2 * Copyright (C) 2012 The Android Open Source Project
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
8 * http://www.apache.org/licenses/LICENSE-2.0
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.
17 #define LOG_TAG "usb_audio_hw"
18 /*#define LOG_NDEBUG 0*/
28 #include <cutils/str_parms.h>
29 #include <cutils/properties.h>
31 #include <hardware/audio.h>
32 #include <hardware/audio_alsaops.h>
33 #include <hardware/hardware.h>
35 #include <system/audio.h>
37 #include <tinyalsa/asoundlib.h>
39 #include <audio_utils/channels.h>
42 * Set k_force_channels to force the number of channels to present to AudioFlinger.
43 * 0 disables (this is default: present the device channels to AudioFlinger).
44 * 2 forces to legacy stereo mode.
46 * Others values can be tried (up to 8).
47 * TODO: AudioFlinger cannot support more than 8 active output channels
48 * at this time, so limiting logic needs to be put here or communicated from above.
50 static const unsigned k_force_channels = 0;
52 #include "alsa_device_profile.h"
53 #include "alsa_device_proxy.h"
56 #define DEFAULT_INPUT_BUFFER_SIZE_MS 20
59 struct audio_hw_device hw_device;
61 pthread_mutex_t lock; /* see note below on mutex acquisition order */
64 alsa_device_profile out_profile;
67 alsa_device_profile in_profile;
73 struct audio_stream_out stream;
75 pthread_mutex_t lock; /* see note below on mutex acquisition order */
78 struct audio_device *dev; /* hardware information - only using this for the lock */
80 alsa_device_profile * profile;
81 alsa_device_proxy proxy; /* state of the stream */
83 unsigned hal_channel_count; /* channel count exposed to AudioFlinger.
84 * This may differ from the device channel count when
85 * the device is not compatible with AudioFlinger
86 * capabilities, e.g. exposes too many channels or
87 * too few channels. */
88 void * conversion_buffer; /* any conversions are put into here
89 * they could come from here too if
90 * there was a previous conversion */
91 size_t conversion_buffer_size; /* in bytes */
95 struct audio_stream_in stream;
97 pthread_mutex_t lock; /* see note below on mutex acquisition order */
100 struct audio_device *dev; /* hardware information - only using this for the lock */
102 alsa_device_profile * profile;
103 alsa_device_proxy proxy; /* state of the stream */
106 // struct audio_config hal_pcm_config;
108 /* We may need to read more data from the device in order to data reduce to 16bit, 4chan */
109 void * conversion_buffer; /* any conversions are put into here
110 * they could come from here too if
111 * there was a previous conversion */
112 size_t conversion_buffer_size; /* in bytes */
119 * Convert a buffer of packed (3-byte) PCM24LE samples to PCM16LE samples.
120 * in_buff points to the buffer of PCM24LE samples
121 * num_in_samples size of input buffer in SAMPLES
122 * out_buff points to the buffer to receive converted PCM16LE LE samples.
124 * the number of BYTES of output data.
125 * We are doing this since we *always* present to The Framework as A PCM16LE device, but need to
126 * support PCM24_3LE (24-bit, packed).
128 * This conversion is safe to do in-place (in_buff == out_buff).
129 * TODO Move this to a utilities module.
131 static size_t convert_24_3_to_16(const unsigned char * in_buff, size_t num_in_samples,
135 * Move from front to back so that the conversion can be done in-place
136 * i.e. in_buff == out_buff
138 /* we need 2 bytes in the output for every 3 bytes in the input */
139 unsigned char* dst_ptr = (unsigned char*)out_buff;
140 const unsigned char* src_ptr = in_buff;
141 size_t src_smpl_index;
142 for (src_smpl_index = 0; src_smpl_index < num_in_samples; src_smpl_index++) {
143 src_ptr++; /* lowest-(skip)-byte */
144 *dst_ptr++ = *src_ptr++; /* low-byte */
145 *dst_ptr++ = *src_ptr++; /* high-byte */
148 /* return number of *bytes* generated: */
149 return num_in_samples * 2;
153 * Convert a buffer of packed (3-byte) PCM32 samples to PCM16LE samples.
154 * in_buff points to the buffer of PCM32 samples
155 * num_in_samples size of input buffer in SAMPLES
156 * out_buff points to the buffer to receive converted PCM16LE LE samples.
158 * the number of BYTES of output data.
159 * We are doing this since we *always* present to The Framework as A PCM16LE device, but need to
160 * support PCM_FORMAT_S32_LE (32-bit).
162 * This conversion is safe to do in-place (in_buff == out_buff).
163 * TODO Move this to a utilities module.
165 static size_t convert_32_to_16(const int32_t * in_buff, size_t num_in_samples, short * out_buff)
168 * Move from front to back so that the conversion can be done in-place
169 * i.e. in_buff == out_buff
172 short * dst_ptr = out_buff;
173 const int32_t* src_ptr = in_buff;
174 size_t src_smpl_index;
175 for (src_smpl_index = 0; src_smpl_index < num_in_samples; src_smpl_index++) {
176 *dst_ptr++ = *src_ptr++ >> 16;
179 /* return number of *bytes* generated: */
180 return num_in_samples * 2;
183 static char * device_get_parameters(alsa_device_profile * profile, const char * keys)
185 ALOGV("usb:audio_hw::device_get_parameters() keys:%s", keys);
187 if (profile->card < 0 || profile->device < 0) {
191 struct str_parms *query = str_parms_create_str(keys);
192 struct str_parms *result = str_parms_create();
194 /* These keys are from hardware/libhardware/include/audio.h */
195 /* supported sample rates */
196 if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
197 char* rates_list = profile_get_sample_rate_strs(profile);
198 str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
203 /* supported channel counts */
204 if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
205 char* channels_list = profile_get_channel_count_strs(profile);
206 str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_CHANNELS,
211 /* supported sample formats */
212 if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
213 char * format_params = profile_get_format_strs(profile);
214 str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS,
218 str_parms_destroy(query);
220 char* result_str = str_parms_to_str(result);
221 str_parms_destroy(result);
223 ALOGV("usb:audio_hw::device_get_parameters = %s", result_str);
232 * NOTE: when multiple mutexes have to be acquired, always respect the
233 * following order: hw device > out stream
239 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
241 uint32_t rate = proxy_get_sample_rate(&((struct stream_out*)stream)->proxy);
242 ALOGV("out_get_sample_rate() = %d", rate);
246 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
251 static size_t out_get_buffer_size(const struct audio_stream *stream)
253 const struct stream_out* out = (const struct stream_out*)stream;
255 proxy_get_period_size(&out->proxy) * audio_stream_out_frame_size(&(out->stream));
256 ALOGV("out_get_buffer_size() = %zu", buffer_size);
260 static uint32_t out_get_channels(const struct audio_stream *stream)
262 const struct stream_out *out = (const struct stream_out*)stream;
263 return audio_channel_out_mask_from_count(out->hal_channel_count);
266 static audio_format_t out_get_format(const struct audio_stream *stream)
268 /* Note: The HAL doesn't do any FORMAT conversion at this time. It
269 * Relies on the framework to provide data in the specified format.
270 * This could change in the future.
272 alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
273 audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
274 ALOGV("out_get_format() = %d", format);
278 static int out_set_format(struct audio_stream *stream, audio_format_t format)
283 static int out_standby(struct audio_stream *stream)
285 struct stream_out *out = (struct stream_out *)stream;
287 pthread_mutex_lock(&out->dev->lock);
288 pthread_mutex_lock(&out->lock);
291 proxy_close(&out->proxy);
295 pthread_mutex_unlock(&out->lock);
296 pthread_mutex_unlock(&out->dev->lock);
301 static int out_dump(const struct audio_stream *stream, int fd)
306 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
308 ALOGV("usb:audio_hw::out out_set_parameters() keys:%s", kvpairs);
310 struct stream_out *out = (struct stream_out *)stream;
319 struct str_parms * parms = str_parms_create_str(kvpairs);
320 pthread_mutex_lock(&out->dev->lock);
321 pthread_mutex_lock(&out->lock);
323 param_val = str_parms_get_str(parms, "card", value, sizeof(value));
327 param_val = str_parms_get_str(parms, "device", value, sizeof(value));
329 device = atoi(value);
331 if ((card >= 0) && (card != out->profile->card) &&
332 (device >= 0) && (device != out->profile->device)) {
333 /* cannot read pcm device info if playback is active */
337 int saved_card = out->profile->card;
338 int saved_device = out->profile->device;
339 out->profile->card = card;
340 out->profile->device = device;
341 ret_value = profile_read_device_info(out->profile) ? 0 : -EINVAL;
342 if (ret_value != 0) {
343 out->profile->card = saved_card;
344 out->profile->device = saved_device;
348 pthread_mutex_unlock(&out->lock);
349 pthread_mutex_unlock(&out->dev->lock);
350 str_parms_destroy(parms);
355 static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
357 struct stream_out *out = (struct stream_out *)stream;
358 pthread_mutex_lock(&out->dev->lock);
359 pthread_mutex_lock(&out->lock);
361 char * params_str = device_get_parameters(out->profile, keys);
363 pthread_mutex_unlock(&out->lock);
364 pthread_mutex_unlock(&out->dev->lock);
369 static uint32_t out_get_latency(const struct audio_stream_out *stream)
371 alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
372 return proxy_get_latency(proxy);
375 static int out_set_volume(struct audio_stream_out *stream, float left, float right)
380 /* must be called with hw device and output stream mutexes locked */
381 static int start_output_stream(struct stream_out *out)
383 ALOGV("usb:audio_hw::out start_output_stream(card:%d device:%d)",
384 out->profile->card, out->profile->device);
386 return proxy_open(&out->proxy);
389 static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
392 struct stream_out *out = (struct stream_out *)stream;
394 pthread_mutex_lock(&out->dev->lock);
395 pthread_mutex_lock(&out->lock);
397 ret = start_output_stream(out);
399 pthread_mutex_unlock(&out->dev->lock);
402 out->standby = false;
404 pthread_mutex_unlock(&out->dev->lock);
407 alsa_device_proxy* proxy = &out->proxy;
408 const void * write_buff = buffer;
409 int num_write_buff_bytes = bytes;
410 const int num_device_channels = proxy_get_channel_count(proxy); /* what we told alsa */
411 const int num_req_channels = out->hal_channel_count; /* what we told AudioFlinger */
412 if (num_device_channels != num_req_channels) {
413 /* allocate buffer */
414 const size_t required_conversion_buffer_size =
415 bytes * num_device_channels / num_req_channels;
416 if (required_conversion_buffer_size > out->conversion_buffer_size) {
417 out->conversion_buffer_size = required_conversion_buffer_size;
418 out->conversion_buffer = realloc(out->conversion_buffer,
419 out->conversion_buffer_size);
422 const audio_format_t audio_format = out_get_format(&(out->stream.common));
423 const unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
424 num_write_buff_bytes =
425 adjust_channels(write_buff, num_req_channels,
426 out->conversion_buffer, num_device_channels,
427 sample_size_in_bytes, num_write_buff_bytes);
428 write_buff = out->conversion_buffer;
431 if (write_buff != NULL && num_write_buff_bytes != 0) {
432 proxy_write(&out->proxy, write_buff, num_write_buff_bytes);
435 pthread_mutex_unlock(&out->lock);
440 pthread_mutex_unlock(&out->lock);
442 usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
443 out_get_sample_rate(&stream->common));
449 static int out_get_render_position(const struct audio_stream_out *stream, uint32_t *dsp_frames)
454 static int out_get_presentation_position(const struct audio_stream_out *stream,
455 uint64_t *frames, struct timespec *timestamp)
457 /* FIXME - This needs to be implemented */
461 static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
466 static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
471 static int out_get_next_write_timestamp(const struct audio_stream_out *stream, int64_t *timestamp)
476 static int adev_open_output_stream(struct audio_hw_device *dev,
477 audio_io_handle_t handle,
478 audio_devices_t devices,
479 audio_output_flags_t flags,
480 struct audio_config *config,
481 struct audio_stream_out **stream_out,
482 const char *address __unused)
484 ALOGV("usb:audio_hw::out adev_open_output_stream() handle:0x%X, device:0x%X, flags:0x%X",
485 handle, devices, flags);
487 struct audio_device *adev = (struct audio_device *)dev;
489 struct stream_out *out;
491 out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
495 /* setup function pointers */
496 out->stream.common.get_sample_rate = out_get_sample_rate;
497 out->stream.common.set_sample_rate = out_set_sample_rate;
498 out->stream.common.get_buffer_size = out_get_buffer_size;
499 out->stream.common.get_channels = out_get_channels;
500 out->stream.common.get_format = out_get_format;
501 out->stream.common.set_format = out_set_format;
502 out->stream.common.standby = out_standby;
503 out->stream.common.dump = out_dump;
504 out->stream.common.set_parameters = out_set_parameters;
505 out->stream.common.get_parameters = out_get_parameters;
506 out->stream.common.add_audio_effect = out_add_audio_effect;
507 out->stream.common.remove_audio_effect = out_remove_audio_effect;
508 out->stream.get_latency = out_get_latency;
509 out->stream.set_volume = out_set_volume;
510 out->stream.write = out_write;
511 out->stream.get_render_position = out_get_render_position;
512 out->stream.get_presentation_position = out_get_presentation_position;
513 out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
517 out->profile = &adev->out_profile;
519 // build this to hand to the alsa_device_proxy
520 struct pcm_config proxy_config;
525 if (config->sample_rate == 0) {
526 proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
527 } else if (profile_is_sample_rate_valid(out->profile, config->sample_rate)) {
528 proxy_config.rate = config->sample_rate;
530 proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
535 if (config->format == AUDIO_FORMAT_DEFAULT) {
536 proxy_config.format = profile_get_default_format(out->profile);
537 config->format = audio_format_from_pcm_format(proxy_config.format);
539 enum pcm_format fmt = pcm_format_from_audio_format(config->format);
540 if (profile_is_format_valid(out->profile, fmt)) {
541 proxy_config.format = fmt;
543 proxy_config.format = profile_get_default_format(out->profile);
544 config->format = audio_format_from_pcm_format(proxy_config.format);
550 unsigned proposed_channel_count = profile_get_default_channel_count(out->profile);
551 if (k_force_channels) {
552 proposed_channel_count = k_force_channels;
553 } else if (config->channel_mask != AUDIO_CHANNEL_NONE) {
554 proposed_channel_count = audio_channel_count_from_out_mask(config->channel_mask);
556 /* we can expose any channel count mask, and emulate internally. */
557 config->channel_mask = audio_channel_out_mask_from_count(proposed_channel_count);
558 out->hal_channel_count = proposed_channel_count;
559 /* no validity checks are needed as proxy_prepare() forces channel_count to be valid.
560 * and we emulate any channel count discrepancies in out_write(). */
561 proxy_config.channels = proposed_channel_count;
563 proxy_prepare(&out->proxy, out->profile, &proxy_config);
565 /* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
568 out->conversion_buffer = NULL;
569 out->conversion_buffer_size = 0;
573 *stream_out = &out->stream;
583 static void adev_close_output_stream(struct audio_hw_device *dev,
584 struct audio_stream_out *stream)
586 ALOGV("usb:audio_hw::out adev_close_output_stream()");
587 struct stream_out *out = (struct stream_out *)stream;
589 /* Close the pcm device */
590 out_standby(&stream->common);
592 free(out->conversion_buffer);
594 out->conversion_buffer = NULL;
595 out->conversion_buffer_size = 0;
600 static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
601 const struct audio_config *config)
603 /* TODO This needs to be calculated based on format/channels/rate */
610 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
612 uint32_t rate = proxy_get_sample_rate(&((const struct stream_in *)stream)->proxy);
613 ALOGV("in_get_sample_rate() = %d", rate);
617 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
619 ALOGV("in_set_sample_rate(%d) - NOPE", rate);
623 static size_t in_get_buffer_size(const struct audio_stream *stream)
625 const struct stream_in * in = ((const struct stream_in*)stream);
627 proxy_get_period_size(&in->proxy) * audio_stream_in_frame_size(&(in->stream));
628 ALOGV("in_get_buffer_size() = %zd", buffer_size);
633 static uint32_t in_get_channels(const struct audio_stream *stream)
635 /* TODO Here is the code we need when we support arbitrary channel counts
636 * alsa_device_proxy * proxy = ((struct stream_in*)stream)->proxy;
637 * unsigned channel_count = proxy_get_channel_count(proxy);
638 * uint32_t channel_mask = audio_channel_in_mask_from_count(channel_count);
639 * ALOGV("in_get_channels() = 0x%X count:%d", channel_mask, channel_count);
640 * return channel_mask;
642 /* TODO When AudioPolicyManager & AudioFlinger supports arbitrary channels
643 rewrite this to return the ACTUAL channel format */
644 return AUDIO_CHANNEL_IN_STEREO;
647 static audio_format_t in_get_format(const struct audio_stream *stream)
649 /* TODO Here is the code we need when we support arbitrary input formats
650 * alsa_device_proxy * proxy = ((struct stream_in*)stream)->proxy;
651 * audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
652 * ALOGV("in_get_format() = %d", format);
655 /* Input only supports PCM16 */
656 /* TODO When AudioPolicyManager & AudioFlinger supports arbitrary input formats
657 rewrite this to return the ACTUAL channel format (above) */
658 return AUDIO_FORMAT_PCM_16_BIT;
661 static int in_set_format(struct audio_stream *stream, audio_format_t format)
663 ALOGV("in_set_format(%d) - NOPE", format);
668 static int in_standby(struct audio_stream *stream)
670 struct stream_in *in = (struct stream_in *)stream;
672 pthread_mutex_lock(&in->dev->lock);
673 pthread_mutex_lock(&in->lock);
676 proxy_close(&in->proxy);
680 pthread_mutex_unlock(&in->lock);
681 pthread_mutex_unlock(&in->dev->lock);
686 static int in_dump(const struct audio_stream *stream, int fd)
691 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
693 ALOGV("usb: audio_hw::in in_set_parameters() keys:%s", kvpairs);
695 struct stream_in *in = (struct stream_in *)stream;
704 struct str_parms * parms = str_parms_create_str(kvpairs);
706 pthread_mutex_lock(&in->dev->lock);
707 pthread_mutex_lock(&in->lock);
710 param_val = str_parms_get_str(parms, "card", value, sizeof(value));
714 param_val = str_parms_get_str(parms, "device", value, sizeof(value));
716 device = atoi(value);
718 if ((card >= 0) && (card != in->profile->card) &&
719 (device >= 0) && (device != in->profile->device)) {
720 /* cannot read pcm device info if playback is active */
724 int saved_card = in->profile->card;
725 int saved_device = in->profile->device;
726 in->profile->card = card;
727 in->profile->device = device;
728 ret_value = profile_read_device_info(in->profile) ? 0 : -EINVAL;
729 if (ret_value != 0) {
730 in->profile->card = saved_card;
731 in->profile->device = saved_device;
736 pthread_mutex_unlock(&in->lock);
737 pthread_mutex_unlock(&in->dev->lock);
739 str_parms_destroy(parms);
744 static char * in_get_parameters(const struct audio_stream *stream, const char *keys)
746 struct stream_in *in = (struct stream_in *)stream;
748 pthread_mutex_lock(&in->dev->lock);
749 pthread_mutex_lock(&in->lock);
751 char * params_str = device_get_parameters(in->profile, keys);
753 pthread_mutex_unlock(&in->lock);
754 pthread_mutex_unlock(&in->dev->lock);
759 static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
764 static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
769 static int in_set_gain(struct audio_stream_in *stream, float gain)
774 /* must be called with hw device and output stream mutexes locked */
775 static int start_input_stream(struct stream_in *in)
777 ALOGV("usb:audio_hw::start_input_stream(card:%d device:%d)",
778 in->profile->card, in->profile->device);
780 return proxy_open(&in->proxy);
783 /* TODO mutex stuff here (see out_write) */
784 static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
786 size_t num_read_buff_bytes = 0;
787 void * read_buff = buffer;
788 void * out_buff = buffer;
790 struct stream_in * in = (struct stream_in *)stream;
792 pthread_mutex_lock(&in->dev->lock);
793 pthread_mutex_lock(&in->lock);
795 if (start_input_stream(in) != 0) {
796 pthread_mutex_unlock(&in->dev->lock);
801 pthread_mutex_unlock(&in->dev->lock);
804 alsa_device_profile * profile = in->profile;
807 * OK, we need to figure out how much data to read to be able to output the requested
808 * number of bytes in the HAL format (16-bit, stereo).
810 num_read_buff_bytes = bytes;
811 int num_device_channels = proxy_get_channel_count(&in->proxy);
812 int num_req_channels = 2; /* always, for now */
814 if (num_device_channels != num_req_channels) {
815 num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels;
818 enum pcm_format format = proxy_get_format(&in->proxy);
819 if (format == PCM_FORMAT_S24_3LE) {
820 /* 24-bit USB device */
821 num_read_buff_bytes = (3 * num_read_buff_bytes) / 2;
822 } else if (format == PCM_FORMAT_S32_LE) {
823 /* 32-bit USB device */
824 num_read_buff_bytes = num_read_buff_bytes * 2;
827 /* Setup/Realloc the conversion buffer (if necessary). */
828 if (num_read_buff_bytes != bytes) {
829 if (num_read_buff_bytes > in->conversion_buffer_size) {
830 /*TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
831 (and do these conversions themselves) */
832 in->conversion_buffer_size = num_read_buff_bytes;
833 in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
835 read_buff = in->conversion_buffer;
838 if (proxy_read(&in->proxy, read_buff, num_read_buff_bytes) == 0) {
840 * Do any conversions necessary to send the data in the format specified to/by the HAL
841 * (but different from the ALSA format), such as 24bit ->16bit, or 4chan -> 2chan.
843 if (format != PCM_FORMAT_S16_LE) {
844 /* we need to convert */
845 if (num_device_channels != num_req_channels) {
846 out_buff = read_buff;
849 if (format == PCM_FORMAT_S24_3LE) {
850 num_read_buff_bytes =
851 convert_24_3_to_16(read_buff, num_read_buff_bytes / 3, out_buff);
852 } else if (format == PCM_FORMAT_S32_LE) {
853 num_read_buff_bytes =
854 convert_32_to_16(read_buff, num_read_buff_bytes / 4, out_buff);
860 if (num_device_channels != num_req_channels) {
861 // ALOGV("chans dev:%d req:%d", num_device_channels, num_req_channels);
864 /* Num Channels conversion */
865 if (num_device_channels != num_req_channels) {
866 audio_format_t audio_format = in_get_format(&(in->stream.common));
867 unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
869 num_read_buff_bytes =
870 adjust_channels(read_buff, num_device_channels,
871 out_buff, num_req_channels,
872 sample_size_in_bytes, num_read_buff_bytes);
878 pthread_mutex_unlock(&in->lock);
880 return num_read_buff_bytes;
883 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
888 static int adev_open_input_stream(struct audio_hw_device *dev,
889 audio_io_handle_t handle,
890 audio_devices_t devices,
891 struct audio_config *config,
892 struct audio_stream_in **stream_in,
893 audio_input_flags_t flags __unused,
894 const char *address __unused,
895 audio_source_t source __unused)
897 ALOGV("usb: in adev_open_input_stream() rate:%" PRIu32 ", chanMask:0x%" PRIX32 ", fmt:%" PRIu8,
898 config->sample_rate, config->channel_mask, config->format);
900 struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
906 /* setup function pointers */
907 in->stream.common.get_sample_rate = in_get_sample_rate;
908 in->stream.common.set_sample_rate = in_set_sample_rate;
909 in->stream.common.get_buffer_size = in_get_buffer_size;
910 in->stream.common.get_channels = in_get_channels;
911 in->stream.common.get_format = in_get_format;
912 in->stream.common.set_format = in_set_format;
913 in->stream.common.standby = in_standby;
914 in->stream.common.dump = in_dump;
915 in->stream.common.set_parameters = in_set_parameters;
916 in->stream.common.get_parameters = in_get_parameters;
917 in->stream.common.add_audio_effect = in_add_audio_effect;
918 in->stream.common.remove_audio_effect = in_remove_audio_effect;
920 in->stream.set_gain = in_set_gain;
921 in->stream.read = in_read;
922 in->stream.get_input_frames_lost = in_get_input_frames_lost;
924 in->dev = (struct audio_device *)dev;
926 in->profile = &in->dev->in_profile;
928 struct pcm_config proxy_config;
931 if (config->sample_rate == 0) {
932 proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
933 } else if (profile_is_sample_rate_valid(in->profile, config->sample_rate)) {
934 proxy_config.rate = config->sample_rate;
936 proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
941 /* until the framework supports format conversion, just take what it asks for
942 * i.e. AUDIO_FORMAT_PCM_16_BIT */
943 if (config->format == AUDIO_FORMAT_DEFAULT) {
944 /* just return AUDIO_FORMAT_PCM_16_BIT until the framework supports other input
946 config->format = AUDIO_FORMAT_PCM_16_BIT;
947 proxy_config.format = PCM_FORMAT_S16_LE;
948 } else if (config->format == AUDIO_FORMAT_PCM_16_BIT) {
949 /* Always accept AUDIO_FORMAT_PCM_16_BIT until the framework supports other input
951 proxy_config.format = PCM_FORMAT_S16_LE;
953 /* When the framework support other formats, validate here */
954 config->format = AUDIO_FORMAT_PCM_16_BIT;
955 proxy_config.format = PCM_FORMAT_S16_LE;
959 if (config->channel_mask == AUDIO_CHANNEL_NONE) {
960 /* just return AUDIO_CHANNEL_IN_STEREO until the framework supports other input
962 config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
964 } else if (config->channel_mask != AUDIO_CHANNEL_IN_STEREO) {
965 /* allow only stereo capture for now */
966 config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
969 // proxy_config.channels = 0; /* don't change */
970 proxy_config.channels = profile_get_default_channel_count(in->profile);
972 proxy_prepare(&in->proxy, in->profile, &proxy_config);
976 in->conversion_buffer = NULL;
977 in->conversion_buffer_size = 0;
979 *stream_in = &in->stream;
984 static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream)
986 struct stream_in *in = (struct stream_in *)stream;
988 /* Close the pcm device */
989 in_standby(&stream->common);
991 free(in->conversion_buffer);
999 static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
1004 static char * adev_get_parameters(const struct audio_hw_device *dev, const char *keys)
1009 static int adev_init_check(const struct audio_hw_device *dev)
1014 static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
1019 static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
1024 static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
1029 static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
1034 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1039 static int adev_dump(const audio_hw_device_t *device, int fd)
1044 static int adev_close(hw_device_t *device)
1046 struct audio_device *adev = (struct audio_device *)device;
1052 static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device)
1054 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
1057 struct audio_device *adev = calloc(1, sizeof(struct audio_device));
1061 profile_init(&adev->out_profile, PCM_OUT);
1062 profile_init(&adev->in_profile, PCM_IN);
1064 adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
1065 adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
1066 adev->hw_device.common.module = (struct hw_module_t *)module;
1067 adev->hw_device.common.close = adev_close;
1069 adev->hw_device.init_check = adev_init_check;
1070 adev->hw_device.set_voice_volume = adev_set_voice_volume;
1071 adev->hw_device.set_master_volume = adev_set_master_volume;
1072 adev->hw_device.set_mode = adev_set_mode;
1073 adev->hw_device.set_mic_mute = adev_set_mic_mute;
1074 adev->hw_device.get_mic_mute = adev_get_mic_mute;
1075 adev->hw_device.set_parameters = adev_set_parameters;
1076 adev->hw_device.get_parameters = adev_get_parameters;
1077 adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
1078 adev->hw_device.open_output_stream = adev_open_output_stream;
1079 adev->hw_device.close_output_stream = adev_close_output_stream;
1080 adev->hw_device.open_input_stream = adev_open_input_stream;
1081 adev->hw_device.close_input_stream = adev_close_input_stream;
1082 adev->hw_device.dump = adev_dump;
1084 *device = &adev->hw_device.common;
1089 static struct hw_module_methods_t hal_module_methods = {
1093 struct audio_module HAL_MODULE_INFO_SYM = {
1095 .tag = HARDWARE_MODULE_TAG,
1096 .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
1097 .hal_api_version = HARDWARE_HAL_API_VERSION,
1098 .id = AUDIO_HARDWARE_MODULE_ID,
1099 .name = "USB audio HW HAL",
1100 .author = "The Android Open Source Project",
1101 .methods = &hal_module_methods,