blob: 8fed8e4f928731f6dc9657a69262d090391c85a8 [file] [log] [blame]
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001/*
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"
Jean-Michel Trivi35a2c162012-09-17 10:13:26 -070018//#define LOG_NDEBUG 0
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070019
20#include <errno.h>
21#include <pthread.h>
22#include <stdint.h>
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070023#include <stdlib.h>
Stewart Milesc049a0a2014-05-01 09:03:27 -070024#include <sys/param.h>
25#include <sys/time.h>
Stewart Milese54c12c2014-05-01 09:03:27 -070026#include <sys/limits.h>
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070027
28#include <cutils/log.h>
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070029#include <cutils/properties.h>
Stewart Milesc049a0a2014-05-01 09:03:27 -070030#include <cutils/str_parms.h>
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070031
Stewart Milesc049a0a2014-05-01 09:03:27 -070032#include <hardware/audio.h>
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070033#include <hardware/hardware.h>
34#include <system/audio.h>
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070035
Stewart Milesc049a0a2014-05-01 09:03:27 -070036#include <media/AudioParameter.h>
37#include <media/AudioBufferProvider.h>
Jean-Michel Trivieec87702012-09-17 09:59:42 -070038#include <media/nbaio/MonoPipe.h>
39#include <media/nbaio/MonoPipeReader.h>
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070040
Jean-Michel Trivid4413032012-09-30 11:08:06 -070041#include <utils/String8.h>
Jean-Michel Trivid4413032012-09-30 11:08:06 -070042
Stewart Miles92854f52014-05-01 09:03:27 -070043#define LOG_STREAMS_TO_FILES 0
44#if LOG_STREAMS_TO_FILES
45#include <fcntl.h>
46#include <stdio.h>
47#include <sys/stat.h>
48#endif // LOG_STREAMS_TO_FILES
49
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -070050extern "C" {
51
52namespace android {
53
Stewart Milesc049a0a2014-05-01 09:03:27 -070054// Set to 1 to enable extremely verbose logging in this module.
55#define SUBMIX_VERBOSE_LOGGING 0
56#if SUBMIX_VERBOSE_LOGGING
57#define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__)
58#define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__)
59#else
60#define SUBMIX_ALOGV(...)
61#define SUBMIX_ALOGE(...)
62#endif // SUBMIX_VERBOSE_LOGGING
63
Stewart Miles3dd36f92014-05-01 09:03:27 -070064// NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe().
65#define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*8)
66// Value used to divide the MonoPipe() buffer into segments that are written to the source and
67// read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer
68// the minimum latency is the MonoPipe buffer size divided by this value.
69#define DEFAULT_PIPE_PERIOD_COUNT 4
Jean-Michel Trivieec87702012-09-17 09:59:42 -070070// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to
71// the duration of a record buffer at the current record sample rate (of the device, not of
72// the recording itself). Here we have:
73// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms
Jean-Michel Trivi6acd9662012-09-11 19:19:08 -070074#define MAX_READ_ATTEMPTS 3
Jean-Michel Trivieec87702012-09-17 09:59:42 -070075#define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty
Stewart Miles568e66f2014-05-01 09:03:27 -070076#define DEFAULT_SAMPLE_RATE_HZ 48000 // default sample rate
77// See NBAIO_Format frameworks/av/include/media/nbaio/NBAIO.h.
78#define DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT
Stewart Miles3dd36f92014-05-01 09:03:27 -070079// A legacy user of this device does not close the input stream when it shuts down, which
80// results in the application opening a new input stream before closing the old input stream
81// handle it was previously using. Setting this value to 1 allows multiple clients to open
82// multiple input streams from this device. If this option is enabled, each input stream returned
83// is *the same stream* which means that readers will race to read data from these streams.
84#define ENABLE_LEGACY_INPUT_OPEN 1
Stewart Milese54c12c2014-05-01 09:03:27 -070085// Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled.
86#define ENABLE_CHANNEL_CONVERSION 1
Stewart Miles02c2f712014-05-01 09:03:27 -070087// Whether resampling is enabled.
88#define ENABLE_RESAMPLING 1
Stewart Miles92854f52014-05-01 09:03:27 -070089#if LOG_STREAMS_TO_FILES
90// Folder to save stream log files to.
91#define LOG_STREAM_FOLDER "/data/misc/media"
92// Log filenames for input and output streams.
93#define LOG_STREAM_OUT_FILENAME LOG_STREAM_FOLDER "/r_submix_out.raw"
94#define LOG_STREAM_IN_FILENAME LOG_STREAM_FOLDER "/r_submix_in.raw"
95// File permissions for stream log files.
96#define LOG_STREAM_FILE_PERMISSIONS (S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH)
97#endif // LOG_STREAMS_TO_FILES
Jean-Michel Trivi793a8542014-10-14 15:31:51 -070098// limit for number of read error log entries to avoid spamming the logs
99#define MAX_READ_ERROR_LOGS 5
Stewart Miles3dd36f92014-05-01 09:03:27 -0700100
101// Common limits macros.
102#ifndef min
103#define min(a, b) ((a) < (b) ? (a) : (b))
104#endif // min
Stewart Milese54c12c2014-05-01 09:03:27 -0700105#ifndef max
106#define max(a, b) ((a) > (b) ? (a) : (b))
107#endif // max
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700108
Stewart Miles70726842014-05-01 09:03:27 -0700109// Set *result_variable_ptr to true if value_to_find is present in the array array_to_search,
110// otherwise set *result_variable_ptr to false.
111#define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \
112 { \
113 size_t i; \
114 *(result_variable_ptr) = false; \
115 for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \
116 if ((value_to_find) == (array_to_search)[i]) { \
117 *(result_variable_ptr) = true; \
118 break; \
119 } \
120 } \
121 }
122
Stewart Miles568e66f2014-05-01 09:03:27 -0700123// Configuration of the submix pipe.
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700124struct submix_config {
Stewart Miles70726842014-05-01 09:03:27 -0700125 // Channel mask field in this data structure is set to either input_channel_mask or
126 // output_channel_mask depending upon the last stream to be opened on this device.
127 struct audio_config common;
128 // Input stream and output stream channel masks. This is required since input and output
129 // channel bitfields are not equivalent.
130 audio_channel_mask_t input_channel_mask;
131 audio_channel_mask_t output_channel_mask;
Stewart Miles02c2f712014-05-01 09:03:27 -0700132#if ENABLE_RESAMPLING
133 // Input stream and output stream sample rates.
134 uint32_t input_sample_rate;
135 uint32_t output_sample_rate;
136#endif // ENABLE_RESAMPLING
Stewart Milese54c12c2014-05-01 09:03:27 -0700137 size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe.
Stewart Miles3dd36f92014-05-01 09:03:27 -0700138 size_t buffer_size_frames; // Size of the audio pipe in frames.
139 // Maximum number of frames buffered by the input and output streams.
140 size_t buffer_period_size_frames;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700141};
142
143struct submix_audio_device {
144 struct audio_hw_device device;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700145 bool input_standby;
Stewart Miles70726842014-05-01 09:03:27 -0700146 bool output_standby;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700147 submix_config config;
148 // Pipe variables: they handle the ring buffer that "pipes" audio:
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700149 // - from the submix virtual audio output == what needs to be played
150 // remotely, seen as an output for AudioFlinger
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700151 // - to the virtual audio source == what is captured by the component
152 // which "records" the submix / virtual audio source, and handles it as needed.
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700153 // A usecase example is one where the component capturing the audio is then sending it over
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700154 // 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.
Stewart Miles568e66f2014-05-01 09:03:27 -0700156 sp<MonoPipe> rsxSink;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700157 sp<MonoPipeReader> rsxSource;
Stewart Miles02c2f712014-05-01 09:03:27 -0700158#if ENABLE_RESAMPLING
159 // Buffer used as temporary storage for resampled data prior to returning data to the output
160 // stream.
161 int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES];
162#endif // ENABLE_RESAMPLING
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700163
Stewart Miles3dd36f92014-05-01 09:03:27 -0700164 // Pointers to the current input and output stream instances. rsxSink and rsxSource are
165 // destroyed if both and input and output streams are destroyed.
166 struct submix_stream_out *output;
167 struct submix_stream_in *input;
168
Stewart Miles568e66f2014-05-01 09:03:27 -0700169 // Device lock, also used to protect access to submix_audio_device from the input and output
170 // streams.
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700171 pthread_mutex_t lock;
172};
173
174struct submix_stream_out {
175 struct audio_stream_out stream;
176 struct submix_audio_device *dev;
Stewart Miles92854f52014-05-01 09:03:27 -0700177#if LOG_STREAMS_TO_FILES
178 int log_fd;
179#endif // LOG_STREAMS_TO_FILES
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700180};
181
182struct submix_stream_in {
183 struct audio_stream_in stream;
184 struct submix_audio_device *dev;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700185 bool output_standby; // output standby state as seen from record thread
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700186
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700187 // wall clock when recording starts
188 struct timespec record_start_time;
189 // how many frames have been requested to be read
190 int64_t read_counter_frames;
Stewart Miles3dd36f92014-05-01 09:03:27 -0700191
192#if ENABLE_LEGACY_INPUT_OPEN
193 // Number of references to this input stream.
194 volatile int32_t ref_count;
195#endif // ENABLE_LEGACY_INPUT_OPEN
Stewart Miles92854f52014-05-01 09:03:27 -0700196#if LOG_STREAMS_TO_FILES
197 int log_fd;
198#endif // LOG_STREAMS_TO_FILES
Jean-Michel Trivi793a8542014-10-14 15:31:51 -0700199
200 volatile int16_t read_error_count;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700201};
Jean-Michel Trivi6acd9662012-09-11 19:19:08 -0700202
Stewart Miles70726842014-05-01 09:03:27 -0700203// Determine whether the specified sample rate is supported by the submix module.
204static bool sample_rate_supported(const uint32_t sample_rate)
205{
206 // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp.
207 static const unsigned int supported_sample_rates[] = {
208 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
209 };
210 bool return_value;
211 SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value);
212 return return_value;
213}
214
215// Determine whether the specified sample rate is supported, if it is return the specified sample
216// rate, otherwise return the default sample rate for the submix module.
217static uint32_t get_supported_sample_rate(uint32_t sample_rate)
218{
219 return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ;
220}
221
222// Determine whether the specified channel in mask is supported by the submix module.
223static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask)
224{
225 // Set of channel in masks supported by Format_from_SR_C()
226 // frameworks/av/media/libnbaio/NAIO.cpp.
227 static const audio_channel_mask_t supported_channel_in_masks[] = {
228 AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO,
229 };
230 bool return_value;
231 SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value);
232 return return_value;
233}
234
235// Determine whether the specified channel in mask is supported, if it is return the specified
236// channel in mask, otherwise return the default channel in mask for the submix module.
237static audio_channel_mask_t get_supported_channel_in_mask(
238 const audio_channel_mask_t channel_in_mask)
239{
240 return channel_in_mask_supported(channel_in_mask) ? channel_in_mask :
241 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO);
242}
243
244// Determine whether the specified channel out mask is supported by the submix module.
245static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask)
246{
247 // Set of channel out masks supported by Format_from_SR_C()
248 // frameworks/av/media/libnbaio/NAIO.cpp.
249 static const audio_channel_mask_t supported_channel_out_masks[] = {
250 AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO,
251 };
252 bool return_value;
253 SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value);
254 return return_value;
255}
256
257// Determine whether the specified channel out mask is supported, if it is return the specified
258// channel out mask, otherwise return the default channel out mask for the submix module.
259static audio_channel_mask_t get_supported_channel_out_mask(
260 const audio_channel_mask_t channel_out_mask)
261{
262 return channel_out_mask_supported(channel_out_mask) ? channel_out_mask :
263 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO);
264}
265
Stewart Milesf645c5e2014-05-01 09:03:27 -0700266// Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the
267// structure.
268static struct submix_stream_out * audio_stream_out_get_submix_stream_out(
269 struct audio_stream_out * const stream)
270{
271 ALOG_ASSERT(stream);
272 return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) -
273 offsetof(struct submix_stream_out, stream));
274}
275
276// Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure.
277static struct submix_stream_out * audio_stream_get_submix_stream_out(
278 struct audio_stream * const stream)
279{
280 ALOG_ASSERT(stream);
281 return audio_stream_out_get_submix_stream_out(
282 reinterpret_cast<struct audio_stream_out *>(stream));
283}
284
285// Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the
286// structure.
287static struct submix_stream_in * audio_stream_in_get_submix_stream_in(
288 struct audio_stream_in * const stream)
289{
290 ALOG_ASSERT(stream);
291 return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) -
292 offsetof(struct submix_stream_in, stream));
293}
294
295// Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure.
296static struct submix_stream_in * audio_stream_get_submix_stream_in(
297 struct audio_stream * const stream)
298{
299 ALOG_ASSERT(stream);
300 return audio_stream_in_get_submix_stream_in(
301 reinterpret_cast<struct audio_stream_in *>(stream));
302}
303
304// Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within
305// the structure.
306static struct submix_audio_device * audio_hw_device_get_submix_audio_device(
307 struct audio_hw_device *device)
308{
309 ALOG_ASSERT(device);
310 return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) -
311 offsetof(struct submix_audio_device, device));
312}
313
Stewart Miles70726842014-05-01 09:03:27 -0700314// Compare an audio_config with input channel mask and an audio_config with output channel mask
315// returning false if they do *not* match, true otherwise.
316static bool audio_config_compare(const audio_config * const input_config,
317 const audio_config * const output_config)
318{
Stewart Milese54c12c2014-05-01 09:03:27 -0700319#if !ENABLE_CHANNEL_CONVERSION
Eric Laurentdd45fd32014-07-01 20:32:28 -0700320 const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask);
321 const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask);
Stewart Miles3dd36f92014-05-01 09:03:27 -0700322 if (input_channels != output_channels) {
323 ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d",
324 input_channels, output_channels);
Stewart Miles70726842014-05-01 09:03:27 -0700325 return false;
326 }
Stewart Milese54c12c2014-05-01 09:03:27 -0700327#endif // !ENABLE_CHANNEL_CONVERSION
Stewart Miles02c2f712014-05-01 09:03:27 -0700328#if ENABLE_RESAMPLING
329 if (input_config->sample_rate != output_config->sample_rate &&
Eric Laurentdd45fd32014-07-01 20:32:28 -0700330 audio_channel_count_from_in_mask(input_config->channel_mask) != 1) {
Stewart Miles02c2f712014-05-01 09:03:27 -0700331#else
Stewart Miles70726842014-05-01 09:03:27 -0700332 if (input_config->sample_rate != output_config->sample_rate) {
Stewart Miles02c2f712014-05-01 09:03:27 -0700333#endif // ENABLE_RESAMPLING
Stewart Miles70726842014-05-01 09:03:27 -0700334 ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul",
335 input_config->sample_rate, output_config->sample_rate);
336 return false;
337 }
338 if (input_config->format != output_config->format) {
339 ALOGE("audio_config_compare() format mismatch %x vs. %x",
340 input_config->format, output_config->format);
341 return false;
342 }
343 // This purposely ignores offload_info as it's not required for the submix device.
344 return true;
345}
346
Stewart Miles3dd36f92014-05-01 09:03:27 -0700347// If one doesn't exist, create a pipe for the submix audio device rsxadev of size
348// buffer_size_frames and optionally associate "in" or "out" with the submix audio device.
349static void submix_audio_device_create_pipe(struct submix_audio_device * const rsxadev,
350 const struct audio_config * const config,
351 const size_t buffer_size_frames,
352 const uint32_t buffer_period_count,
353 struct submix_stream_in * const in,
354 struct submix_stream_out * const out)
355{
356 ALOG_ASSERT(in || out);
Jean-Michel Trivi793a8542014-10-14 15:31:51 -0700357 ALOGD("submix_audio_device_create_pipe()");
Stewart Miles3dd36f92014-05-01 09:03:27 -0700358 pthread_mutex_lock(&rsxadev->lock);
359 // Save a reference to the specified input or output stream and the associated channel
360 // mask.
361 if (in) {
362 rsxadev->input = in;
363 rsxadev->config.input_channel_mask = config->channel_mask;
Stewart Miles02c2f712014-05-01 09:03:27 -0700364#if ENABLE_RESAMPLING
365 rsxadev->config.input_sample_rate = config->sample_rate;
366 // If the output isn't configured yet, set the output sample rate to the maximum supported
367 // sample rate such that the smallest possible input buffer is created.
368 if (!rsxadev->output) {
369 rsxadev->config.output_sample_rate = 48000;
370 }
371#endif // ENABLE_RESAMPLING
Stewart Miles3dd36f92014-05-01 09:03:27 -0700372 }
373 if (out) {
374 rsxadev->output = out;
375 rsxadev->config.output_channel_mask = config->channel_mask;
Stewart Miles02c2f712014-05-01 09:03:27 -0700376#if ENABLE_RESAMPLING
377 rsxadev->config.output_sample_rate = config->sample_rate;
378#endif // ENABLE_RESAMPLING
Stewart Miles3dd36f92014-05-01 09:03:27 -0700379 }
380 // If a pipe isn't associated with the device, create one.
381 if (rsxadev->rsxSink == NULL || rsxadev->rsxSource == NULL) {
382 struct submix_config * const device_config = &rsxadev->config;
Eric Laurentdd45fd32014-07-01 20:32:28 -0700383 uint32_t channel_count;
384 if (out)
385 channel_count = audio_channel_count_from_out_mask(config->channel_mask);
386 else
387 channel_count = audio_channel_count_from_in_mask(config->channel_mask);
Stewart Miles10f1a802014-06-09 20:54:37 -0700388#if ENABLE_CHANNEL_CONVERSION
389 // If channel conversion is enabled, allocate enough space for the maximum number of
390 // possible channels stored in the pipe for the situation when the number of channels in
391 // the output stream don't match the number in the input stream.
392 const uint32_t pipe_channel_count = max(channel_count, 2);
393#else
394 const uint32_t pipe_channel_count = channel_count;
395#endif // ENABLE_CHANNEL_CONVERSION
396 const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count,
397 config->format);
Stewart Miles3dd36f92014-05-01 09:03:27 -0700398 const NBAIO_Format offers[1] = {format};
399 size_t numCounterOffers = 0;
400 // Create a MonoPipe with optional blocking set to true.
401 MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/);
402 // Negotiation between the source and sink cannot fail as the device open operation
403 // creates both ends of the pipe using the same audio format.
404 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers);
405 ALOG_ASSERT(index == 0);
406 MonoPipeReader* source = new MonoPipeReader(sink);
407 numCounterOffers = 0;
408 index = source->negotiate(offers, 1, NULL, numCounterOffers);
409 ALOG_ASSERT(index == 0);
410 ALOGV("submix_audio_device_create_pipe(): created pipe");
411
412 // Save references to the source and sink.
413 ALOG_ASSERT(rsxadev->rsxSink == NULL);
414 ALOG_ASSERT(rsxadev->rsxSource == NULL);
415 rsxadev->rsxSink = sink;
416 rsxadev->rsxSource = source;
417 // Store the sanitized audio format in the device so that it's possible to determine
418 // the format of the pipe source when opening the input device.
419 memcpy(&device_config->common, config, sizeof(device_config->common));
420 device_config->buffer_size_frames = sink->maxFrames();
421 device_config->buffer_period_size_frames = device_config->buffer_size_frames /
422 buffer_period_count;
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700423 if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream);
424 if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream);
Stewart Miles10f1a802014-06-09 20:54:37 -0700425#if ENABLE_CHANNEL_CONVERSION
426 // Calculate the pipe frame size based upon the number of channels.
427 device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) /
428 channel_count;
429#endif // ENABLE_CHANNEL_CONVERSION
Stewart Milese54c12c2014-05-01 09:03:27 -0700430 SUBMIX_ALOGV("submix_audio_device_create_pipe(): pipe frame size %zd, pipe size %zd, "
431 "period size %zd", device_config->pipe_frame_size,
432 device_config->buffer_size_frames, device_config->buffer_period_size_frames);
Stewart Miles3dd36f92014-05-01 09:03:27 -0700433 }
434 pthread_mutex_unlock(&rsxadev->lock);
435}
436
437// Release references to the sink and source. Input and output threads may maintain references
438// to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use
439// before they shutdown.
440static void submix_audio_device_release_pipe(struct submix_audio_device * const rsxadev)
441{
Jean-Michel Trivi793a8542014-10-14 15:31:51 -0700442 ALOGD("submix_audio_device_release_pipe()");
Stewart Miles3dd36f92014-05-01 09:03:27 -0700443 rsxadev->rsxSink.clear();
444 rsxadev->rsxSource.clear();
445}
446
447// Remove references to the specified input and output streams. When the device no longer
448// references input and output streams destroy the associated pipe.
449static void submix_audio_device_destroy_pipe(struct submix_audio_device * const rsxadev,
450 const struct submix_stream_in * const in,
451 const struct submix_stream_out * const out)
452{
453 MonoPipe* sink;
454 pthread_mutex_lock(&rsxadev->lock);
455 ALOGV("submix_audio_device_destroy_pipe()");
456 ALOG_ASSERT(in == NULL || rsxadev->input == in);
457 ALOG_ASSERT(out == NULL || rsxadev->output == out);
458 if (in != NULL) {
459#if ENABLE_LEGACY_INPUT_OPEN
460 const_cast<struct submix_stream_in*>(in)->ref_count--;
461 if (in->ref_count == 0) {
462 rsxadev->input = NULL;
463 }
464 ALOGV("submix_audio_device_destroy_pipe(): input ref_count %d", in->ref_count);
465#else
466 rsxadev->input = NULL;
467#endif // ENABLE_LEGACY_INPUT_OPEN
468 }
469 if (out != NULL) rsxadev->output = NULL;
Jean-Michel Trivi793a8542014-10-14 15:31:51 -0700470 if (rsxadev->input == NULL && rsxadev->output == NULL) {
Stewart Miles3dd36f92014-05-01 09:03:27 -0700471 submix_audio_device_release_pipe(rsxadev);
Jean-Michel Trivi793a8542014-10-14 15:31:51 -0700472 ALOGD("submix_audio_device_destroy_pipe(): pipe destroyed");
Stewart Miles3dd36f92014-05-01 09:03:27 -0700473 }
474 pthread_mutex_unlock(&rsxadev->lock);
475}
476
Stewart Miles70726842014-05-01 09:03:27 -0700477// Sanitize the user specified audio config for a submix input / output stream.
478static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format)
479{
480 config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) :
481 get_supported_channel_out_mask(config->channel_mask);
482 config->sample_rate = get_supported_sample_rate(config->sample_rate);
483 config->format = DEFAULT_FORMAT;
484}
485
486// Verify a submix input or output stream can be opened.
487static bool submix_open_validate(const struct submix_audio_device * const rsxadev,
488 pthread_mutex_t * const lock,
489 const struct audio_config * const config,
490 const bool opening_input)
491{
Stewart Miles3dd36f92014-05-01 09:03:27 -0700492 bool input_open;
493 bool output_open;
Stewart Miles70726842014-05-01 09:03:27 -0700494 audio_config pipe_config;
495
496 // Query the device for the current audio config and whether input and output streams are open.
497 pthread_mutex_lock(lock);
Stewart Miles3dd36f92014-05-01 09:03:27 -0700498 output_open = rsxadev->output != NULL;
499 input_open = rsxadev->input != NULL;
Stewart Miles70726842014-05-01 09:03:27 -0700500 memcpy(&pipe_config, &rsxadev->config.common, sizeof(pipe_config));
501 pthread_mutex_unlock(lock);
502
Stewart Miles3dd36f92014-05-01 09:03:27 -0700503 // If the stream is already open, don't open it again.
504 if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) {
505 ALOGE("submix_open_validate(): %s stream already open.", opening_input ? "Input" :
506 "Output");
507 return false;
508 }
509
510 SUBMIX_ALOGV("submix_open_validate(): sample rate=%d format=%x "
511 "%s_channel_mask=%x", config->sample_rate, config->format,
512 opening_input ? "in" : "out", config->channel_mask);
513
514 // If either stream is open, verify the existing audio config the pipe matches the user
Stewart Miles70726842014-05-01 09:03:27 -0700515 // specified config.
Stewart Miles3dd36f92014-05-01 09:03:27 -0700516 if (input_open || output_open) {
Stewart Miles70726842014-05-01 09:03:27 -0700517 const audio_config * const input_config = opening_input ? config : &pipe_config;
518 const audio_config * const output_config = opening_input ? &pipe_config : config;
519 // Get the channel mask of the open device.
520 pipe_config.channel_mask =
521 opening_input ? rsxadev->config.output_channel_mask :
522 rsxadev->config.input_channel_mask;
523 if (!audio_config_compare(input_config, output_config)) {
524 ALOGE("submix_open_validate(): Unsupported format.");
Stewart Miles3dd36f92014-05-01 09:03:27 -0700525 return false;
Stewart Miles70726842014-05-01 09:03:27 -0700526 }
527 }
528 return true;
529}
530
Stewart Milese54c12c2014-05-01 09:03:27 -0700531// Calculate the maximum size of the pipe buffer in frames for the specified stream.
532static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream,
533 const struct submix_config *config,
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700534 const size_t pipe_frames,
535 const size_t stream_frame_size)
Stewart Milese54c12c2014-05-01 09:03:27 -0700536{
Stewart Milese54c12c2014-05-01 09:03:27 -0700537 const size_t pipe_frame_size = config->pipe_frame_size;
538 const size_t max_frame_size = max(stream_frame_size, pipe_frame_size);
539 return (pipe_frames * config->pipe_frame_size) / max_frame_size;
540}
541
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700542/* audio HAL functions */
543
544static uint32_t out_get_sample_rate(const struct audio_stream *stream)
545{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700546 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
547 const_cast<struct audio_stream *>(stream));
Stewart Miles02c2f712014-05-01 09:03:27 -0700548#if ENABLE_RESAMPLING
549 const uint32_t out_rate = out->dev->config.output_sample_rate;
550#else
Stewart Miles70726842014-05-01 09:03:27 -0700551 const uint32_t out_rate = out->dev->config.common.sample_rate;
Stewart Miles02c2f712014-05-01 09:03:27 -0700552#endif // ENABLE_RESAMPLING
Stewart Milesc049a0a2014-05-01 09:03:27 -0700553 SUBMIX_ALOGV("out_get_sample_rate() returns %u", out_rate);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700554 return out_rate;
555}
556
557static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
558{
Stewart Miles02c2f712014-05-01 09:03:27 -0700559 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
560#if ENABLE_RESAMPLING
561 // The sample rate of the stream can't be changed once it's set since this would change the
562 // output buffer size and hence break playback to the shared pipe.
563 if (rate != out->dev->config.output_sample_rate) {
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700564 ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from "
Stewart Miles02c2f712014-05-01 09:03:27 -0700565 "%u to %u", out->dev->config.output_sample_rate, rate);
566 return -ENOSYS;
567 }
568#endif // ENABLE_RESAMPLING
Stewart Miles70726842014-05-01 09:03:27 -0700569 if (!sample_rate_supported(rate)) {
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700570 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
571 return -ENOSYS;
572 }
Stewart Milesc049a0a2014-05-01 09:03:27 -0700573 SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate);
Stewart Miles70726842014-05-01 09:03:27 -0700574 out->dev->config.common.sample_rate = rate;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700575 return 0;
576}
577
578static size_t out_get_buffer_size(const struct audio_stream *stream)
579{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700580 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
581 const_cast<struct audio_stream *>(stream));
Stewart Miles568e66f2014-05-01 09:03:27 -0700582 const struct submix_config * const config = &out->dev->config;
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700583 const size_t stream_frame_size =
584 audio_stream_out_frame_size((const struct audio_stream_out *)stream);
Stewart Milese54c12c2014-05-01 09:03:27 -0700585 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700586 stream, config, config->buffer_period_size_frames, stream_frame_size);
587 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
Stewart Miles568e66f2014-05-01 09:03:27 -0700588 SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames",
Stewart Milese54c12c2014-05-01 09:03:27 -0700589 buffer_size_bytes, buffer_size_frames);
590 return buffer_size_bytes;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700591}
592
593static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
594{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700595 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
596 const_cast<struct audio_stream *>(stream));
Stewart Miles70726842014-05-01 09:03:27 -0700597 uint32_t channel_mask = out->dev->config.output_channel_mask;
Stewart Miles568e66f2014-05-01 09:03:27 -0700598 SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask);
599 return channel_mask;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700600}
601
602static audio_format_t out_get_format(const struct audio_stream *stream)
603{
Stewart Miles568e66f2014-05-01 09:03:27 -0700604 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
605 const_cast<struct audio_stream *>(stream));
Stewart Miles70726842014-05-01 09:03:27 -0700606 const audio_format_t format = out->dev->config.common.format;
Stewart Miles568e66f2014-05-01 09:03:27 -0700607 SUBMIX_ALOGV("out_get_format() returns %x", format);
608 return format;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700609}
610
611static int out_set_format(struct audio_stream *stream, audio_format_t format)
612{
Stewart Miles568e66f2014-05-01 09:03:27 -0700613 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
Stewart Miles70726842014-05-01 09:03:27 -0700614 if (format != out->dev->config.common.format) {
Stewart Milesc049a0a2014-05-01 09:03:27 -0700615 ALOGE("out_set_format(format=%x) format unsupported", format);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700616 return -ENOSYS;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700617 }
Stewart Milesc049a0a2014-05-01 09:03:27 -0700618 SUBMIX_ALOGV("out_set_format(format=%x)", format);
619 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700620}
621
622static int out_standby(struct audio_stream *stream)
623{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700624 struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_out(stream)->dev;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700625 ALOGI("out_standby()");
626
Stewart Milesf645c5e2014-05-01 09:03:27 -0700627 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700628
Stewart Milesf645c5e2014-05-01 09:03:27 -0700629 rsxadev->output_standby = true;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700630
Stewart Milesf645c5e2014-05-01 09:03:27 -0700631 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700632
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700633 return 0;
634}
635
636static int out_dump(const struct audio_stream *stream, int fd)
637{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700638 (void)stream;
639 (void)fd;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700640 return 0;
641}
642
643static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
644{
Jean-Michel Trivid4413032012-09-30 11:08:06 -0700645 int exiting = -1;
646 AudioParameter parms = AudioParameter(String8(kvpairs));
Stewart Milesc049a0a2014-05-01 09:03:27 -0700647 SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs);
Stewart Miles3dd36f92014-05-01 09:03:27 -0700648
Jean-Michel Trivid4413032012-09-30 11:08:06 -0700649 // FIXME this is using hard-coded strings but in the future, this functionality will be
650 // converted to use audio HAL extensions required to support tunneling
651 if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) {
Stewart Milesf645c5e2014-05-01 09:03:27 -0700652 struct submix_audio_device * const rsxadev =
653 audio_stream_get_submix_stream_out(stream)->dev;
654 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800655 { // using the sink
Stewart Miles3dd36f92014-05-01 09:03:27 -0700656 sp<MonoPipe> sink = rsxadev->rsxSink;
Stewart Milesf645c5e2014-05-01 09:03:27 -0700657 if (sink == NULL) {
658 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800659 return 0;
660 }
Jean-Michel Trivid4413032012-09-30 11:08:06 -0700661
Jean-Michel Trivi793a8542014-10-14 15:31:51 -0700662 ALOGD("out_set_parameters(): shutting down MonoPipe sink");
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800663 sink->shutdown(true);
664 } // done using the sink
Stewart Milesf645c5e2014-05-01 09:03:27 -0700665 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivid4413032012-09-30 11:08:06 -0700666 }
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700667 return 0;
668}
669
670static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
671{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700672 (void)stream;
673 (void)keys;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700674 return strdup("");
675}
676
677static uint32_t out_get_latency(const struct audio_stream_out *stream)
678{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700679 const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(
680 const_cast<struct audio_stream_out *>(stream));
Stewart Miles568e66f2014-05-01 09:03:27 -0700681 const struct submix_config * const config = &out->dev->config;
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700682 const size_t stream_frame_size =
683 audio_stream_out_frame_size(stream);
Stewart Milese54c12c2014-05-01 09:03:27 -0700684 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700685 &stream->common, config, config->buffer_size_frames, stream_frame_size);
Stewart Miles10f1a802014-06-09 20:54:37 -0700686 const uint32_t sample_rate = out_get_sample_rate(&stream->common);
687 const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate;
Stewart Milese54c12c2014-05-01 09:03:27 -0700688 SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u",
Stewart Miles10f1a802014-06-09 20:54:37 -0700689 latency_ms, buffer_size_frames, sample_rate);
Stewart Miles568e66f2014-05-01 09:03:27 -0700690 return latency_ms;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700691}
692
693static int out_set_volume(struct audio_stream_out *stream, float left,
694 float right)
695{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700696 (void)stream;
697 (void)left;
698 (void)right;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700699 return -ENOSYS;
700}
701
702static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
703 size_t bytes)
704{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700705 SUBMIX_ALOGV("out_write(bytes=%zd)", bytes);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700706 ssize_t written_frames = 0;
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700707 const size_t frame_size = audio_stream_out_frame_size(stream);
Stewart Miles3dd36f92014-05-01 09:03:27 -0700708 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream);
709 struct submix_audio_device * const rsxadev = out->dev;
Jean-Michel Trivi90b0fbd2012-10-30 19:03:22 -0700710 const size_t frames = bytes / frame_size;
711
Stewart Milesf645c5e2014-05-01 09:03:27 -0700712 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700713
Stewart Milesf645c5e2014-05-01 09:03:27 -0700714 rsxadev->output_standby = false;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700715
Stewart Miles3dd36f92014-05-01 09:03:27 -0700716 sp<MonoPipe> sink = rsxadev->rsxSink;
Stewart Milesf645c5e2014-05-01 09:03:27 -0700717 if (sink != NULL) {
Jean-Michel Trivi90b0fbd2012-10-30 19:03:22 -0700718 if (sink->isShutdown()) {
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800719 sink.clear();
Stewart Milesf645c5e2014-05-01 09:03:27 -0700720 pthread_mutex_unlock(&rsxadev->lock);
Stewart Milesc049a0a2014-05-01 09:03:27 -0700721 SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write.");
Jean-Michel Trivi90b0fbd2012-10-30 19:03:22 -0700722 // the pipe has already been shutdown, this buffer will be lost but we must
723 // simulate timing so we don't drain the output faster than realtime
724 usleep(frames * 1000000 / out_get_sample_rate(&stream->common));
725 return bytes;
726 }
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700727 } else {
Stewart Milesf645c5e2014-05-01 09:03:27 -0700728 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700729 ALOGE("out_write without a pipe!");
730 ALOG_ASSERT("out_write without a pipe!");
731 return 0;
732 }
733
Stewart Miles2d199fe2014-05-01 09:03:27 -0700734 // If the write to the sink would block when no input stream is present, flush enough frames
735 // from the pipe to make space to write the most recent data.
736 {
737 const size_t availableToWrite = sink->availableToWrite();
738 sp<MonoPipeReader> source = rsxadev->rsxSource;
739 if (rsxadev->input == NULL && availableToWrite < frames) {
740 static uint8_t flush_buffer[64];
741 const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size;
742 size_t frames_to_flush_from_source = frames - availableToWrite;
743 SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking",
744 frames_to_flush_from_source);
745 while (frames_to_flush_from_source) {
746 const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames);
747 frames_to_flush_from_source -= flush_size;
748 source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS);
749 }
750 }
751 }
752
Stewart Milesf645c5e2014-05-01 09:03:27 -0700753 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700754
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700755 written_frames = sink->write(buffer, frames);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800756
Stewart Miles92854f52014-05-01 09:03:27 -0700757#if LOG_STREAMS_TO_FILES
758 if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size);
759#endif // LOG_STREAMS_TO_FILES
760
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700761 if (written_frames < 0) {
762 if (written_frames == (ssize_t)NEGOTIATE) {
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700763 ALOGE("out_write() write to pipe returned NEGOTIATE");
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700764
Stewart Milesf645c5e2014-05-01 09:03:27 -0700765 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800766 sink.clear();
Stewart Milesf645c5e2014-05-01 09:03:27 -0700767 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700768
769 written_frames = 0;
Jean-Michel Trivi6acd9662012-09-11 19:19:08 -0700770 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700771 } else {
772 // write() returned UNDERRUN or WOULD_BLOCK, retry
Colin Cross5685a082014-04-18 15:45:42 -0700773 ALOGE("out_write() write to pipe returned unexpected %zd", written_frames);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700774 written_frames = sink->write(buffer, frames);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700775 }
776 }
777
Stewart Milesf645c5e2014-05-01 09:03:27 -0700778 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800779 sink.clear();
Stewart Milesf645c5e2014-05-01 09:03:27 -0700780 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700781
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700782 if (written_frames < 0) {
Colin Cross5685a082014-04-18 15:45:42 -0700783 ALOGE("out_write() failed writing to pipe with %zd", written_frames);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700784 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700785 }
Stewart Milesc049a0a2014-05-01 09:03:27 -0700786 const ssize_t written_bytes = written_frames * frame_size;
Stewart Miles02c2f712014-05-01 09:03:27 -0700787 SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames);
Stewart Milesc049a0a2014-05-01 09:03:27 -0700788 return written_bytes;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700789}
790
791static int out_get_render_position(const struct audio_stream_out *stream,
792 uint32_t *dsp_frames)
793{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700794 (void)stream;
795 (void)dsp_frames;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700796 return -EINVAL;
797}
798
799static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
800{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700801 (void)stream;
802 (void)effect;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700803 return 0;
804}
805
806static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
807{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700808 (void)stream;
809 (void)effect;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700810 return 0;
811}
812
813static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
814 int64_t *timestamp)
815{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700816 (void)stream;
817 (void)timestamp;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700818 return -EINVAL;
819}
820
821/** audio_stream_in implementation **/
822static uint32_t in_get_sample_rate(const struct audio_stream *stream)
823{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700824 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
825 const_cast<struct audio_stream*>(stream));
Stewart Miles02c2f712014-05-01 09:03:27 -0700826#if ENABLE_RESAMPLING
827 const uint32_t rate = in->dev->config.input_sample_rate;
828#else
829 const uint32_t rate = in->dev->config.common.sample_rate;
830#endif // ENABLE_RESAMPLING
831 SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate);
832 return rate;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700833}
834
835static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
836{
Stewart Miles568e66f2014-05-01 09:03:27 -0700837 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
Stewart Miles02c2f712014-05-01 09:03:27 -0700838#if ENABLE_RESAMPLING
839 // The sample rate of the stream can't be changed once it's set since this would change the
840 // input buffer size and hence break recording from the shared pipe.
841 if (rate != in->dev->config.input_sample_rate) {
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700842 ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from "
Stewart Miles02c2f712014-05-01 09:03:27 -0700843 "%u to %u", in->dev->config.input_sample_rate, rate);
844 return -ENOSYS;
845 }
846#endif // ENABLE_RESAMPLING
Stewart Miles70726842014-05-01 09:03:27 -0700847 if (!sample_rate_supported(rate)) {
848 ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate);
849 return -ENOSYS;
850 }
851 in->dev->config.common.sample_rate = rate;
Stewart Miles568e66f2014-05-01 09:03:27 -0700852 SUBMIX_ALOGV("in_set_sample_rate() set %u", rate);
853 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700854}
855
856static size_t in_get_buffer_size(const struct audio_stream *stream)
857{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700858 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
859 const_cast<struct audio_stream*>(stream));
Stewart Milese54c12c2014-05-01 09:03:27 -0700860 const struct submix_config * const config = &in->dev->config;
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700861 const size_t stream_frame_size =
862 audio_stream_in_frame_size((const struct audio_stream_in *)stream);
Stewart Miles02c2f712014-05-01 09:03:27 -0700863 size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700864 stream, config, config->buffer_period_size_frames, stream_frame_size);
Stewart Miles02c2f712014-05-01 09:03:27 -0700865#if ENABLE_RESAMPLING
866 // Scale the size of the buffer based upon the maximum number of frames that could be returned
867 // given the ratio of output to input sample rate.
868 buffer_size_frames = (size_t)(((float)buffer_size_frames *
869 (float)config->input_sample_rate) /
870 (float)config->output_sample_rate);
871#endif // ENABLE_RESAMPLING
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700872 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
Stewart Milese54c12c2014-05-01 09:03:27 -0700873 SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes,
874 buffer_size_frames);
875 return buffer_size_bytes;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700876}
877
878static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
879{
Stewart Miles70726842014-05-01 09:03:27 -0700880 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
881 const_cast<struct audio_stream*>(stream));
882 const audio_channel_mask_t channel_mask = in->dev->config.input_channel_mask;
883 SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask);
884 return channel_mask;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700885}
886
887static audio_format_t in_get_format(const struct audio_stream *stream)
888{
Stewart Miles568e66f2014-05-01 09:03:27 -0700889 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
Stewart Miles70726842014-05-01 09:03:27 -0700890 const_cast<struct audio_stream*>(stream));
891 const audio_format_t format = in->dev->config.common.format;
Stewart Miles568e66f2014-05-01 09:03:27 -0700892 SUBMIX_ALOGV("in_get_format() returns %x", format);
893 return format;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700894}
895
896static int in_set_format(struct audio_stream *stream, audio_format_t format)
897{
Stewart Miles568e66f2014-05-01 09:03:27 -0700898 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
Stewart Miles70726842014-05-01 09:03:27 -0700899 if (format != in->dev->config.common.format) {
Stewart Milesc049a0a2014-05-01 09:03:27 -0700900 ALOGE("in_set_format(format=%x) format unsupported", format);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700901 return -ENOSYS;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700902 }
Stewart Milesc049a0a2014-05-01 09:03:27 -0700903 SUBMIX_ALOGV("in_set_format(format=%x)", format);
904 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700905}
906
907static int in_standby(struct audio_stream *stream)
908{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700909 struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_in(stream)->dev;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700910 ALOGI("in_standby()");
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700911
Stewart Milesf645c5e2014-05-01 09:03:27 -0700912 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700913
Stewart Milesf645c5e2014-05-01 09:03:27 -0700914 rsxadev->input_standby = true;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700915
Stewart Milesf645c5e2014-05-01 09:03:27 -0700916 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700917
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700918 return 0;
919}
920
921static int in_dump(const struct audio_stream *stream, int fd)
922{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700923 (void)stream;
924 (void)fd;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700925 return 0;
926}
927
928static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
929{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700930 (void)stream;
931 (void)kvpairs;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700932 return 0;
933}
934
935static char * in_get_parameters(const struct audio_stream *stream,
936 const char *keys)
937{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700938 (void)stream;
939 (void)keys;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700940 return strdup("");
941}
942
943static int in_set_gain(struct audio_stream_in *stream, float gain)
944{
Stewart Milesc049a0a2014-05-01 09:03:27 -0700945 (void)stream;
946 (void)gain;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700947 return 0;
948}
949
950static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
951 size_t bytes)
952{
Stewart Milesf645c5e2014-05-01 09:03:27 -0700953 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream);
954 struct submix_audio_device * const rsxadev = in->dev;
Stewart Milese54c12c2014-05-01 09:03:27 -0700955 struct audio_config *format;
Eric Laurentc5ae6a02014-07-02 13:45:32 -0700956 const size_t frame_size = audio_stream_in_frame_size(stream);
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700957 const size_t frames_to_read = bytes / frame_size;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700958
Stewart Milesc049a0a2014-05-01 09:03:27 -0700959 SUBMIX_ALOGV("in_read bytes=%zu", bytes);
Stewart Milesf645c5e2014-05-01 09:03:27 -0700960 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700961
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700962 const bool output_standby_transition = (in->output_standby != in->dev->output_standby);
Stewart Milesf645c5e2014-05-01 09:03:27 -0700963 in->output_standby = rsxadev->output_standby;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700964
Stewart Milesf645c5e2014-05-01 09:03:27 -0700965 if (rsxadev->input_standby || output_standby_transition) {
966 rsxadev->input_standby = false;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700967 // keep track of when we exit input standby (== first read == start "real recording")
968 // or when we start recording silence, and reset projected time
969 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time);
970 if (rc == 0) {
971 in->read_counter_frames = 0;
972 }
973 }
974
975 in->read_counter_frames += frames_to_read;
Jean-Michel Trivieec87702012-09-17 09:59:42 -0700976 size_t remaining_frames = frames_to_read;
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800977
978 {
979 // about to read from audio source
Stewart Milesf645c5e2014-05-01 09:03:27 -0700980 sp<MonoPipeReader> source = rsxadev->rsxSource;
981 if (source == NULL) {
Jean-Michel Trivi793a8542014-10-14 15:31:51 -0700982 in->read_error_count++;// ok if it rolls over
983 ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS,
984 "no audio pipe yet we're trying to read! (not all errors will be logged)");
Stewart Milesf645c5e2014-05-01 09:03:27 -0700985 pthread_mutex_unlock(&rsxadev->lock);
Stewart Miles3dd36f92014-05-01 09:03:27 -0700986 usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common));
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800987 memset(buffer, 0, bytes);
988 return bytes;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -0700989 }
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800990
Stewart Milesf645c5e2014-05-01 09:03:27 -0700991 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -0800992
993 // read the data from the pipe (it's non blocking)
994 int attempts = 0;
995 char* buff = (char*)buffer;
Stewart Milese54c12c2014-05-01 09:03:27 -0700996#if ENABLE_CHANNEL_CONVERSION
997 // Determine whether channel conversion is required.
Eric Laurentdd45fd32014-07-01 20:32:28 -0700998 const uint32_t input_channels = audio_channel_count_from_in_mask(
Stewart Milese54c12c2014-05-01 09:03:27 -0700999 rsxadev->config.input_channel_mask);
Eric Laurentdd45fd32014-07-01 20:32:28 -07001000 const uint32_t output_channels = audio_channel_count_from_out_mask(
Stewart Milese54c12c2014-05-01 09:03:27 -07001001 rsxadev->config.output_channel_mask);
1002 if (input_channels != output_channels) {
1003 SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d "
1004 "input channels", output_channels, input_channels);
1005 // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono.
1006 ALOG_ASSERT(rsxadev->config.common.format == AUDIO_FORMAT_PCM_16_BIT);
1007 ALOG_ASSERT((input_channels == 1 && output_channels == 2) ||
1008 (input_channels == 2 && output_channels == 1));
1009 }
1010#endif // ENABLE_CHANNEL_CONVERSION
1011
Stewart Miles02c2f712014-05-01 09:03:27 -07001012#if ENABLE_RESAMPLING
1013 const uint32_t input_sample_rate = in_get_sample_rate(&stream->common);
1014 const uint32_t output_sample_rate = rsxadev->config.output_sample_rate;
1015 const size_t resampler_buffer_size_frames =
1016 sizeof(rsxadev->resampler_buffer) / sizeof(rsxadev->resampler_buffer[0]);
1017 float resampler_ratio = 1.0f;
1018 // Determine whether resampling is required.
1019 if (input_sample_rate != output_sample_rate) {
1020 resampler_ratio = (float)output_sample_rate / (float)input_sample_rate;
1021 // Only support 16-bit PCM mono resampling.
1022 // NOTE: Resampling is performed after the channel conversion step.
1023 ALOG_ASSERT(rsxadev->config.common.format == AUDIO_FORMAT_PCM_16_BIT);
Eric Laurentdd45fd32014-07-01 20:32:28 -07001024 ALOG_ASSERT(audio_channel_count_from_in_mask(rsxadev->config.input_channel_mask) == 1);
Stewart Miles02c2f712014-05-01 09:03:27 -07001025 }
1026#endif // ENABLE_RESAMPLING
1027
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -08001028 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
Stewart Miles02c2f712014-05-01 09:03:27 -07001029 ssize_t frames_read = -1977;
Stewart Milese54c12c2014-05-01 09:03:27 -07001030 size_t read_frames = remaining_frames;
Stewart Miles02c2f712014-05-01 09:03:27 -07001031#if ENABLE_RESAMPLING
1032 char* const saved_buff = buff;
1033 if (resampler_ratio != 1.0f) {
1034 // Calculate the number of frames from the pipe that need to be read to generate
1035 // the data for the input stream read.
1036 const size_t frames_required_for_resampler = (size_t)(
1037 (float)read_frames * (float)resampler_ratio);
1038 read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames);
1039 // Read into the resampler buffer.
1040 buff = (char*)rsxadev->resampler_buffer;
1041 }
1042#endif // ENABLE_RESAMPLING
Stewart Milese54c12c2014-05-01 09:03:27 -07001043#if ENABLE_CHANNEL_CONVERSION
1044 if (output_channels == 1 && input_channels == 2) {
1045 // Need to read half the requested frames since the converted output
1046 // data will take twice the space (mono->stereo).
1047 read_frames /= 2;
1048 }
1049#endif // ENABLE_CHANNEL_CONVERSION
1050
1051 SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead());
1052
1053 frames_read = source->read(buff, read_frames, AudioBufferProvider::kInvalidPTS);
1054
1055 SUBMIX_ALOGV("in_read(): frames read %zd", frames_read);
1056
1057#if ENABLE_CHANNEL_CONVERSION
1058 // Perform in-place channel conversion.
1059 // NOTE: In the following "input stream" refers to the data returned by this function
1060 // and "output stream" refers to the data read from the pipe.
1061 if (input_channels != output_channels && frames_read > 0) {
1062 int16_t *data = (int16_t*)buff;
1063 if (output_channels == 2 && input_channels == 1) {
1064 // Offset into the output stream data in samples.
1065 ssize_t output_stream_offset = 0;
1066 for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read;
1067 input_stream_frame++, output_stream_offset += 2) {
1068 // Average the content from both channels.
1069 data[input_stream_frame] = ((int32_t)data[output_stream_offset] +
1070 (int32_t)data[output_stream_offset + 1]) / 2;
1071 }
1072 } else if (output_channels == 1 && input_channels == 2) {
1073 // Offset into the input stream data in samples.
1074 ssize_t input_stream_offset = (frames_read - 1) * 2;
1075 for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0;
1076 output_stream_frame--, input_stream_offset -= 2) {
1077 const short sample = data[output_stream_frame];
1078 data[input_stream_offset] = sample;
1079 data[input_stream_offset + 1] = sample;
1080 }
1081 }
1082 }
1083#endif // ENABLE_CHANNEL_CONVERSION
Stewart Miles3dd36f92014-05-01 09:03:27 -07001084
Stewart Miles02c2f712014-05-01 09:03:27 -07001085#if ENABLE_RESAMPLING
1086 if (resampler_ratio != 1.0f) {
1087 SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read);
1088 const int16_t * const data = (int16_t*)buff;
1089 int16_t * const resampled_buffer = (int16_t*)saved_buff;
1090 // Resample with *no* filtering - if the data from the ouptut stream was really
1091 // sampled at a different rate this will result in very nasty aliasing.
1092 const float output_stream_frames = (float)frames_read;
1093 size_t input_stream_frame = 0;
1094 for (float output_stream_frame = 0.0f;
1095 output_stream_frame < output_stream_frames &&
1096 input_stream_frame < remaining_frames;
1097 output_stream_frame += resampler_ratio, input_stream_frame++) {
1098 resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame];
1099 }
1100 ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames);
1101 SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame);
1102 frames_read = input_stream_frame;
1103 buff = saved_buff;
1104 }
1105#endif // ENABLE_RESAMPLING
1106
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -08001107 if (frames_read > 0) {
Stewart Miles92854f52014-05-01 09:03:27 -07001108#if LOG_STREAMS_TO_FILES
1109 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size);
1110#endif // LOG_STREAMS_TO_FILES
1111
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -08001112 remaining_frames -= frames_read;
1113 buff += frames_read * frame_size;
Stewart Milesc049a0a2014-05-01 09:03:27 -07001114 SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu",
1115 attempts, frames_read, remaining_frames);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -08001116 } else {
Stewart Miles3dd36f92014-05-01 09:03:27 -07001117 attempts++;
Stewart Milesc049a0a2014-05-01 09:03:27 -07001118 SUBMIX_ALOGE(" in_read read returned %zd", frames_read);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -08001119 usleep(READ_ATTEMPT_SLEEP_MS * 1000);
1120 }
1121 }
1122 // done using the source
Stewart Milesf645c5e2014-05-01 09:03:27 -07001123 pthread_mutex_lock(&rsxadev->lock);
Jean-Michel Trivieafbfa42012-12-18 11:30:33 -08001124 source.clear();
Stewart Milesf645c5e2014-05-01 09:03:27 -07001125 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001126 }
1127
Jean-Michel Trivi6acd9662012-09-11 19:19:08 -07001128 if (remaining_frames > 0) {
Stewart Miles3dd36f92014-05-01 09:03:27 -07001129 const size_t remaining_bytes = remaining_frames * frame_size;
Stewart Miles10f1a802014-06-09 20:54:37 -07001130 SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames);
Stewart Miles3dd36f92014-05-01 09:03:27 -07001131 memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes);
Jean-Michel Trivi6acd9662012-09-11 19:19:08 -07001132 }
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001133
Jean-Michel Trivieec87702012-09-17 09:59:42 -07001134 // compute how much we need to sleep after reading the data by comparing the wall clock with
1135 // the projected time at which we should return.
1136 struct timespec time_after_read;// wall clock after reading from the pipe
1137 struct timespec record_duration;// observed record duration
1138 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read);
1139 const uint32_t sample_rate = in_get_sample_rate(&stream->common);
1140 if (rc == 0) {
1141 // for how long have we been recording?
1142 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec;
1143 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec;
1144 if (record_duration.tv_nsec < 0) {
1145 record_duration.tv_sec--;
1146 record_duration.tv_nsec += 1000000000;
1147 }
1148
Stewart Milesf645c5e2014-05-01 09:03:27 -07001149 // read_counter_frames contains the number of frames that have been read since the
1150 // beginning of recording (including this call): it's converted to usec and compared to
1151 // how long we've been recording for, which gives us how long we must wait to sync the
1152 // projected recording time, and the observed recording time.
Jean-Michel Trivieec87702012-09-17 09:59:42 -07001153 long projected_vs_observed_offset_us =
1154 ((int64_t)(in->read_counter_frames
1155 - (record_duration.tv_sec*sample_rate)))
1156 * 1000000 / sample_rate
1157 - (record_duration.tv_nsec / 1000);
1158
Stewart Milesc049a0a2014-05-01 09:03:27 -07001159 SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus",
Jean-Michel Trivieec87702012-09-17 09:59:42 -07001160 record_duration.tv_sec, record_duration.tv_nsec/1000000,
1161 projected_vs_observed_offset_us);
1162 if (projected_vs_observed_offset_us > 0) {
1163 usleep(projected_vs_observed_offset_us);
1164 }
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001165 }
Jean-Michel Trivieec87702012-09-17 09:59:42 -07001166
Stewart Milesc049a0a2014-05-01 09:03:27 -07001167 SUBMIX_ALOGV("in_read returns %zu", bytes);
Jean-Michel Trivieec87702012-09-17 09:59:42 -07001168 return bytes;
1169
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001170}
1171
1172static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1173{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001174 (void)stream;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001175 return 0;
1176}
1177
1178static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
1179{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001180 (void)stream;
1181 (void)effect;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001182 return 0;
1183}
1184
1185static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
1186{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001187 (void)stream;
1188 (void)effect;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001189 return 0;
1190}
1191
1192static int adev_open_output_stream(struct audio_hw_device *dev,
1193 audio_io_handle_t handle,
1194 audio_devices_t devices,
1195 audio_output_flags_t flags,
1196 struct audio_config *config,
Eric Laurentf5e24692014-07-27 16:14:57 -07001197 struct audio_stream_out **stream_out,
1198 const char *address __unused)
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001199{
Stewart Milesf645c5e2014-05-01 09:03:27 -07001200 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev);
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001201 ALOGD("adev_open_output_stream()");
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001202 struct submix_stream_out *out;
Stewart Miles10f1a802014-06-09 20:54:37 -07001203 bool force_pipe_creation = false;
Stewart Milesc049a0a2014-05-01 09:03:27 -07001204 (void)handle;
1205 (void)devices;
1206 (void)flags;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001207
Stewart Miles3dd36f92014-05-01 09:03:27 -07001208 *stream_out = NULL;
1209
Stewart Miles70726842014-05-01 09:03:27 -07001210 // Make sure it's possible to open the device given the current audio config.
1211 submix_sanitize_config(config, false);
1212 if (!submix_open_validate(rsxadev, &rsxadev->lock, config, false)) {
1213 ALOGE("adev_open_output_stream(): Unable to open output stream.");
1214 return -EINVAL;
1215 }
1216
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001217 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out));
Stewart Miles3dd36f92014-05-01 09:03:27 -07001218 if (!out) return -ENOMEM;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001219
Stewart Miles568e66f2014-05-01 09:03:27 -07001220 // Initialize the function pointer tables (v-tables).
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001221 out->stream.common.get_sample_rate = out_get_sample_rate;
1222 out->stream.common.set_sample_rate = out_set_sample_rate;
1223 out->stream.common.get_buffer_size = out_get_buffer_size;
1224 out->stream.common.get_channels = out_get_channels;
1225 out->stream.common.get_format = out_get_format;
1226 out->stream.common.set_format = out_set_format;
1227 out->stream.common.standby = out_standby;
1228 out->stream.common.dump = out_dump;
1229 out->stream.common.set_parameters = out_set_parameters;
1230 out->stream.common.get_parameters = out_get_parameters;
1231 out->stream.common.add_audio_effect = out_add_audio_effect;
1232 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1233 out->stream.get_latency = out_get_latency;
1234 out->stream.set_volume = out_set_volume;
1235 out->stream.write = out_write;
1236 out->stream.get_render_position = out_get_render_position;
1237 out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
1238
Stewart Miles10f1a802014-06-09 20:54:37 -07001239#if ENABLE_RESAMPLING
1240 // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits
1241 // writes correctly.
1242 force_pipe_creation = rsxadev->config.common.sample_rate != config->sample_rate;
1243#endif // ENABLE_RESAMPLING
1244
1245 // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so
1246 // that it's recreated.
Stewart Miles3dd36f92014-05-01 09:03:27 -07001247 pthread_mutex_lock(&rsxadev->lock);
Stewart Miles10f1a802014-06-09 20:54:37 -07001248 if ((rsxadev->rsxSink != NULL && rsxadev->rsxSink->isShutdown()) || force_pipe_creation) {
Stewart Miles3dd36f92014-05-01 09:03:27 -07001249 submix_audio_device_release_pipe(rsxadev);
1250 }
1251 pthread_mutex_unlock(&rsxadev->lock);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001252
Stewart Miles568e66f2014-05-01 09:03:27 -07001253 // Store a pointer to the device from the output stream.
1254 out->dev = rsxadev;
Stewart Miles3dd36f92014-05-01 09:03:27 -07001255 // Initialize the pipe.
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001256 ALOGV("adev_open_output_stream(): about to create pipe");
Stewart Miles3dd36f92014-05-01 09:03:27 -07001257 submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES,
1258 DEFAULT_PIPE_PERIOD_COUNT, NULL, out);
Stewart Miles92854f52014-05-01 09:03:27 -07001259#if LOG_STREAMS_TO_FILES
1260 out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY,
1261 LOG_STREAM_FILE_PERMISSIONS);
1262 ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s",
1263 strerror(errno));
1264 ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd);
1265#endif // LOG_STREAMS_TO_FILES
Stewart Miles568e66f2014-05-01 09:03:27 -07001266 // Return the output stream.
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001267 *stream_out = &out->stream;
1268
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001269 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001270}
1271
1272static void adev_close_output_stream(struct audio_hw_device *dev,
1273 struct audio_stream_out *stream)
1274{
Stewart Miles3dd36f92014-05-01 09:03:27 -07001275 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream);
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001276 ALOGD("adev_close_output_stream()");
Stewart Miles3dd36f92014-05-01 09:03:27 -07001277 submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), NULL, out);
Stewart Miles92854f52014-05-01 09:03:27 -07001278#if LOG_STREAMS_TO_FILES
1279 if (out->log_fd >= 0) close(out->log_fd);
1280#endif // LOG_STREAMS_TO_FILES
Stewart Miles3dd36f92014-05-01 09:03:27 -07001281 free(out);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001282}
1283
1284static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
1285{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001286 (void)dev;
1287 (void)kvpairs;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001288 return -ENOSYS;
1289}
1290
1291static char * adev_get_parameters(const struct audio_hw_device *dev,
1292 const char *keys)
1293{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001294 (void)dev;
1295 (void)keys;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001296 return strdup("");;
1297}
1298
1299static int adev_init_check(const struct audio_hw_device *dev)
1300{
1301 ALOGI("adev_init_check()");
Stewart Milesc049a0a2014-05-01 09:03:27 -07001302 (void)dev;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001303 return 0;
1304}
1305
1306static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
1307{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001308 (void)dev;
1309 (void)volume;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001310 return -ENOSYS;
1311}
1312
1313static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
1314{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001315 (void)dev;
1316 (void)volume;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001317 return -ENOSYS;
1318}
1319
1320static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
1321{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001322 (void)dev;
1323 (void)volume;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001324 return -ENOSYS;
1325}
1326
1327static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
1328{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001329 (void)dev;
1330 (void)muted;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001331 return -ENOSYS;
1332}
1333
1334static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
1335{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001336 (void)dev;
1337 (void)muted;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001338 return -ENOSYS;
1339}
1340
1341static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
1342{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001343 (void)dev;
1344 (void)mode;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001345 return 0;
1346}
1347
1348static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
1349{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001350 (void)dev;
1351 (void)state;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001352 return -ENOSYS;
1353}
1354
1355static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1356{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001357 (void)dev;
1358 (void)state;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001359 return -ENOSYS;
1360}
1361
1362static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
1363 const struct audio_config *config)
1364{
Stewart Miles568e66f2014-05-01 09:03:27 -07001365 if (audio_is_linear_pcm(config->format)) {
1366 const size_t buffer_period_size_frames =
1367 audio_hw_device_get_submix_audio_device(const_cast<struct audio_hw_device*>(dev))->
Stewart Miles3dd36f92014-05-01 09:03:27 -07001368 config.buffer_period_size_frames;
Eric Laurentdd45fd32014-07-01 20:32:28 -07001369 const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) *
Stewart Miles568e66f2014-05-01 09:03:27 -07001370 audio_bytes_per_sample(config->format);
1371 const size_t buffer_size = buffer_period_size_frames * frame_size_in_bytes;
Stewart Miles10f1a802014-06-09 20:54:37 -07001372 SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames",
Stewart Miles568e66f2014-05-01 09:03:27 -07001373 buffer_size, buffer_period_size_frames);
1374 return buffer_size;
1375 }
1376 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001377}
1378
1379static int adev_open_input_stream(struct audio_hw_device *dev,
1380 audio_io_handle_t handle,
1381 audio_devices_t devices,
1382 struct audio_config *config,
Glenn Kasten7d973ad2014-07-15 11:10:38 -07001383 struct audio_stream_in **stream_in,
Eric Laurentf5e24692014-07-27 16:14:57 -07001384 audio_input_flags_t flags __unused,
1385 const char *address __unused,
1386 audio_source_t source __unused)
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001387{
Stewart Milesf645c5e2014-05-01 09:03:27 -07001388 struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev);
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001389 struct submix_stream_in *in;
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001390 ALOGD("adev_open_input_stream()");
Stewart Milesc049a0a2014-05-01 09:03:27 -07001391 (void)handle;
1392 (void)devices;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001393
Stewart Miles3dd36f92014-05-01 09:03:27 -07001394 *stream_in = NULL;
1395
Stewart Miles70726842014-05-01 09:03:27 -07001396 // Make sure it's possible to open the device given the current audio config.
1397 submix_sanitize_config(config, true);
1398 if (!submix_open_validate(rsxadev, &rsxadev->lock, config, true)) {
1399 ALOGE("adev_open_input_stream(): Unable to open input stream.");
1400 return -EINVAL;
1401 }
1402
Stewart Miles3dd36f92014-05-01 09:03:27 -07001403#if ENABLE_LEGACY_INPUT_OPEN
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001404 pthread_mutex_lock(&rsxadev->lock);
Stewart Miles3dd36f92014-05-01 09:03:27 -07001405 in = rsxadev->input;
1406 if (in) {
1407 in->ref_count++;
1408 sp<MonoPipe> sink = rsxadev->rsxSink;
1409 ALOG_ASSERT(sink != NULL);
1410 // If the sink has been shutdown, delete the pipe.
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001411 if (sink != NULL) {
1412 if (sink->isShutdown()) {
1413 ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d",
1414 in->ref_count);
1415 submix_audio_device_release_pipe(rsxadev);
1416 } else {
1417 ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count);
1418 }
1419 } else {
1420 ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count);
1421 }
Stewart Miles3dd36f92014-05-01 09:03:27 -07001422 }
1423 pthread_mutex_unlock(&rsxadev->lock);
1424#else
1425 in = NULL;
1426#endif // ENABLE_LEGACY_INPUT_OPEN
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001427
Stewart Miles3dd36f92014-05-01 09:03:27 -07001428 if (!in) {
1429 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in));
1430 if (!in) return -ENOMEM;
1431 in->ref_count = 1;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001432
Stewart Miles3dd36f92014-05-01 09:03:27 -07001433 // Initialize the function pointer tables (v-tables).
1434 in->stream.common.get_sample_rate = in_get_sample_rate;
1435 in->stream.common.set_sample_rate = in_set_sample_rate;
1436 in->stream.common.get_buffer_size = in_get_buffer_size;
1437 in->stream.common.get_channels = in_get_channels;
1438 in->stream.common.get_format = in_get_format;
1439 in->stream.common.set_format = in_set_format;
1440 in->stream.common.standby = in_standby;
1441 in->stream.common.dump = in_dump;
1442 in->stream.common.set_parameters = in_set_parameters;
1443 in->stream.common.get_parameters = in_get_parameters;
1444 in->stream.common.add_audio_effect = in_add_audio_effect;
1445 in->stream.common.remove_audio_effect = in_remove_audio_effect;
1446 in->stream.set_gain = in_set_gain;
1447 in->stream.read = in_read;
1448 in->stream.get_input_frames_lost = in_get_input_frames_lost;
1449 }
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001450
Stewart Miles568e66f2014-05-01 09:03:27 -07001451 // Initialize the input stream.
Jean-Michel Trivieec87702012-09-17 09:59:42 -07001452 in->read_counter_frames = 0;
1453 in->output_standby = rsxadev->output_standby;
Stewart Miles3dd36f92014-05-01 09:03:27 -07001454 in->dev = rsxadev;
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001455 in->read_error_count = 0;
Stewart Miles3dd36f92014-05-01 09:03:27 -07001456 // Initialize the pipe.
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001457 ALOGV("adev_open_input_stream(): about to create pipe");
Stewart Miles3dd36f92014-05-01 09:03:27 -07001458 submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES,
1459 DEFAULT_PIPE_PERIOD_COUNT, in, NULL);
Stewart Miles92854f52014-05-01 09:03:27 -07001460#if LOG_STREAMS_TO_FILES
1461 in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY,
1462 LOG_STREAM_FILE_PERMISSIONS);
1463 ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s",
1464 strerror(errno));
1465 ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd);
1466#endif // LOG_STREAMS_TO_FILES
Stewart Miles3dd36f92014-05-01 09:03:27 -07001467 // Return the input stream.
1468 *stream_in = &in->stream;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001469
1470 return 0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001471}
1472
1473static void adev_close_input_stream(struct audio_hw_device *dev,
Stewart Milesc049a0a2014-05-01 09:03:27 -07001474 struct audio_stream_in *stream)
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001475{
Stewart Miles3dd36f92014-05-01 09:03:27 -07001476 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream);
Jean-Michel Trivi793a8542014-10-14 15:31:51 -07001477 ALOGD("adev_close_input_stream()");
Stewart Miles3dd36f92014-05-01 09:03:27 -07001478 submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), in, NULL);
Stewart Miles92854f52014-05-01 09:03:27 -07001479#if LOG_STREAMS_TO_FILES
1480 if (in->log_fd >= 0) close(in->log_fd);
1481#endif // LOG_STREAMS_TO_FILES
Stewart Miles3dd36f92014-05-01 09:03:27 -07001482#if ENABLE_LEGACY_INPUT_OPEN
1483 if (in->ref_count == 0) free(in);
1484#else
1485 free(in);
1486#endif // ENABLE_LEGACY_INPUT_OPEN
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001487}
1488
1489static int adev_dump(const audio_hw_device_t *device, int fd)
1490{
Stewart Milesc049a0a2014-05-01 09:03:27 -07001491 (void)device;
1492 (void)fd;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001493 return 0;
1494}
1495
1496static int adev_close(hw_device_t *device)
1497{
1498 ALOGI("adev_close()");
1499 free(device);
1500 return 0;
1501}
1502
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001503static int adev_open(const hw_module_t* module, const char* name,
1504 hw_device_t** device)
1505{
1506 ALOGI("adev_open(name=%s)", name);
1507 struct submix_audio_device *rsxadev;
1508
1509 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
1510 return -EINVAL;
1511
1512 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device));
1513 if (!rsxadev)
1514 return -ENOMEM;
1515
1516 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG;
Eric Laurent5d85c532012-09-10 10:36:09 -07001517 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001518 rsxadev->device.common.module = (struct hw_module_t *) module;
1519 rsxadev->device.common.close = adev_close;
1520
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001521 rsxadev->device.init_check = adev_init_check;
1522 rsxadev->device.set_voice_volume = adev_set_voice_volume;
1523 rsxadev->device.set_master_volume = adev_set_master_volume;
1524 rsxadev->device.get_master_volume = adev_get_master_volume;
1525 rsxadev->device.set_master_mute = adev_set_master_mute;
1526 rsxadev->device.get_master_mute = adev_get_master_mute;
1527 rsxadev->device.set_mode = adev_set_mode;
1528 rsxadev->device.set_mic_mute = adev_set_mic_mute;
1529 rsxadev->device.get_mic_mute = adev_get_mic_mute;
1530 rsxadev->device.set_parameters = adev_set_parameters;
1531 rsxadev->device.get_parameters = adev_get_parameters;
1532 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size;
1533 rsxadev->device.open_output_stream = adev_open_output_stream;
1534 rsxadev->device.close_output_stream = adev_close_output_stream;
1535 rsxadev->device.open_input_stream = adev_open_input_stream;
1536 rsxadev->device.close_input_stream = adev_close_input_stream;
1537 rsxadev->device.dump = adev_dump;
1538
Jean-Michel Trivieec87702012-09-17 09:59:42 -07001539 rsxadev->input_standby = true;
1540 rsxadev->output_standby = true;
1541
Jean-Michel Trivi88b79cb2012-08-16 13:56:03 -07001542 *device = &rsxadev->device.common;
1543
1544 return 0;
1545}
1546
1547static struct hw_module_methods_t hal_module_methods = {
1548 /* open */ adev_open,
1549};
1550
1551struct audio_module HAL_MODULE_INFO_SYM = {
1552 /* common */ {
1553 /* tag */ HARDWARE_MODULE_TAG,
1554 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1,
1555 /* hal_api_version */ HARDWARE_HAL_API_VERSION,
1556 /* id */ AUDIO_HARDWARE_MODULE_ID,
1557 /* name */ "Wifi Display audio HAL",
1558 /* author */ "The Android Open Source Project",
1559 /* methods */ &hal_module_methods,
1560 /* dso */ NULL,
1561 /* reserved */ { 0 },
1562 },
1563};
1564
1565} //namespace android
1566
1567} //extern "C"