
    835 #if LOG_STREAMS_TO_FILES
    836     if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size);
    837 #endif // LOG_STREAMS_TO_FILES
    838 
    839     if (written_frames < 0) {
    840         if (written_frames == (ssize_t)NEGOTIATE) {
    841             ALOGE("out_write() write to pipe returned NEGOTIATE");
    842 
    843             pthread_mutex_lock(&rsxadev->lock);
    844             sink.clear();
    845             pthread_mutex_unlock(&rsxadev->lock);
    846 
    847             written_frames = 0;
    848             return 0;
    849         } else {
    850             // write() returned UNDERRUN or WOULD_BLOCK, retry
    851             ALOGE("out_write() write to pipe returned unexpected %zd", written_frames);
    852             written_frames = sink->write(buffer, frames);
    853         }
    854     }
    855 
    856     pthread_mutex_lock(&rsxadev->lock);
    857     sink.clear();
    858     if (written_frames > 0) {
    859         out->frames_written_since_standby += written_frames;
    860         out->frames_written += written_frames;
    861     }
    862     pthread_mutex_unlock(&rsxadev->lock);
    863 
    864     if (written_frames < 0) {
    865         ALOGE("out_write() failed writing to pipe with %zd", written_frames);
    866         return 0;
    867     }
    868     const ssize_t written_bytes = written_frames * frame_size;
    869     SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames);
    870     return written_bytes;
    871 }
    872 
    873 static int out_get_presentation_position(const struct audio_stream_out *stream,
    874                                    uint64_t *frames, struct timespec *timestamp)
    875 {
    876     if (stream == NULL || frames == NULL || timestamp == NULL) {
    877         return -EINVAL;
    878     }
    879 
    880     const submix_stream_out *out = audio_stream_out_get_submix_stream_out(
    881             const_cast<struct audio_stream_out *>(stream));
    882     struct submix_audio_device * const rsxadev = out->dev;
    883 
    884     int ret = -EWOULDBLOCK;
    885     pthread_mutex_lock(&rsxadev->lock);
    886     const ssize_t frames_in_pipe =
    887             rsxadev->routes[out->route_handle].rsxSource->availableToRead();
    888     if (CC_UNLIKELY(frames_in_pipe < 0)) {
    889         *frames = out->frames_written;
    890         ret = 0;
    891     } else if (out->frames_written >= (uint64_t)frames_in_pipe) {
    892         *frames = out->frames_written - frames_in_pipe;
    893         ret = 0;
    894     }
    895     pthread_mutex_unlock(&rsxadev->lock);
    896 
    897     if (ret == 0) {
    898         clock_gettime(CLOCK_MONOTONIC, timestamp);
    899     }
    900 
    901     SUBMIX_ALOGV("out_get_presentation_position() got frames=%llu timestamp sec=%llu",
    902             frames ? *frames : -1, timestamp ? timestamp->tv_sec : -1);
    903 
    904     return ret;
    905 }
    906 
    907 static int out_get_render_position(const struct audio_stream_out *stream,
    908                                    uint32_t *dsp_frames)
    909 {
    910     if (stream == NULL || dsp_frames == NULL) {
    911         return -EINVAL;
    912     }
    913 
    914     const submix_stream_out *out = audio_stream_out_get_submix_stream_out(
    915             const_cast<struct audio_stream_out *>(stream));
    916     struct submix_audio_device * const rsxadev = out->dev;
    917 
    918     pthread_mutex_lock(&rsxadev->lock);
    919     const ssize_t frames_in_pipe =
    920             rsxadev->routes[out->route_handle].rsxSource->availableToRead();
    921     if (CC_UNLIKELY(frames_in_pipe < 0)) {
    922         *dsp_frames = (uint32_t)out->frames_written_since_standby;
    923     } else {
    924         *dsp_frames = out->frames_written_since_standby > (uint64_t) frames_in_pipe ?
    925                 (uint32_t)(out->frames_written_since_standby - frames_in_pipe) : 0;
    926     }
    927     pthread_mutex_unlock(&rsxadev->lock);
    928 
    929     return 0;
    930 }
    931 
    932 static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
    933 {
    934     (void)stream;
    935     (void)effect;
    936     return 0;
    937 }
    938 
    939 static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
    940 {
    941     (void)stream;
    942     (void)effect;
    943     return 0;
    944 }
    945 
    946 static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
    947                                         int64_t *timestamp)
    948 {
    949     (void)stream;
    950     (void)timestamp;
    951     return -EINVAL;
    952 }
    953 
    954 /** audio_stream_in implementation **/
    955 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
    956 {
    957     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
    958         const_cast<struct audio_stream*>(stream));
    959 #if ENABLE_RESAMPLING
    960     const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate;
    961 #else
    962     const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate;
    963 #endif // ENABLE_RESAMPLING
    964     SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate);
    965     return rate;
    966 }
    967 
    968 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
    969 {
    970     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
    971 #if ENABLE_RESAMPLING
    972     // The sample rate of the stream can't be changed once it's set since this would change the
    973     // input buffer size and hence break recording from the shared pipe.
    974     if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) {
    975         ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from "
    976               "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate);
    977         return -ENOSYS;
    978     }
    979 #endif // ENABLE_RESAMPLING
    980     if (!sample_rate_supported(rate)) {
    981         ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate);
    982         return -ENOSYS;
    983     }
    984     in->dev->routes[in->route_handle].config.common.sample_rate = rate;
    985     SUBMIX_ALOGV("in_set_sample_rate() set %u", rate);
    986     return 0;
    987 }
    988 
    989 static size_t in_get_buffer_size(const struct audio_stream *stream)
    990 {
    991     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
    992             const_cast<struct audio_stream*>(stream));
    993     const struct submix_config * const config = &in->dev->routes[in->route_handle].config;
    994     const size_t stream_frame_size =
    995                             audio_stream_in_frame_size((const struct audio_stream_in *)stream);
    996     size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
    997         stream, config, config->buffer_period_size_frames, stream_frame_size);
    998 #if ENABLE_RESAMPLING
    999     // Scale the size of the buffer based upon the maximum number of frames that could be returned
   1000     // given the ratio of output to input sample rate.
   1001     buffer_size_frames = (size_t)(((float)buffer_size_frames *
   1002                                    (float)config->input_sample_rate) /
   1003                                   (float)config->output_sample_rate);
   1004 #endif // ENABLE_RESAMPLING
   1005     const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
   1006     SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes,
   1007                  buffer_size_frames);
   1008     return buffer_size_bytes;
   1009 }
   1010 
   1011 static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
   1012 {
   1013     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
   1014             const_cast<struct audio_stream*>(stream));
   1015     const audio_channel_mask_t channel_mask =
   1016             in->dev->routes[in->route_handle].config.input_channel_mask;
   1017     SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask);
   1018     return channel_mask;
   1019 }
   1020 
   1021 static audio_format_t in_get_format(const struct audio_stream *stream)
   1022 {
   1023     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
   1024             const_cast<struct audio_stream*>(stream));
   1025     const audio_format_t format = in->dev->routes[in->route_handle].config.common.format;
   1026     SUBMIX_ALOGV("in_get_format() returns %x", format);
   1027     return format;
   1028 }
   1029 
   1030 static int in_set_format(struct audio_stream *stream, audio_format_t format)
   1031 {
   1032     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
   1033     if (format != in->dev->routes[in->route_handle].config.common.format) {
   1034         ALOGE("in_set_format(format=%x) format unsupported", format);
   1035         return -ENOSYS;
   1036     }
   1037     SUBMIX_ALOGV("in_set_format(format=%x)", format);
   1038     return 0;
   1039 }
   1040 
   1041 static int in_standby(struct audio_stream *stream)
   1042 {
   1043     ALOGI("in_standby()");
   1044     struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
   1045     struct submix_audio_device * const rsxadev = in->dev;
   1046 
   1047     pthread_mutex_lock(&rsxadev->lock);
   1048 
   1049     in->input_standby = true;
   1050 
   1051     pthread_mutex_unlock(&rsxadev->lock);
   1052 
   1053     return 0;
   1054 }
   1055 
   1056 static int in_dump(const struct audio_stream *stream, int fd)
   1057 {
   1058     (void)stream;
   1059     (void)fd;
   1060     return 0;
   1061 }
   1062 
   1063 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
   1064 {
   1065     (void)stream;
   1066     (void)kvpairs;
   1067     return 0;
   1068 }
   1069 
   1070 static char * in_get_parameters(const struct audio_stream *stream,
   1071                                 const char *keys)
   1072 {
   1073     (void)stream;
   1074     (void)keys;
   1075     return strdup("");
   1076 }
   1077 
   1078 static int in_set_gain(struct audio_stream_in *stream, float gain)
   1079 {
   1080     (void)stream;
   1081     (void)gain;
   1082     return 0;
   1083 }
   1084 
   1085 static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
   1086                        size_t bytes)
   1087 {
   1088     struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream);
   1089     struct submix_audio_device * const rsxadev = in->dev;
   1090     struct audio_config *format;
   1091     const size_t frame_size = audio_stream_in_frame_size(stream);
   1092     const size_t frames_to_read = bytes / frame_size;
   1093 
   1094     SUBMIX_ALOGV("in_read bytes=%zu", bytes);
   1095     pthread_mutex_lock(&rsxadev->lock);
   1096 
   1097     const bool output_standby = rsxadev->routes[in->route_handle].output == NULL
   1098             ? true : rsxadev->routes[in->route_handle].output->output_standby;
   1099     const bool output_standby_transition = (in->output_standby_rec_thr != output_standby);
   1100     in->output_standby_rec_thr = output_standby;
   1101 
   1102     if (in->input_standby || output_standby_transition) {
   1103         in->input_standby = false;
   1104         // keep track of when we exit input standby (== first read == start "real recording")
   1105         // or when we start recording silence, and reset projected time
   1106         int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time);
   1107         if (rc == 0) {
   1108             in->read_counter_frames = 0;
   1109         }
   1110     }
   1111 
   1112     in->read_counter_frames += frames_to_read;
   1113     size_t remaining_frames = frames_to_read;
   1114 
   1115     {
   1116         // about to read from audio source
   1117         sp<MonoPipeReader> source = rsxadev->routes[in->route_handle].rsxSource;
   1118         if (source == NULL) {
   1119             in->read_error_count++;// ok if it rolls over
   1120             ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS,
   1121                     "no audio pipe yet we're trying to read! (not all errors will be logged)");
   1122             pthread_mutex_unlock(&rsxadev->lock);
   1123             usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common));
   1124             memset(buffer, 0, bytes);
   1125             return bytes;
   1126         }
   1127 
   1128         pthread_mutex_unlock(&rsxadev->lock);
   1129 
   1130         // read the data from the pipe (it's non blocking)
   1131         int attempts = 0;
   1132         char* buff = (char*)buffer;
   1133 #if ENABLE_CHANNEL_CONVERSION
   1134         // Determine whether channel conversion is required.
   1135         const uint32_t input_channels = audio_channel_count_from_in_mask(
   1136             rsxadev->routes[in->route_handle].config.input_channel_mask);
   1137         const uint32_t output_channels = audio_channel_count_from_out_mask(
   1138             rsxadev->routes[in->route_handle].config.output_channel_mask);
   1139         if (input_channels != output_channels) {
   1140             SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d "
   1141                          "input channels", output_channels, input_channels);
   1142             // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono.
   1143             ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
   1144                     AUDIO_FORMAT_PCM_16_BIT);
   1145             ALOG_ASSERT((input_channels == 1 && output_channels == 2) ||
   1146                         (input_channels == 2 && output_channels == 1));
   1147         }
   1148 #endif // ENABLE_CHANNEL_CONVERSION
   1149 
   1150 #if ENABLE_RESAMPLING
   1151         const uint32_t input_sample_rate = in_get_sample_rate(&stream->common);
   1152         const uint32_t output_sample_rate =
   1153                 rsxadev->routes[in->route_handle].config.output_sample_rate;
   1154         const size_t resampler_buffer_size_frames =
   1155             sizeof(rsxadev->routes[in->route_handle].resampler_buffer) /
   1156                 sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]);
   1157         float resampler_ratio = 1.0f;
   1158         // Determine whether resampling is required.
   1159         if (input_sample_rate != output_sample_rate) {
   1160             resampler_ratio = (float)output_sample_rate / (float)input_sample_rate;
   1161             // Only support 16-bit PCM mono resampling.
   1162             // NOTE: Resampling is performed after the channel conversion step.
   1163             ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
   1164                     AUDIO_FORMAT_PCM_16_BIT);
   1165             ALOG_ASSERT(audio_channel_count_from_in_mask(
   1166                     rsxadev->routes[in->route_handle].config.input_channel_mask) == 1);
   1167         }
   1168 #endif // ENABLE_RESAMPLING
   1169 
   1170         while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
   1171             ssize_t frames_read = -1977;
   1172             size_t read_frames = remaining_frames;
   1173 #if ENABLE_RESAMPLING
   1174             char* const saved_buff = buff;
   1175             if (resampler_ratio != 1.0f) {
   1176                 // Calculate the number of frames from the pipe that need to be read to generate
   1177                 // the data for the input stream read.
   1178                 const size_t frames_required_for_resampler = (size_t)(
   1179                     (float)read_frames * (float)resampler_ratio);
   1180                 read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames);
   1181                 // Read into the resampler buffer.
   1182                 buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer;
   1183             }
   1184 #endif // ENABLE_RESAMPLING
   1185 #if ENABLE_CHANNEL_CONVERSION
   1186             if (output_channels == 1 && input_channels == 2) {
   1187                 // Need to read half the requested frames since the converted output
   1188                 // data will take twice the space (mono->stereo).
   1189                 read_frames /= 2;
   1190             }
   1191 #endif // ENABLE_CHANNEL_CONVERSION
   1192 
   1193             SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead());
   1194 
   1195             frames_read = source->read(buff, read_frames);
   1196 
   1197             SUBMIX_ALOGV("in_read(): frames read %zd", frames_read);
   1198 
   1199 #if ENABLE_CHANNEL_CONVERSION
   1200             // Perform in-place channel conversion.
   1201             // NOTE: In the following "input stream" refers to the data returned by this function
   1202             // and "output stream" refers to the data read from the pipe.
   1203             if (input_channels != output_channels && frames_read > 0) {
   1204                 int16_t *data = (int16_t*)buff;
   1205                 if (output_channels == 2 && input_channels == 1) {
   1206                     // Offset into the output stream data in samples.
   1207                     ssize_t output_stream_offset = 0;
   1208                     for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read;
   1209                          input_stream_frame++, output_stream_offset += 2) {
   1210                         // Average the content from both channels.
   1211                         data[input_stream_frame] = ((int32_t)data[output_stream_offset] +
   1212                                                     (int32_t)data[output_stream_offset + 1]) / 2;
   1213                     }
   1214                 } else if (output_channels == 1 && input_channels == 2) {
   1215                     // Offset into the input stream data in samples.
   1216                     ssize_t input_stream_offset = (frames_read - 1) * 2;
   1217                     for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0;
   1218                          output_stream_frame--, input_stream_offset -= 2) {
   1219                         const short sample = data[output_stream_frame];
   1220                         data[input_stream_offset] = sample;
   1221                         data[input_stream_offset + 1] = sample;
   1222                     }
   1223                 }
   1224             }
   1225 #endif // ENABLE_CHANNEL_CONVERSION
   1226 
   1227 #if ENABLE_RESAMPLING
   1228             if (resampler_ratio != 1.0f) {
   1229                 SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read);
   1230                 const int16_t * const data = (int16_t*)buff;
   1231                 int16_t * const resampled_buffer = (int16_t*)saved_buff;
   1232                 // Resample with *no* filtering - if the data from the ouptut stream was really
   1233                 // sampled at a different rate this will result in very nasty aliasing.
   1234                 const float output_stream_frames = (float)frames_read;
   1235                 size_t input_stream_frame = 0;
   1236                 for (float output_stream_frame = 0.0f;
   1237                      output_stream_frame < output_stream_frames &&
   1238                      input_stream_frame < remaining_frames;
   1239                      output_stream_frame += resampler_ratio, input_stream_frame++) {
   1240                     resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame];
   1241                 }
   1242                 ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames);
   1243                 SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame);
   1244                 frames_read = input_stream_frame;
   1245                 buff = saved_buff;
   1246             }
   1247 #endif // ENABLE_RESAMPLING
   1248 
   1249             if (frames_read > 0) {
   1250 #if LOG_STREAMS_TO_FILES
   1251                 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size);
   1252 #endif // LOG_STREAMS_TO_FILES
   1253 
   1254                 remaining_frames -= frames_read;
   1255                 buff += frames_read * frame_size;
   1256                 SUBMIX_ALOGV("  in_read (att=%d) got %zd frames, remaining=%zu",
   1257                              attempts, frames_read, remaining_frames);
   1258             } else {
   1259                 attempts++;
   1260                 SUBMIX_ALOGE("  in_read read returned %zd", frames_read);
   1261                 usleep(READ_ATTEMPT_SLEEP_MS * 1000);
   1262             }
   1263         }
   1264         // done using the source
   1265         pthread_mutex_lock(&rsxadev->lock);
   1266         source.clear();
   1267         pthread_mutex_unlock(&rsxadev->lock);
   1268     }
   1269 
   1270     if (remaining_frames > 0) {
   1271         const size_t remaining_bytes = remaining_frames * frame_size;
   1272         SUBMIX_ALOGV("  clearing remaining_frames = %zu", remaining_frames);
   1273         memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes);
   1274     }
   1275 
   1276     // compute how much we need to sleep after reading the data by comparing the wall clock with
   1277     //   the projected time at which we should return.
   1278     struct timespec time_after_read;// wall clock after reading from the pipe
   1279     struct timespec record_duration;// observed record duration
   1280     int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read);
   1281     const uint32_t sample_rate = in_get_sample_rate(&stream->common);
   1282     if (rc == 0) {
   1283         // for how long have we been recording?
   1284         record_duration.tv_sec  = time_after_read.tv_sec - in->record_start_time.tv_sec;
   1285         record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec;
   1286         if (record_duration.tv_nsec < 0) {
   1287             record_duration.tv_sec--;
   1288             record_duration.tv_nsec += 1000000000;
   1289         }
   1290 
   1291         // read_counter_frames contains the number of frames that have been read since the
   1292         // beginning of recording (including this call): it's converted to usec and compared to
   1293         // how long we've been recording for, which gives us how long we must wait to sync the
   1294         // projected recording time, and the observed recording time.
   1295         long projected_vs_observed_offset_us =
   1296                 ((int64_t)(in->read_counter_frames
   1297                             - (record_duration.tv_sec*sample_rate)))
   1298                         * 1000000 / sample_rate
   1299                 - (record_duration.tv_nsec / 1000);
   1300 
   1301         SUBMIX_ALOGV("  record duration %5lds %3ldms, will wait: %7ldus",
   1302                 record_duration.tv_sec, record_duration.tv_nsec/1000000,
   1303                 projected_vs_observed_offset_us);
   1304         if (projected_vs_observed_offset_us > 0) {
   1305             usleep(projected_vs_observed_offset_us);
   1306         }
   1307     }
   1308 
   1309     SUBMIX_ALOGV("in_read returns %zu", bytes);
   1310     return bytes;
   1311 
   1312 }
   1313 
   1314 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
   1315 {
   1316     (void)stream;
   1317     return 0;
   1318 }
   1319 
   1320 static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
   1321 {
   1322     (void)stream;
   1323     (void)effect;
   1324     return 0;
   1325 }
   1326 
   1327 static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
   1328 {
   1329     (void)stream;
   1330     (void)effect;
   1331     return 0;
   1332 }
   1333 
   1334 static int adev_open_output_stream(struct audio_hw_device *dev,
   1335                                    audio_io_handle_t handle,
   1336                                    audio_devices_t devices,
   1337                                    audio_output_flags_t flags,
   1338                                    struct audio_config *config,
   1339                                    struct audio_stream_out **stream_out,
   1340                                    const char *address)
   1341 {
   1342     struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev);
   1343     ALOGD("adev_open_output_stream(address=%s)", address);
   1344     struct submix_stream_out *out;
   1345     bool force_pipe_creation = false;
   1346     (void)handle;
   1347     (void)devices;
   1348     (void)flags;
   1349 
   1350     *stream_out = NULL;
   1351 
   1352     // Make sure it's possible to open the device given the current audio config.
   1353     submix_sanitize_config(config, false);
   1354 
   1355     int route_idx = -1;
   1356 
   1357     pthread_mutex_lock(&rsxadev->lock);
   1358 
   1359     status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx);
   1360     if (res != OK) {
   1361         ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address);
   1362         pthread_mutex_unlock(&rsxadev->lock);
   1363         return res;
   1364     }
   1365 
   1366     if (!submix_open_validate_l(rsxadev, route_idx, config, false)) {
   1367         ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address);
   1368         pthread_mutex_unlock(&rsxadev->lock);
   1369         return -EINVAL;
   1370     }
   1371 
   1372     out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out));
   1373     if (!out) {
   1374         pthread_mutex_unlock(&rsxadev->lock);
   1375         return -ENOMEM;
   1376     }
   1377 
   1378     // Initialize the function pointer tables (v-tables).
   1379     out->stream.common.get_sample_rate = out_get_sample_rate;
   1380     out->stream.common.set_sample_rate = out_set_sample_rate;
   1381     out->stream.common.get_buffer_size = out_get_buffer_size;
   1382     out->stream.common.get_channels = out_get_channels;
   1383     out->stream.common.get_format = out_get_format;
   1384     out->stream.common.set_format = out_set_format;
   1385     out->stream.common.standby = out_standby;
   1386     out->stream.common.dump = out_dump;
   1387     out->stream.common.set_parameters = out_set_parameters;
   1388     out->stream.common.get_parameters = out_get_parameters;
   1389     out->stream.common.add_audio_effect = out_add_audio_effect;
   1390     out->stream.common.remove_audio_effect = out_remove_audio_effect;
   1391     out->stream.get_latency = out_get_latency;
   1392     out->stream.set_volume = out_set_volume;
   1393     out->stream.write = out_write;
   1394     out->stream.get_render_position = out_get_render_position;
   1395     out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
   1396     out->stream.get_presentation_position = out_get_presentation_position;
   1397 
   1398 #if ENABLE_RESAMPLING
   1399     // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits
   1400     // writes correctly.
   1401     force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate
   1402             != config->sample_rate;
   1403 #endif // ENABLE_RESAMPLING
   1404 
   1405     // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so
   1406     // that it's recreated.
   1407     if ((rsxadev->routes[route_idx].rsxSink != NULL
   1408             && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) {
   1409         submix_audio_device_release_pipe_l(rsxadev, route_idx);
   1410     }
   1411 
   1412     // Store a pointer to the device from the output stream.
   1413     out->dev = rsxadev;
   1414     // Initialize the pipe.
   1415     ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx);
   1416     submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES,
   1417             DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx);
   1418 #if LOG_STREAMS_TO_FILES
   1419     out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY,
   1420                        LOG_STREAM_FILE_PERMISSIONS);
   1421     ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s",
   1422              strerror(errno));
   1423     ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd);
   1424 #endif // LOG_STREAMS_TO_FILES
   1425     // Return the output stream.
   1426     *stream_out = &out->stream;
   1427 
   1428     pthread_mutex_unlock(&rsxadev->lock);
   1429     return 0;
   1430 }
   1431 
   1432 static void adev_close_output_stream(struct audio_hw_device *dev,
   1433                                      struct audio_stream_out *stream)
   1434 {
   1435     struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device(
   1436                     const_cast<struct audio_hw_device*>(dev));
   1437     struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream);
   1438 
   1439     pthread_mutex_lock(&rsxadev->lock);
   1440     ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address);
   1441     submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out);
   1442 #if LOG_STREAMS_TO_FILES
   1443     if (out->log_fd >= 0) close(out->log_fd);
   1444 #endif // LOG_STREAMS_TO_FILES
   1445 
   1446     pthread_mutex_unlock(&rsxadev->lock);
   1447     free(out);
   1448 }
   1449 
   1450 static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
   1451 {
   1452     (void)dev;
   1453     (void)kvpairs;
   1454     return -ENOSYS;
   1455 }
   1456 
   1457 static char * adev_get_parameters(const struct audio_hw_device *dev,
   1458                                   const char *keys)
   1459 {
   1460     (void)dev;
   1461     (void)keys;
   1462     return strdup("");;
   1463 }
   1464 
   1465 static int adev_init_check(const struct audio_hw_device *dev)
   1466 {
   1467     ALOGI("adev_init_check()");
   1468     (void)dev;
   1469     return 0;
   1470 }
   1471 
   1472 static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
   1473 {
   1474     (void)dev;
   1475     (void)volume;
   1476     return -ENOSYS;
   1477 }
   1478 
   1479 static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
   1480 {
   1481     (void)dev;
   1482     (void)volume;
   1483     return -ENOSYS;
   1484 }
   1485 
   1486 static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
   1487 {
   1488     (void)dev;
   1489     (void)volume;
   1490     return -ENOSYS;
   1491 }
   1492 
   1493 static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
   1494 {
   1495     (void)dev;
   1496     (void)muted;
   1497     return -ENOSYS;
   1498 }
   1499 
   1500 static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
   1501 {
   1502     (void)dev;
   1503     (void)muted;
   1504     return -ENOSYS;
   1505 }
   1506 
   1507 static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
   1508 {
   1509     (void)dev;
   1510     (void)mode;
   1511     return 0;
   1512 }
   1513 
   1514 static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
   1515 {
   1516     (void)dev;
   1517     (void)state;
   1518     return -ENOSYS;
   1519 }
   1520 
   1521 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
   1522 {
   1523     (void)dev;
   1524     (void)state;
   1525     return -ENOSYS;
   1526 }
   1527 
   1528 static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
   1529                                          const struct audio_config *config)
   1530 {
   1531     if (audio_is_linear_pcm(config->format)) {
   1532         size_t max_buffer_period_size_frames = 0;
   1533         struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device(
   1534                 const_cast<struct audio_hw_device*>(dev));
   1535         // look for the largest buffer period size
   1536         for (int i = 0 ; i < MAX_ROUTES ; i++) {
   1537             if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames)
   1538             {
   1539                 max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames;
   1540             }
   1541         }
   1542         const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) *
   1543                 audio_bytes_per_sample(config->format);
   1544         const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes;
   1545         SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames",
   1546                  buffer_size, buffer_period_size_frames);
   1547         return buffer_size;
   1548     }
   1549     return 0;
   1550 }
   1551 
   1552 static int adev_open_input_stream(struct audio_hw_device *dev,
   1553                                   audio_io_handle_t handle,
   1554                                   audio_devices_t devices,
   1555                                   struct audio_config *config,
   1556                                   struct audio_stream_in **stream_in,
   1557                                   audio_input_flags_t flags __unused,
   1558                                   const char *address,
   1559                                   audio_source_t source __unused)
   1560 {
   1561     struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev);
   1562     struct submix_stream_in *in;
   1563     ALOGD("adev_open_input_stream(addr=%s)", address);
   1564     (void)handle;
   1565     (void)devices;
   1566 
   1567     *stream_in = NULL;
   1568 
   1569     // Do we already have a route for this address
   1570     int route_idx = -1;
   1571 
   1572     pthread_mutex_lock(&rsxadev->lock);
   1573 
   1574     status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx);
   1575     if (res != OK) {
   1576         ALOGE("Error %d looking for address=%s in adev_open_input_stream", res, address);
   1577         pthread_mutex_unlock(&rsxadev->lock);
   1578         return res;
   1579     }
   1580 
   1581     // Make sure it's possible to open the device given the current audio config.
   1582     submix_sanitize_config(config, true);
   1583     if (!submix_open_validate_l(rsxadev, route_idx, config, true)) {
   1584         ALOGE("adev_open_input_stream(): Unable to open input stream.");
   1585         pthread_mutex_unlock(&rsxadev->lock);
   1586         return -EINVAL;
   1587     }
   1588 
   1589 #if ENABLE_LEGACY_INPUT_OPEN
   1590     in = rsxadev->routes[route_idx].input;
   1591     if (in) {
   1592         in->ref_count++;
   1593         sp<MonoPipe> sink = rsxadev->routes[route_idx].rsxSink;
   1594         ALOG_ASSERT(sink != NULL);
   1595         // If the sink has been shutdown, delete the pipe.
   1596         if (sink != NULL) {
   1597             if (sink->isShutdown()) {
   1598                 ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d",
   1599                         in->ref_count);
   1600                 submix_audio_device_release_pipe_l(rsxadev, in->route_handle);
   1601             } else {
   1602                 ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count);
   1603             }
   1604         } else {
   1605             ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count);
   1606         }
   1607     }
   1608 #else
   1609     in = NULL;
   1610 #endif // ENABLE_LEGACY_INPUT_OPEN
   1611 
   1612     if (!in) {
   1613         in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in));
   1614         if (!in) return -ENOMEM;
   1615         in->ref_count = 1;
   1616 
   1617         // Initialize the function pointer tables (v-tables).
   1618         in->stream.common.get_sample_rate = in_get_sample_rate;
   1619         in->stream.common.set_sample_rate = in_set_sample_rate;
   1620         in->stream.common.get_buffer_size = in_get_buffer_size;
   1621         in->stream.common.get_channels = in_get_channels;
   1622         in->stream.common.get_format = in_get_format;
   1623         in->stream.common.set_format = in_set_format;
   1624         in->stream.common.standby = in_standby;
   1625         in->stream.common.dump = in_dump;
   1626         in->stream.common.set_parameters = in_set_parameters;
   1627         in->stream.common.get_parameters = in_get_parameters;
   1628         in->stream.common.add_audio_effect = in_add_audio_effect;
   1629         in->stream.common.remove_audio_effect = in_remove_audio_effect;
   1630         in->stream.set_gain = in_set_gain;
   1631         in->stream.read = in_read;
   1632         in->stream.get_input_frames_lost = in_get_input_frames_lost;
   1633 
   1634         in->dev = rsxadev;
   1635 #if LOG_STREAMS_TO_FILES
   1636         in->log_fd = -1;
   1637 #endif
   1638     }
   1639 
   1640     // Initialize the input stream.
   1641     in->read_counter_frames = 0;
   1642     in->input_standby = true;
   1643     if (rsxadev->routes[route_idx].output != NULL) {
   1644         in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby;
   1645     } else {
   1646         in->output_standby_rec_thr = true;
   1647     }
   1648 
   1649     in->read_error_count = 0;
   1650     // Initialize the pipe.
   1651     ALOGV("adev_open_input_stream(): about to create pipe");
   1652     submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES,
   1653                                     DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx);
   1654 #if LOG_STREAMS_TO_FILES
   1655     if (in->log_fd >= 0) close(in->log_fd);
   1656     in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY,
   1657                       LOG_STREAM_FILE_PERMISSIONS);
   1658     ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s",
   1659              strerror(errno));
   1660     ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd);
   1661 #endif // LOG_STREAMS_TO_FILES
   1662     // Return the input stream.
   1663     *stream_in = &in->stream;
   1664 
   1665     pthread_mutex_unlock(&rsxadev->lock);
   1666     return 0;
   1667 }
   1668 
   1669 static void adev_close_input_stream(struct audio_hw_device *dev,
   1670                                     struct audio_stream_in *stream)
   1671 {
   1672     struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev);
   1673 
   1674     struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream);
   1675     ALOGD("adev_close_input_stream()");
   1676     pthread_mutex_lock(&rsxadev->lock);
   1677     submix_audio_device_destroy_pipe_l(rsxadev, in, NULL);
   1678 #if LOG_STREAMS_TO_FILES
   1679     if (in->log_fd >= 0) close(in->log_fd);
   1680 #endif // LOG_STREAMS_TO_FILES
   1681 #if ENABLE_LEGACY_INPUT_OPEN
   1682     if (in->ref_count == 0) free(in);
   1683 #else
   1684     free(in);
   1685 #endif // ENABLE_LEGACY_INPUT_OPEN
   1686 
   1687     pthread_mutex_unlock(&rsxadev->lock);
   1688 }
   1689 
   1690 static int adev_dump(const audio_hw_device_t *device, int fd)
   1691 {
   1692     const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device);
   1693             reinterpret_cast<const struct submix_audio_device *>(
   1694                     reinterpret_cast<const uint8_t *>(device) -
   1695                             offsetof(struct submix_audio_device, device));
   1696     char msg[100];
   1697     int n = sprintf(msg, "\nReroute submix audio module:\n");
   1698     write(fd, &msg, n);
   1699     for (int i=0 ; i < MAX_ROUTES ; i++) {
   1700         n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i,
   1701                 rsxadev->routes[i].config.input_sample_rate,
   1702                 rsxadev->routes[i].config.output_sample_rate,
   1703                 rsxadev->routes[i].address);
   1704         write(fd, &msg, n);
   1705     }
   1706     return 0;
   1707 }
   1708 
   1709 static int adev_close(hw_device_t *device)
   1710 {
   1711     ALOGI("adev_close()");
   1712     free(device);
   1713     return 0;
   1714 }
   1715 
   1716 static int adev_open(const hw_module_t* module, const char* name,
   1717                      hw_device_t** device)
   1718 {
   1719     ALOGI("adev_open(name=%s)", name);
   1720     struct submix_audio_device *rsxadev;
   1721 
   1722     if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
   1723         return -EINVAL;
   1724 
   1725     rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device));
   1726     if (!rsxadev)
   1727         return -ENOMEM;
   1728 
   1729     rsxadev->device.common.tag = HARDWARE_DEVICE_TAG;
   1730     rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
   1731     rsxadev->device.common.module = (struct hw_module_t *) module;
   1732     rsxadev->device.common.close = adev_close;
   1733 
   1734     rsxadev->device.init_check = adev_init_check;
   1735     rsxadev->device.set_voice_volume = adev_set_voice_volume;
   1736     rsxadev->device.set_master_volume = adev_set_master_volume;
   1737     rsxadev->device.get_master_volume = adev_get_master_volume;
   1738     rsxadev->device.set_master_mute = adev_set_master_mute;
   1739     rsxadev->device.get_master_mute = adev_get_master_mute;
   1740     rsxadev->device.set_mode = adev_set_mode;
   1741     rsxadev->device.set_mic_mute = adev_set_mic_mute;
   1742     rsxadev->device.get_mic_mute = adev_get_mic_mute;
   1743     rsxadev->device.set_parameters = adev_set_parameters;
   1744     rsxadev->device.get_parameters = adev_get_parameters;
   1745     rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size;
   1746     rsxadev->device.open_output_stream = adev_open_output_stream;
   1747     rsxadev->device.close_output_stream = adev_close_output_stream;
   1748     rsxadev->device.open_input_stream = adev_open_input_stream;
   1749     rsxadev->device.close_input_stream = adev_close_input_stream;
   1750     rsxadev->device.dump = adev_dump;
   1751 
   1752     for (int i=0 ; i < MAX_ROUTES ; i++) {
   1753             memset(&rsxadev->routes[i], 0, sizeof(route_config));
   1754             strcpy(rsxadev->routes[i].address, "");
   1755         }
   1756 
   1757     *device = &rsxadev->device.common;
   1758 
   1759     return 0;
   1760 }
   1761 
   1762 static struct hw_module_methods_t hal_module_methods = {
   1763     /* open */ adev_open,
   1764 };
   1765 
   1766 struct audio_module HAL_MODULE_INFO_SYM = {
   1767     /* common */ {
   1768         /* tag */                HARDWARE_MODULE_TAG,
   1769         /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1,
   1770         /* hal_api_version */    HARDWARE_HAL_API_VERSION,
   1771         /* id */                 AUDIO_HARDWARE_MODULE_ID,
   1772         /* name */               "Wifi Display audio HAL",
   1773         /* author */             "The Android Open Source Project",
   1774         /* methods */            &hal_module_methods,
   1775         /* dso */                NULL,
   1776         /* reserved */           { 0 },
   1777     },
   1778 };
   1779 
   1780 } //namespace android
   1781 
   1782 } //extern "C"
   1783