mediatek: Import legacy libalsautils
* Taken from system/media/alsa_utils as of commit 9a0e6f0c19cd9a46eb35ff1980c42ea8b562d682
(android-11.0.0_r34)
Signed-off-by: bengris32 <bengris32@protonmail.ch>
Change-Id: I272735d93db39729f53a174983c8d39eee5c2ac2
diff --git a/alsa_utils_legacy/Android.bp b/alsa_utils_legacy/Android.bp
new file mode 100644
index 0000000..d2bfae0
--- /dev/null
+++ b/alsa_utils_legacy/Android.bp
@@ -0,0 +1,42 @@
+// Copyright (C) 2015 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+cc_library_shared {
+ name: "libalsautils_legacy",
+ vendor: true,
+ srcs: [
+ "alsa_device_profile.c",
+ "alsa_device_proxy.c",
+ "alsa_logging.c",
+ "alsa_format.c",
+ ],
+ export_include_dirs: ["include"],
+ header_libs: [
+ "libaudio_system_headers",
+ ],
+ export_header_lib_headers: [
+ "libaudio_system_headers",
+ ],
+ shared_libs: [
+ "liblog",
+ "libcutils",
+ "libtinyalsa",
+ "libaudioutils",
+ ],
+ cflags: [
+ "-Werror",
+ "-Wall",
+ "-Wno-unused-parameter",
+ ],
+}
diff --git a/alsa_utils_legacy/alsa_device_profile.c b/alsa_utils_legacy/alsa_device_profile.c
new file mode 100644
index 0000000..6b76bbe
--- /dev/null
+++ b/alsa_utils_legacy/alsa_device_profile.c
@@ -0,0 +1,654 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "alsa_device_profile"
+/*#define LOG_NDEBUG 0*/
+/*#define LOG_PCM_PARAMS 0*/
+
+#include <errno.h>
+#include <inttypes.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <cutils/properties.h>
+
+#include <log/log.h>
+
+#include "include/alsa_device_profile.h"
+#include "include/alsa_format.h"
+#include "include/alsa_logging.h"
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+#define PERIOD_DURATION_US (5 * 1000)
+
+#define DEFAULT_PERIOD_SIZE 1024
+
+static const char * const format_string_map[] = {
+ "AUDIO_FORMAT_PCM_16_BIT", /* "PCM_FORMAT_S16_LE", */
+ "AUDIO_FORMAT_PCM_32_BIT", /* "PCM_FORMAT_S32_LE", */
+ "AUDIO_FORMAT_PCM_8_BIT", /* "PCM_FORMAT_S8", */
+ "AUDIO_FORMAT_PCM_8_24_BIT", /* "PCM_FORMAT_S24_LE", */
+ "AUDIO_FORMAT_PCM_24_BIT_PACKED"/* "PCM_FORMAT_S24_3LE" */
+};
+
+extern int8_t const pcm_format_value_map[50];
+
+/* Sort these in terms of preference (best first).
+ 192 kHz is not first because it requires significant resources for possibly worse
+ quality and driver instability (depends on device).
+ The order here determines the default sample rate for the device.
+ AudioPolicyManager may not respect this ordering when picking sample rates.
+ Update MAX_PROFILE_SAMPLE_RATES after changing the array size.
+
+ TODO: remove 32000, 22050, 12000, 11025? Each sample rate check
+ requires opening the device which may cause pops. */
+static const unsigned std_sample_rates[] =
+ {96000, 88200, 192000, 176400, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000};
+
+static void profile_reset(alsa_device_profile* profile)
+{
+ profile->card = profile->device = -1;
+
+ /* terminate the attribute arrays with invalid values */
+ profile->formats[0] = PCM_FORMAT_INVALID;
+ profile->sample_rates[0] = 0;
+ profile->channel_counts[0] = 0;
+
+ profile->min_period_size = profile->max_period_size = 0;
+ profile->min_channel_count = profile->max_channel_count = DEFAULT_CHANNEL_COUNT;
+
+ profile->is_valid = false;
+}
+
+void profile_init(alsa_device_profile* profile, int direction)
+{
+ profile->direction = direction;
+ profile_reset(profile);
+}
+
+bool profile_is_initialized(const alsa_device_profile* profile)
+{
+ return profile->card >= 0 && profile->device >= 0;
+}
+
+bool profile_is_valid(const alsa_device_profile* profile) {
+ return profile->is_valid;
+}
+
+bool profile_is_cached_for(const alsa_device_profile* profile, int card, int device) {
+ return card == profile->card && device == profile->device;
+}
+
+void profile_decache(alsa_device_profile* profile) {
+ profile_reset(profile);
+}
+
+/*
+ * Returns the supplied value rounded up to the next even multiple of 16
+ */
+static unsigned int round_to_16_mult(unsigned int size)
+{
+ return (size + 15) & ~15; /* 0xFFFFFFF0; */
+}
+
+/*
+ * Returns the system defined minimum period size based on the supplied sample rate.
+ */
+unsigned profile_calc_min_period_size(const alsa_device_profile* profile, unsigned sample_rate)
+{
+ ALOGV("profile_calc_min_period_size(%p, rate:%d)", profile, sample_rate);
+ if (profile == NULL) {
+ return DEFAULT_PERIOD_SIZE;
+ } else {
+ unsigned period_us = property_get_int32("ro.audio.usb.period_us", PERIOD_DURATION_US);
+ unsigned num_sample_frames = ((uint64_t)sample_rate * period_us) / 1000000;
+
+ if (num_sample_frames < profile->min_period_size) {
+ num_sample_frames = profile->min_period_size;
+ }
+ return round_to_16_mult(num_sample_frames);
+ }
+}
+
+unsigned int profile_get_period_size(const alsa_device_profile* profile, unsigned sample_rate)
+{
+ unsigned int period_size = profile_calc_min_period_size(profile, sample_rate);
+ ALOGV("profile_get_period_size(rate:%d) = %d", sample_rate, period_size);
+ return period_size;
+}
+
+/*
+ * Sample Rate
+ */
+unsigned profile_get_default_sample_rate(const alsa_device_profile* profile)
+{
+ /*
+ * This is probably a poor algorithm. The default sample rate should be the highest (within
+ * limits) rate that is available for both input and output. HOWEVER, the profile has only
+ * one or the other, so that will need to be done at a higher level, like in the HAL.
+ */
+ /*
+ * TODO this won't be right in general. we should store a preferred rate as we are scanning.
+ * But right now it will return the highest rate, which may be correct.
+ */
+ return profile_is_valid(profile) ? profile->sample_rates[0] : DEFAULT_SAMPLE_RATE;
+}
+
+unsigned profile_get_highest_sample_rate(const alsa_device_profile* profile) {
+ /* The hightest sample rate is always stored in the first element of sample_rates.
+ * Note that profile_reset() initiaizes the first element of samples_rates to 0
+ * Which is what we want to return if the profile had not been read anyway.
+ */
+ return profile->sample_rates[0];
+}
+
+bool profile_is_sample_rate_valid(const alsa_device_profile* profile, unsigned rate)
+{
+ if (profile_is_valid(profile)) {
+ size_t index;
+ for (index = 0; profile->sample_rates[index] != 0; index++) {
+ if (profile->sample_rates[index] == rate) {
+ return true;
+ }
+ }
+
+ return false;
+ } else {
+ ALOGW("**** PROFILE NOT VALID!");
+ return rate == DEFAULT_SAMPLE_RATE;
+ }
+}
+
+/*
+ * Format
+ */
+enum pcm_format profile_get_default_format(const alsa_device_profile* profile)
+{
+ /*
+ * TODO this won't be right in general. we should store a preferred format as we are scanning.
+ */
+ return profile_is_valid(profile) ? profile->formats[0] : DEFAULT_SAMPLE_FORMAT;
+}
+
+bool profile_is_format_valid(const alsa_device_profile* profile, enum pcm_format fmt) {
+ if (profile_is_valid(profile)) {
+ size_t index;
+ for (index = 0; profile->formats[index] != PCM_FORMAT_INVALID; index++) {
+ if (profile->formats[index] == fmt) {
+ return true;
+ }
+ }
+
+ return false;
+ } else {
+ return fmt == DEFAULT_SAMPLE_FORMAT;
+ }
+}
+
+/*
+ * Channels
+ */
+unsigned profile_get_default_channel_count(const alsa_device_profile* profile)
+{
+ return profile_is_valid(profile) ? profile->channel_counts[0] : DEFAULT_CHANNEL_COUNT;
+}
+
+unsigned profile_get_closest_channel_count(const alsa_device_profile* profile, unsigned count)
+{
+ if (profile_is_valid(profile)) {
+ if (count < profile->min_channel_count) {
+ return profile->min_channel_count;
+ } else if (count > profile->max_channel_count) {
+ return profile->max_channel_count;
+ } else {
+ return count;
+ }
+ } else {
+ return 0;
+ }
+}
+
+bool profile_is_channel_count_valid(const alsa_device_profile* profile, unsigned count)
+{
+ if (profile_is_initialized(profile)) {
+ return count >= profile->min_channel_count && count <= profile->max_channel_count;
+ } else {
+ return count == DEFAULT_CHANNEL_COUNT;
+ }
+}
+
+static bool profile_test_sample_rate(const alsa_device_profile* profile, unsigned rate)
+{
+ struct pcm_config config = profile->default_config;
+ config.rate = rate;
+
+ bool works = false; /* let's be pessimistic */
+ struct pcm * pcm = pcm_open(profile->card, profile->device,
+ profile->direction, &config);
+
+ if (pcm != NULL) {
+ works = pcm_is_ready(pcm);
+ pcm_close(pcm);
+ }
+
+ return works;
+}
+
+static unsigned profile_enum_sample_rates(alsa_device_profile* profile, unsigned min, unsigned max)
+{
+ unsigned num_entries = 0;
+ unsigned index;
+
+ for (index = 0; index < ARRAY_SIZE(std_sample_rates) &&
+ num_entries < ARRAY_SIZE(profile->sample_rates) - 1;
+ index++) {
+ if (std_sample_rates[index] >= min && std_sample_rates[index] <= max
+ && profile_test_sample_rate(profile, std_sample_rates[index])) {
+ profile->sample_rates[num_entries++] = std_sample_rates[index];
+ }
+ }
+ profile->sample_rates[num_entries] = 0; /* terminate */
+ return num_entries; /* return # of supported rates */
+}
+
+static unsigned profile_enum_sample_formats(alsa_device_profile* profile, struct pcm_mask * mask)
+{
+ const int num_slots = ARRAY_SIZE(mask->bits);
+ const int bits_per_slot = sizeof(mask->bits[0]) * 8;
+
+ const int table_size = ARRAY_SIZE(pcm_format_value_map);
+
+ int slot_index, bit_index, table_index;
+ table_index = 0;
+ int num_written = 0;
+ for (slot_index = 0; slot_index < num_slots && table_index < table_size;
+ slot_index++) {
+ unsigned bit_mask = 1;
+ for (bit_index = 0;
+ bit_index < bits_per_slot && table_index < table_size;
+ bit_index++) {
+ if ((mask->bits[slot_index] & bit_mask) != 0) {
+ enum pcm_format format = pcm_format_value_map[table_index];
+ /* Never return invalid (unrecognized) or 8-bit */
+ if (format != PCM_FORMAT_INVALID && format != PCM_FORMAT_S8) {
+ profile->formats[num_written++] = format;
+ if (num_written == ARRAY_SIZE(profile->formats) - 1) {
+ /* leave at least one PCM_FORMAT_INVALID at the end */
+ goto end;
+ }
+ }
+ }
+ bit_mask <<= 1;
+ table_index++;
+ }
+ }
+end:
+ profile->formats[num_written] = PCM_FORMAT_INVALID;
+ return num_written;
+}
+
+static unsigned profile_enum_channel_counts(alsa_device_profile* profile, unsigned min,
+ unsigned max)
+{
+ /* modify alsa_device_profile.h if you change the std_channel_counts[] array. */
+ static const unsigned std_channel_counts[] = {8, 7, 6, 5, 4, 3, 2, 1};
+
+ unsigned num_counts = 0;
+ unsigned index;
+ /* TODO write a profile_test_channel_count() */
+ /* Ensure there is at least one invalid channel count to terminate the channel counts array */
+ for (index = 0; index < ARRAY_SIZE(std_channel_counts) &&
+ num_counts < ARRAY_SIZE(profile->channel_counts) - 1;
+ index++) {
+ /* TODO Do we want a channel counts test? */
+ if (std_channel_counts[index] >= min && std_channel_counts[index] <= max /* &&
+ profile_test_channel_count(profile, channel_counts[index])*/) {
+ profile->channel_counts[num_counts++] = std_channel_counts[index];
+ }
+ }
+ // if we have no match with the standard counts, we use the largest (preferred) std count.
+ if (num_counts == 0) {
+ ALOGW("usb device does not match std channel counts, setting to %d",
+ std_channel_counts[0]);
+ profile->channel_counts[num_counts++] = std_channel_counts[0];
+ }
+ profile->channel_counts[num_counts] = 0;
+ return num_counts; /* return # of supported counts */
+}
+
+/*
+ * Reads and decodes configuration info from the specified ALSA card/device.
+ */
+static int read_alsa_device_config(alsa_device_profile * profile, struct pcm_config * config)
+{
+ ALOGV("usb:audio_hw - read_alsa_device_config(c:%d d:%d t:0x%X)",
+ profile->card, profile->device, profile->direction);
+
+ if (profile->card < 0 || profile->device < 0) {
+ return -EINVAL;
+ }
+
+ struct pcm_params * alsa_hw_params =
+ pcm_params_get(profile->card, profile->device, profile->direction);
+ if (alsa_hw_params == NULL) {
+ return -EINVAL;
+ }
+
+ profile->min_period_size = pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIOD_SIZE);
+ profile->max_period_size = pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIOD_SIZE);
+
+ profile->min_channel_count = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+ profile->max_channel_count = pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS);
+
+ int ret = 0;
+
+ /*
+ * This Logging will be useful when testing new USB devices.
+ */
+#ifdef LOG_PCM_PARAMS
+ log_pcm_params(alsa_hw_params);
+#endif
+
+ config->channels = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+ // For output devices, let's make sure we choose at least stereo
+ // (assuming the device supports it).
+ if (profile->direction == PCM_OUT &&
+ config->channels < 2 && pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS) >= 2) {
+ config->channels = 2;
+ }
+ config->rate = pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE);
+ // Prefer 48K or 44.1K
+ if (config->rate < 48000 &&
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE) >= 48000) {
+ config->rate = 48000;
+ } else if (config->rate < 44100 &&
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE) >= 44100) {
+ config->rate = 44100;
+ }
+ config->period_size = profile_calc_min_period_size(profile, config->rate);
+ config->period_count = pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIODS);
+ config->format = get_pcm_format_for_mask(pcm_params_get_mask(alsa_hw_params, PCM_PARAM_FORMAT));
+#ifdef LOG_PCM_PARAMS
+ log_pcm_config(config, "read_alsa_device_config");
+#endif
+ if (config->format == PCM_FORMAT_INVALID) {
+ ret = -EINVAL;
+ }
+
+ pcm_params_free(alsa_hw_params);
+
+ return ret;
+}
+
+bool profile_read_device_info(alsa_device_profile* profile)
+{
+ if (!profile_is_initialized(profile)) {
+ return false;
+ }
+
+ /* let's get some defaults */
+ read_alsa_device_config(profile, &profile->default_config);
+ ALOGV("default_config chans:%d rate:%d format:%d count:%d size:%d",
+ profile->default_config.channels, profile->default_config.rate,
+ profile->default_config.format, profile->default_config.period_count,
+ profile->default_config.period_size);
+
+ struct pcm_params * alsa_hw_params = pcm_params_get(profile->card,
+ profile->device,
+ profile->direction);
+ if (alsa_hw_params == NULL) {
+ return false;
+ }
+
+ /* Formats */
+ struct pcm_mask * format_mask = pcm_params_get_mask(alsa_hw_params, PCM_PARAM_FORMAT);
+ profile_enum_sample_formats(profile, format_mask);
+
+ /* Channels */
+ profile_enum_channel_counts(
+ profile, pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS));
+
+ /* Sample Rates */
+ profile_enum_sample_rates(
+ profile, pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE));
+
+ profile->is_valid = true;
+
+ pcm_params_free(alsa_hw_params);
+ return true;
+}
+
+char * profile_get_sample_rate_strs(const alsa_device_profile* profile)
+{
+ /* if we assume that rate strings are about 5 characters (48000 is 5), plus ~1 for a
+ * delimiter "|" this buffer has room for about 22 rate strings which seems like
+ * way too much, but it's a stack variable so only temporary.
+ */
+ char buffer[128];
+ buffer[0] = '\0';
+ size_t buffSize = ARRAY_SIZE(buffer);
+ size_t curStrLen = 0;
+
+ char numBuffer[32];
+
+ size_t numEntries = 0;
+ size_t index;
+ for (index = 0; profile->sample_rates[index] != 0; index++) {
+ snprintf(numBuffer, sizeof(numBuffer), "%u", profile->sample_rates[index]);
+ // account for both the null, and potentially the bar.
+ if (buffSize - curStrLen < strlen(numBuffer) + (numEntries != 0 ? 2 : 1)) {
+ /* we don't have room for another, so bail at this point rather than
+ * return a malformed rate string
+ */
+ break;
+ }
+ if (numEntries++ != 0) {
+ strlcat(buffer, "|", buffSize);
+ }
+ curStrLen = strlcat(buffer, numBuffer, buffSize);
+ }
+
+ return strdup(buffer);
+}
+
+char * profile_get_format_strs(const alsa_device_profile* profile)
+{
+ /* if we assume that format strings are about 24 characters (AUDIO_FORMAT_PCM_16_BIT is 23),
+ * plus ~1 for a delimiter "|" this buffer has room for about 10 format strings which seems
+ * like way too much, but it's a stack variable so only temporary.
+ */
+ char buffer[256];
+ buffer[0] = '\0';
+ size_t buffSize = ARRAY_SIZE(buffer);
+ size_t curStrLen = 0;
+
+ size_t numEntries = 0;
+ size_t index = 0;
+ for (index = 0; profile->formats[index] != PCM_FORMAT_INVALID; index++) {
+ // account for both the null, and potentially the bar.
+ if (buffSize - curStrLen < strlen(format_string_map[profile->formats[index]])
+ + (numEntries != 0 ? 2 : 1)) {
+ /* we don't have room for another, so bail at this point rather than
+ * return a malformed rate string
+ */
+ break;
+ }
+ if (numEntries++ != 0) {
+ strlcat(buffer, "|", buffSize);
+ }
+ curStrLen = strlcat(buffer, format_string_map[profile->formats[index]], buffSize);
+ }
+
+ return strdup(buffer);
+}
+
+char * profile_get_channel_count_strs(const alsa_device_profile* profile)
+{
+ // FIXME implicit fixed channel count assumption here (FCC_8).
+ // we use only the canonical even number channel position masks.
+ static const char * const out_chans_strs[] = {
+ /* 0 */"AUDIO_CHANNEL_NONE", /* will never be taken as this is a terminator */
+ /* 1 */"AUDIO_CHANNEL_OUT_MONO",
+ /* 2 */"AUDIO_CHANNEL_OUT_STEREO",
+ /* 3 */ /* "AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_FRONT_CENTER" */ NULL,
+ /* 4 */"AUDIO_CHANNEL_OUT_QUAD",
+ /* 5 */ /* "AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_FRONT_CENTER" */ NULL,
+ /* 6 */"AUDIO_CHANNEL_OUT_5POINT1",
+ /* 7 */ /* "AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_BACK_CENTER" */ NULL,
+ /* 8 */"AUDIO_CHANNEL_OUT_7POINT1",
+ /* channel counts greater than this not considered */
+ };
+
+ static const char * const in_chans_strs[] = {
+ /* 0 */"AUDIO_CHANNEL_NONE", /* will never be taken as this is a terminator */
+ /* 1 */"AUDIO_CHANNEL_IN_MONO",
+ /* 2 */"AUDIO_CHANNEL_IN_STEREO",
+ /* channel counts greater than this not considered */
+ };
+
+ static const char * const index_chans_strs[] = {
+ /* 0 */"AUDIO_CHANNEL_NONE", /* will never be taken as this is a terminator */
+ /* 1 */"AUDIO_CHANNEL_INDEX_MASK_1",
+ /* 2 */"AUDIO_CHANNEL_INDEX_MASK_2",
+ /* 3 */"AUDIO_CHANNEL_INDEX_MASK_3",
+ /* 4 */"AUDIO_CHANNEL_INDEX_MASK_4",
+ /* 5 */"AUDIO_CHANNEL_INDEX_MASK_5",
+ /* 6 */"AUDIO_CHANNEL_INDEX_MASK_6",
+ /* 7 */"AUDIO_CHANNEL_INDEX_MASK_7",
+ /* 8 */"AUDIO_CHANNEL_INDEX_MASK_8",
+ };
+
+ const bool isOutProfile = profile->direction == PCM_OUT;
+
+ const char * const * const chans_strs = isOutProfile ? out_chans_strs : in_chans_strs;
+ const size_t chans_strs_size =
+ isOutProfile ? ARRAY_SIZE(out_chans_strs) : ARRAY_SIZE(in_chans_strs);
+
+ /*
+ * If we assume each channel string is 26 chars ("AUDIO_CHANNEL_INDEX_MASK_8" is 26) + 1 for,
+ * the "|" delimiter, then we allocate room for 16 strings.
+ */
+ char buffer[27 * 16 + 1]; /* caution, may need to be expanded */
+ buffer[0] = '\0';
+ size_t buffSize = ARRAY_SIZE(buffer);
+ size_t curStrLen = 0;
+
+ /* We currently support MONO and STEREO, and always report STEREO but some (many)
+ * USB Audio Devices may only announce support for MONO (a headset mic for example), or
+ * The total number of output channels. SO, if the device itself doesn't explicitly
+ * support STEREO, append to the channel config strings we are generating.
+ *
+ * The MONO and STEREO positional channel masks are provided for legacy compatibility.
+ * For multichannel (n > 2) we only expose channel index masks.
+ */
+ // Always support stereo
+ curStrLen = strlcat(buffer, chans_strs[2], buffSize);
+
+ size_t index;
+ unsigned channel_count;
+ for (index = 0;
+ (channel_count = profile->channel_counts[index]) != 0;
+ index++) {
+
+ /* we only show positional information for mono (stereo handled already) */
+ if (channel_count < chans_strs_size
+ && chans_strs[channel_count] != NULL
+ && channel_count < 2 /* positional only for fewer than 2 channels */) {
+ // account for the '|' and the '\0'
+ if (buffSize - curStrLen < strlen(chans_strs[channel_count]) + 2) {
+ /* we don't have room for another, so bail at this point rather than
+ * return a malformed rate string
+ */
+ break;
+ }
+
+ strlcat(buffer, "|", buffSize);
+ curStrLen = strlcat(buffer, chans_strs[channel_count], buffSize);
+ }
+
+ // handle channel index masks for both input and output
+ // +2 to account for the '|' and the '\0'
+ if (buffSize - curStrLen < strlen(index_chans_strs[channel_count]) + 2) {
+ /* we don't have room for another, so bail at this point rather than
+ * return a malformed rate string
+ */
+ break;
+ }
+
+ strlcat(buffer, "|", buffSize);
+ curStrLen = strlcat(buffer, index_chans_strs[channel_count], buffSize);
+ }
+
+ return strdup(buffer);
+}
+
+void profile_dump(const alsa_device_profile* profile, int fd)
+{
+ if (profile == NULL) {
+ dprintf(fd, " %s\n", "No USB Profile");
+ return; /* bail early */
+ }
+
+ if (!profile->is_valid) {
+ dprintf(fd, " Profile is INVALID");
+ }
+
+ /* card/device/direction */
+ dprintf(fd, " card:%d, device:%d - %s\n",
+ profile->card, profile->device, profile->direction == PCM_OUT ? "OUT" : "IN");
+
+ /* formats */
+ dprintf(fd, " Formats: ");
+ for (int fmtIndex = 0;
+ profile->formats[fmtIndex] != PCM_FORMAT_INVALID && fmtIndex < MAX_PROFILE_FORMATS;
+ fmtIndex++) {
+ dprintf(fd, "%d ", profile->formats[fmtIndex]);
+ }
+ dprintf(fd, "\n");
+
+ /* sample rates */
+ dprintf(fd, " Rates: ");
+ for (int rateIndex = 0;
+ profile->sample_rates[rateIndex] != 0 && rateIndex < MAX_PROFILE_SAMPLE_RATES;
+ rateIndex++) {
+ dprintf(fd, "%u ", profile->sample_rates[rateIndex]);
+ }
+ dprintf(fd, "\n");
+
+ // channel counts
+ dprintf(fd, " Channel Counts: ");
+ for (int cntIndex = 0;
+ profile->channel_counts[cntIndex] != 0 && cntIndex < MAX_PROFILE_CHANNEL_COUNTS;
+ cntIndex++) {
+ dprintf(fd, "%u ", profile->channel_counts[cntIndex]);
+ }
+ dprintf(fd, "\n");
+
+ dprintf(fd, " min/max period size [%u : %u]\n",
+ profile->min_period_size,profile-> max_period_size);
+ dprintf(fd, " min/max channel count [%u : %u]\n",
+ profile->min_channel_count, profile->max_channel_count);
+
+ // struct pcm_config default_config;
+ dprintf(fd, " Default Config:\n");
+ dprintf(fd, " channels: %d\n", profile->default_config.channels);
+ dprintf(fd, " rate: %d\n", profile->default_config.rate);
+ dprintf(fd, " period_size: %d\n", profile->default_config.period_size);
+ dprintf(fd, " period_count: %d\n", profile->default_config.period_count);
+ dprintf(fd, " format: %d\n", profile->default_config.format);
+}
diff --git a/alsa_utils_legacy/alsa_device_proxy.c b/alsa_utils_legacy/alsa_device_proxy.c
new file mode 100644
index 0000000..376ae89
--- /dev/null
+++ b/alsa_utils_legacy/alsa_device_proxy.c
@@ -0,0 +1,321 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "alsa_device_proxy"
+/*#define LOG_NDEBUG 0*/
+/*#define LOG_PCM_PARAMS 0*/
+
+#include <log/log.h>
+
+#include <errno.h>
+#include <stdio.h>
+#include <string.h>
+
+#include <audio_utils/clock.h>
+
+#include "include/alsa_device_proxy.h"
+
+#include "include/alsa_logging.h"
+
+#define DEFAULT_PERIOD_SIZE 1024
+#define DEFAULT_PERIOD_COUNT 2
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+static const unsigned format_byte_size_map[] = {
+ 2, /* PCM_FORMAT_S16_LE */
+ 4, /* PCM_FORMAT_S32_LE */
+ 1, /* PCM_FORMAT_S8 */
+ 4, /* PCM_FORMAT_S24_LE */
+ 3, /* PCM_FORMAT_S24_3LE */
+};
+
+int proxy_prepare(alsa_device_proxy * proxy, const alsa_device_profile* profile,
+ struct pcm_config * config)
+{
+ int ret = 0;
+
+ ALOGV("proxy_prepare(c:%d, d:%d)", profile->card, profile->device);
+
+ proxy->profile = profile;
+
+#ifdef LOG_PCM_PARAMS
+ log_pcm_config(config, "proxy_setup()");
+#endif
+
+ if (config->format != PCM_FORMAT_INVALID && profile_is_format_valid(profile, config->format)) {
+ proxy->alsa_config.format = config->format;
+ } else {
+ proxy->alsa_config.format = profile->default_config.format;
+ ALOGW("Invalid format %d - using default %d.",
+ config->format, profile->default_config.format);
+ // Indicate override when default format was not requested
+ if (config->format != PCM_FORMAT_INVALID) {
+ ret = -EINVAL;
+ }
+ }
+
+ if (config->rate != 0 && profile_is_sample_rate_valid(profile, config->rate)) {
+ proxy->alsa_config.rate = config->rate;
+ } else {
+ proxy->alsa_config.rate = profile->default_config.rate;
+ ALOGW("Invalid sample rate %u - using default %u.",
+ config->rate, profile->default_config.rate);
+ // Indicate override when default rate was not requested
+ if (config->rate != 0) {
+ ret = -EINVAL;
+ }
+ }
+
+ if (config->channels != 0 && profile_is_channel_count_valid(profile, config->channels)) {
+ proxy->alsa_config.channels = config->channels;
+ } else {
+ proxy->alsa_config.channels = profile_get_closest_channel_count(profile, config->channels);
+ ALOGW("Invalid channel count %u - using closest %u.",
+ config->channels, proxy->alsa_config.channels);
+ // Indicate override when default channel count was not requested
+ if (config->channels != 0) {
+ ret = -EINVAL;
+ }
+ }
+
+ proxy->alsa_config.period_count = profile->default_config.period_count;
+ proxy->alsa_config.period_size =
+ profile_get_period_size(proxy->profile, proxy->alsa_config.rate);
+
+ // Hack for USB accessory audio.
+ // Here we set the correct value for period_count if tinyalsa fails to get it from the
+ // f_audio_source driver.
+ if (proxy->alsa_config.period_count == 0) {
+ proxy->alsa_config.period_count = 4;
+ }
+
+ proxy->pcm = NULL;
+ // config format should be checked earlier against profile.
+ if (config->format >= 0 && (size_t)config->format < ARRAY_SIZE(format_byte_size_map)) {
+ proxy->frame_size = format_byte_size_map[config->format] * proxy->alsa_config.channels;
+ } else {
+ proxy->frame_size = 1;
+ }
+
+ // let's check to make sure we can ACTUALLY use the maximum rate (with the channel count)
+ // Note that profile->sample_rates is sorted highest to lowest, so the scan will get
+ // us the highest working rate
+ int max_rate_index = proxy_scan_rates(proxy, profile->sample_rates);
+ if (max_rate_index >= 0) {
+ if (proxy->alsa_config.rate > profile->sample_rates[max_rate_index]) {
+ ALOGW("Limiting sampling rate from %u to %u.",
+ proxy->alsa_config.rate, profile->sample_rates[max_rate_index]);
+ proxy->alsa_config.rate = profile->sample_rates[max_rate_index];
+ ret = -EINVAL;
+ }
+ }
+ return ret;
+}
+
+int proxy_open(alsa_device_proxy * proxy)
+{
+ const alsa_device_profile* profile = proxy->profile;
+ ALOGV("proxy_open(card:%d device:%d %s)", profile->card, profile->device,
+ profile->direction == PCM_OUT ? "PCM_OUT" : "PCM_IN");
+
+ if (profile->card < 0 || profile->device < 0) {
+ return -EINVAL;
+ }
+
+ proxy->pcm = pcm_open(profile->card, profile->device,
+ profile->direction | PCM_MONOTONIC, &proxy->alsa_config);
+ if (proxy->pcm == NULL) {
+ return -ENOMEM;
+ }
+
+ if (!pcm_is_ready(proxy->pcm)) {
+ ALOGE(" proxy_open() pcm_is_ready() failed: %s", pcm_get_error(proxy->pcm));
+#if defined(LOG_PCM_PARAMS)
+ log_pcm_config(&proxy->alsa_config, "config");
+#endif
+ pcm_close(proxy->pcm);
+ proxy->pcm = NULL;
+ return -ENOMEM;
+ }
+
+ return 0;
+}
+
+void proxy_close(alsa_device_proxy * proxy)
+{
+ ALOGV("proxy_close() [pcm:%p]", proxy->pcm);
+
+ if (proxy->pcm != NULL) {
+ pcm_close(proxy->pcm);
+ proxy->pcm = NULL;
+ }
+}
+
+/*
+ * Sample Rate
+ */
+unsigned proxy_get_sample_rate(const alsa_device_proxy * proxy)
+{
+ return proxy->alsa_config.rate;
+}
+
+/*
+ * Format
+ */
+enum pcm_format proxy_get_format(const alsa_device_proxy * proxy)
+{
+ return proxy->alsa_config.format;
+}
+
+/*
+ * Channel Count
+ */
+unsigned proxy_get_channel_count(const alsa_device_proxy * proxy)
+{
+ return proxy->alsa_config.channels;
+}
+
+/*
+ * Other
+ */
+unsigned int proxy_get_period_size(const alsa_device_proxy * proxy)
+{
+ return proxy->alsa_config.period_size;
+}
+
+unsigned int proxy_get_period_count(const alsa_device_proxy * proxy)
+{
+ return proxy->alsa_config.period_count;
+}
+
+unsigned proxy_get_latency(const alsa_device_proxy * proxy)
+{
+ return (proxy_get_period_size(proxy) * proxy_get_period_count(proxy) * 1000)
+ / proxy_get_sample_rate(proxy);
+}
+
+int proxy_get_presentation_position(const alsa_device_proxy * proxy,
+ uint64_t *frames, struct timespec *timestamp)
+{
+ int ret = -EPERM; // -1
+ unsigned int avail;
+ if (proxy->pcm != NULL
+ && pcm_get_htimestamp(proxy->pcm, &avail, timestamp) == 0) {
+ const size_t kernel_buffer_size =
+ proxy->alsa_config.period_size * proxy->alsa_config.period_count;
+ if (avail > kernel_buffer_size) {
+ ALOGE("available frames(%u) > buffer size(%zu)", avail, kernel_buffer_size);
+ } else {
+ int64_t signed_frames = proxy->transferred - kernel_buffer_size + avail;
+ // It is possible to compensate for additional driver and device delay
+ // by changing signed_frames. Example:
+ // signed_frames -= 20 /* ms */ * proxy->alsa_config.rate / 1000;
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ ret = 0;
+ }
+ }
+ }
+ return ret;
+}
+
+int proxy_get_capture_position(const alsa_device_proxy * proxy,
+ int64_t *frames, int64_t *time)
+{
+ int ret = -ENOSYS;
+ unsigned int avail;
+ struct timespec timestamp;
+ // TODO: add logging for tinyalsa errors.
+ if (proxy->pcm != NULL
+ && pcm_get_htimestamp(proxy->pcm, &avail, ×tamp) == 0) {
+ const size_t kernel_buffer_size =
+ proxy->alsa_config.period_size * proxy->alsa_config.period_count;
+ if (avail > kernel_buffer_size) {
+ ALOGE("available frames(%u) > buffer size(%zu)", avail, kernel_buffer_size);
+ } else {
+ *frames = proxy->transferred + avail;
+ *time = audio_utils_ns_from_timespec(×tamp);
+ ret = 0;
+ }
+ }
+ return ret;
+}
+
+/*
+ * I/O
+ */
+int proxy_write(alsa_device_proxy * proxy, const void *data, unsigned int count)
+{
+ int ret = pcm_write(proxy->pcm, data, count);
+ if (ret == 0) {
+ proxy->transferred += count / proxy->frame_size;
+ }
+ return ret;
+}
+
+int proxy_read(alsa_device_proxy * proxy, void *data, unsigned int count)
+{
+ int ret = pcm_read(proxy->pcm, data, count);
+ if (ret == 0) {
+ proxy->transferred += count / proxy->frame_size;
+ }
+ return ret;
+}
+
+/*
+ * Debugging
+ */
+void proxy_dump(const alsa_device_proxy* proxy, int fd)
+{
+ if (proxy != NULL) {
+ dprintf(fd, " channels: %d\n", proxy->alsa_config.channels);
+ dprintf(fd, " rate: %d\n", proxy->alsa_config.rate);
+ dprintf(fd, " period_size: %d\n", proxy->alsa_config.period_size);
+ dprintf(fd, " period_count: %d\n", proxy->alsa_config.period_count);
+ dprintf(fd, " format: %d\n", proxy->alsa_config.format);
+ }
+}
+
+int proxy_scan_rates(alsa_device_proxy * proxy, const unsigned sample_rates[]) {
+ const alsa_device_profile* profile = proxy->profile;
+ if (profile->card < 0 || profile->device < 0) {
+ return -EINVAL;
+ }
+
+ struct pcm_config alsa_config;
+ memcpy(&alsa_config, &proxy->alsa_config, sizeof(alsa_config));
+
+ struct pcm * alsa_pcm;
+ int rate_index = 0;
+ while (sample_rates[rate_index] != 0) {
+ alsa_config.rate = sample_rates[rate_index];
+ alsa_pcm = pcm_open(profile->card, profile->device,
+ profile->direction | PCM_MONOTONIC, &alsa_config);
+ if (alsa_pcm != NULL) {
+ if (pcm_is_ready(alsa_pcm)) {
+ pcm_close(alsa_pcm);
+ return rate_index;
+ }
+
+ pcm_close(alsa_pcm);
+ }
+
+ rate_index++;
+ }
+
+ return -EINVAL;
+}
diff --git a/alsa_utils_legacy/alsa_format.c b/alsa_utils_legacy/alsa_format.c
new file mode 100644
index 0000000..de8fcf2
--- /dev/null
+++ b/alsa_utils_legacy/alsa_format.c
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "alsa_format"
+/*#define LOG_NDEBUG 0*/
+
+#include "include/alsa_format.h"
+
+#include <tinyalsa/asoundlib.h>
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+/*
+ * Maps from bit position in pcm_mask to PCM_ format constants.
+ */
+int8_t const pcm_format_value_map[50] = {
+ PCM_FORMAT_S8, /* 00 - SNDRV_PCM_FORMAT_S8 */
+ PCM_FORMAT_INVALID, /* 01 - SNDRV_PCM_FORMAT_U8 */
+ PCM_FORMAT_S16_LE, /* 02 - SNDRV_PCM_FORMAT_S16_LE */
+ PCM_FORMAT_INVALID, /* 03 - SNDRV_PCM_FORMAT_S16_BE */
+ PCM_FORMAT_INVALID, /* 04 - SNDRV_PCM_FORMAT_U16_LE */
+ PCM_FORMAT_INVALID, /* 05 - SNDRV_PCM_FORMAT_U16_BE */
+ PCM_FORMAT_S24_LE, /* 06 - SNDRV_PCM_FORMAT_S24_LE */
+ PCM_FORMAT_INVALID, /* 07 - SNDRV_PCM_FORMAT_S24_BE */
+ PCM_FORMAT_INVALID, /* 08 - SNDRV_PCM_FORMAT_U24_LE */
+ PCM_FORMAT_INVALID, /* 09 - SNDRV_PCM_FORMAT_U24_BE */
+ PCM_FORMAT_S32_LE, /* 10 - SNDRV_PCM_FORMAT_S32_LE */
+ PCM_FORMAT_INVALID, /* 11 - SNDRV_PCM_FORMAT_S32_BE */
+ PCM_FORMAT_INVALID, /* 12 - SNDRV_PCM_FORMAT_U32_LE */
+ PCM_FORMAT_INVALID, /* 13 - SNDRV_PCM_FORMAT_U32_BE */
+ PCM_FORMAT_INVALID, /* 14 - SNDRV_PCM_FORMAT_FLOAT_LE */
+ PCM_FORMAT_INVALID, /* 15 - SNDRV_PCM_FORMAT_FLOAT_BE */
+ PCM_FORMAT_INVALID, /* 16 - SNDRV_PCM_FORMAT_FLOAT64_LE */
+ PCM_FORMAT_INVALID, /* 17 - SNDRV_PCM_FORMAT_FLOAT64_BE */
+ PCM_FORMAT_INVALID, /* 18 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE */
+ PCM_FORMAT_INVALID, /* 19 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE */
+ PCM_FORMAT_INVALID, /* 20 - SNDRV_PCM_FORMAT_MU_LAW */
+ PCM_FORMAT_INVALID, /* 21 - SNDRV_PCM_FORMAT_A_LAW */
+ PCM_FORMAT_INVALID, /* 22 - SNDRV_PCM_FORMAT_IMA_ADPCM */
+ PCM_FORMAT_INVALID, /* 23 - SNDRV_PCM_FORMAT_MPEG */
+ PCM_FORMAT_INVALID, /* 24 - SNDRV_PCM_FORMAT_GSM */
+ PCM_FORMAT_INVALID, /* 25 -> 30 (not assigned) */
+ PCM_FORMAT_INVALID,
+ PCM_FORMAT_INVALID,
+ PCM_FORMAT_INVALID,
+ PCM_FORMAT_INVALID,
+ PCM_FORMAT_INVALID,
+ PCM_FORMAT_INVALID, /* 31 - SNDRV_PCM_FORMAT_SPECIAL */
+ PCM_FORMAT_S24_3LE, /* 32 - SNDRV_PCM_FORMAT_S24_3LE */
+ PCM_FORMAT_INVALID, /* 33 - SNDRV_PCM_FORMAT_S24_3BE */
+ PCM_FORMAT_INVALID, /* 34 - SNDRV_PCM_FORMAT_U24_3LE */
+ PCM_FORMAT_INVALID, /* 35 - SNDRV_PCM_FORMAT_U24_3BE */
+ PCM_FORMAT_INVALID, /* 36 - SNDRV_PCM_FORMAT_S20_3LE */
+ PCM_FORMAT_INVALID, /* 37 - SNDRV_PCM_FORMAT_S20_3BE */
+ PCM_FORMAT_INVALID, /* 38 - SNDRV_PCM_FORMAT_U20_3LE */
+ PCM_FORMAT_INVALID, /* 39 - SNDRV_PCM_FORMAT_U20_3BE */
+ PCM_FORMAT_INVALID, /* 40 - SNDRV_PCM_FORMAT_S18_3LE */
+ PCM_FORMAT_INVALID, /* 41 - SNDRV_PCM_FORMAT_S18_3BE */
+ PCM_FORMAT_INVALID, /* 42 - SNDRV_PCM_FORMAT_U18_3LE */
+ PCM_FORMAT_INVALID, /* 43 - SNDRV_PCM_FORMAT_U18_3BE */
+ PCM_FORMAT_INVALID, /* 44 - SNDRV_PCM_FORMAT_G723_24 */
+ PCM_FORMAT_INVALID, /* 45 - SNDRV_PCM_FORMAT_G723_24_1B */
+ PCM_FORMAT_INVALID, /* 46 - SNDRV_PCM_FORMAT_G723_40 */
+ PCM_FORMAT_INVALID, /* 47 - SNDRV_PCM_FORMAT_G723_40_1B */
+ PCM_FORMAT_INVALID, /* 48 - SNDRV_PCM_FORMAT_DSD_U8 */
+ PCM_FORMAT_INVALID /* 49 - SNDRV_PCM_FORMAT_DSD_U16_LE */
+};
+
+/*
+ * Scans the provided format mask and returns the first non-8 bit sample
+ * format supported by the devices.
+ */
+enum pcm_format get_pcm_format_for_mask(struct pcm_mask* mask)
+{
+ int num_slots = ARRAY_SIZE(mask->bits);
+ int bits_per_slot = sizeof(mask->bits[0]) * 8;
+
+ int table_size = ARRAY_SIZE(pcm_format_value_map);
+
+ int slot_index, bit_index, table_index;
+ table_index = 0;
+ for (slot_index = 0; slot_index < num_slots && table_index < table_size; slot_index++) {
+ unsigned bit_mask = 1;
+ for (bit_index = 0; bit_index < bits_per_slot && table_index < table_size; bit_index++) {
+ /* skip any 8-bit formats */
+ if (table_index >= 2 && (mask->bits[slot_index] & bit_mask) != 0) {
+ /* just return the first one which will be at least 16-bit */
+ return (int)pcm_format_value_map[table_index];
+ }
+ bit_mask <<= 1;
+ table_index++;
+ }
+ }
+
+ return PCM_FORMAT_INVALID;
+}
diff --git a/alsa_utils_legacy/alsa_logging.c b/alsa_utils_legacy/alsa_logging.c
new file mode 100644
index 0000000..e90797d
--- /dev/null
+++ b/alsa_utils_legacy/alsa_logging.c
@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "alsa_logging"
+/*#define LOG_NDEBUG 0*/
+
+#include <string.h>
+
+#include <log/log.h>
+
+#include "include/alsa_logging.h"
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+/*
+ * Logging
+ */
+void log_pcm_mask(const char* mask_name, struct pcm_mask* mask)
+{
+ const size_t num_slots = ARRAY_SIZE(mask->bits);
+ const size_t bits_per_slot = (sizeof(mask->bits[0]) * 8);
+ const size_t chars_per_slot = (bits_per_slot + 1); /* comma */
+
+ const size_t BUFF_SIZE =
+ (num_slots * chars_per_slot + 2 + 1); /* brackets and null-terminator */
+ char buff[BUFF_SIZE];
+ buff[0] = '\0';
+
+ size_t slot_index, bit_index;
+ strcat(buff, "[");
+ for (slot_index = 0; slot_index < num_slots; slot_index++) {
+ unsigned bit_mask = 1;
+ for (bit_index = 0; bit_index < bits_per_slot; bit_index++) {
+ strcat(buff, (mask->bits[slot_index] & bit_mask) != 0 ? "1" : "0");
+ bit_mask <<= 1;
+ }
+ if (slot_index < num_slots - 1) {
+ strcat(buff, ",");
+ }
+ }
+ strcat(buff, "]");
+
+ ALOGV("%s: mask:%s", mask_name, buff);
+}
+
+void log_pcm_params(struct pcm_params * alsa_hw_params)
+{
+ ALOGV("usb:audio_hw - PCM_PARAM_SAMPLE_BITS min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_SAMPLE_BITS));
+ ALOGV("usb:audio_hw - PCM_PARAM_FRAME_BITS min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_FRAME_BITS),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_FRAME_BITS));
+ log_pcm_mask("PCM_PARAM_FORMAT",
+ pcm_params_get_mask(alsa_hw_params, PCM_PARAM_FORMAT));
+ log_pcm_mask("PCM_PARAM_SUBFORMAT",
+ pcm_params_get_mask(alsa_hw_params, PCM_PARAM_SUBFORMAT));
+ ALOGV("usb:audio_hw - PCM_PARAM_CHANNELS min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS));
+ ALOGV("usb:audio_hw - PCM_PARAM_RATE min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE));
+ ALOGV("usb:audio_hw - PCM_PARAM_PERIOD_TIME min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIOD_TIME),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIOD_TIME));
+ ALOGV("usb:audio_hw - PCM_PARAM_PERIOD_SIZE min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIOD_SIZE),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIOD_SIZE));
+ ALOGV("usb:audio_hw - PCM_PARAM_PERIOD_BYTES min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIOD_BYTES),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIOD_BYTES));
+ ALOGV("usb:audio_hw - PCM_PARAM_PERIODS min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIODS),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIODS));
+ ALOGV("usb:audio_hw - PCM_PARAM_BUFFER_TIME min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_BUFFER_TIME),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_BUFFER_TIME));
+ ALOGV("usb:audio_hw - PCM_PARAM_BUFFER_SIZE min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_BUFFER_SIZE),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_BUFFER_SIZE));
+ ALOGV("usb:audio_hw - PCM_PARAM_BUFFER_BYTES min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_BUFFER_BYTES),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_BUFFER_BYTES));
+ ALOGV("usb:audio_hw - PCM_PARAM_TICK_TIME min:%u, max:%u",
+ pcm_params_get_min(alsa_hw_params, PCM_PARAM_TICK_TIME),
+ pcm_params_get_max(alsa_hw_params, PCM_PARAM_TICK_TIME));
+}
+
+void log_pcm_config(struct pcm_config * config, const char* label) {
+ ALOGV("log_pcm_config() - %s", label);
+ ALOGV(" channels:%d", config->channels);
+ ALOGV(" rate:%d", config->rate);
+ ALOGV(" period_size:%d", config->period_size);
+ ALOGV(" period_count:%d", config->period_count);
+ ALOGV(" format:%d", config->format);
+#if 0
+ /* Values to use for the ALSA start, stop and silence thresholds. Setting
+ * any one of these values to 0 will cause the default tinyalsa values to be
+ * used instead. Tinyalsa defaults are as follows.
+ *
+ * start_threshold : period_count * period_size
+ * stop_threshold : period_count * period_size
+ * silence_threshold : 0
+ */
+ unsigned int start_threshold;
+ unsigned int stop_threshold;
+ unsigned int silence_threshold;
+
+ /* Minimum number of frames available before pcm_mmap_write() will actually
+ * write into the kernel buffer. Only used if the stream is opened in mmap mode
+ * (pcm_open() called with PCM_MMAP flag set). Use 0 for default.
+ */
+ int avail_min;
+#endif
+}
diff --git a/alsa_utils_legacy/include/alsa_device_profile.h b/alsa_utils_legacy/include/alsa_device_profile.h
new file mode 100644
index 0000000..117d6fa
--- /dev/null
+++ b/alsa_utils_legacy/include/alsa_device_profile.h
@@ -0,0 +1,96 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_DEVICE_PROFILE_H
+#define ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_DEVICE_PROFILE_H
+
+#include <stdbool.h>
+
+#include <tinyalsa/asoundlib.h>
+
+#define MAX_PROFILE_FORMATS 6 /* We long support the 5 standard formats defined
+ * in asound.h, so we just need this to be 1 more
+ * than that */
+#define MAX_PROFILE_SAMPLE_RATES 14 /* this number needs to be 1 more than the number of
+ * sample rates in std_sample_rates[]
+ * (in alsa_device_profile.c) */
+#define MAX_PROFILE_CHANNEL_COUNTS 9 /* this number need to be 1 more than the number of
+ * standard channel formats in std_channel_counts[]
+ * (in alsa_device_profile.c) */
+
+#define DEFAULT_SAMPLE_RATE 44100
+#define DEFAULT_SAMPLE_FORMAT PCM_FORMAT_S16_LE
+#define DEFAULT_CHANNEL_COUNT 2
+
+typedef struct {
+ int card;
+ int device;
+ int direction; /* PCM_OUT or PCM_IN */
+
+ enum pcm_format formats[MAX_PROFILE_FORMATS];
+
+ /* note that this list is sorted highest rate to lowest */
+ unsigned sample_rates[MAX_PROFILE_SAMPLE_RATES];
+
+ unsigned channel_counts[MAX_PROFILE_CHANNEL_COUNTS];
+
+ bool is_valid;
+
+ /* read from the hardware device */
+ struct pcm_config default_config;
+
+ unsigned min_period_size;
+ unsigned max_period_size;
+
+ unsigned min_channel_count;
+ unsigned max_channel_count;
+} alsa_device_profile;
+
+void profile_init(alsa_device_profile* profile, int direction);
+bool profile_is_initialized(const alsa_device_profile* profile);
+bool profile_is_valid(const alsa_device_profile* profile);
+bool profile_is_cached_for(const alsa_device_profile* profile, int card, int device);
+void profile_decache(alsa_device_profile* profile);
+
+bool profile_read_device_info(alsa_device_profile* profile);
+
+/* Audio Config Strings Methods */
+char * profile_get_sample_rate_strs(const alsa_device_profile* profile);
+char * profile_get_format_strs(const alsa_device_profile* profile);
+char * profile_get_channel_count_strs(const alsa_device_profile* profile);
+
+/* Sample Rate Methods */
+unsigned profile_get_default_sample_rate(const alsa_device_profile* profile);
+unsigned profile_get_highest_sample_rate(const alsa_device_profile* profile);
+bool profile_is_sample_rate_valid(const alsa_device_profile* profile, unsigned rate);
+
+/* Format Methods */
+enum pcm_format profile_get_default_format(const alsa_device_profile* profile);
+bool profile_is_format_valid(const alsa_device_profile* profile, enum pcm_format fmt);
+
+/* Channel Methods */
+unsigned profile_get_default_channel_count(const alsa_device_profile* profile);
+unsigned profile_get_closest_channel_count(const alsa_device_profile* profile, unsigned count);
+bool profile_is_channel_count_valid(const alsa_device_profile* profile, unsigned count);
+
+/* Utility */
+unsigned profile_calc_min_period_size(const alsa_device_profile* profile, unsigned sample_rate);
+unsigned int profile_get_period_size(const alsa_device_profile* profile, unsigned sample_rate);
+
+/* Debugging */
+void profile_dump(const alsa_device_profile* profile, int fd);
+
+#endif /* ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_DEVICE_PROFILE_H */
diff --git a/alsa_utils_legacy/include/alsa_device_proxy.h b/alsa_utils_legacy/include/alsa_device_proxy.h
new file mode 100644
index 0000000..49f7019
--- /dev/null
+++ b/alsa_utils_legacy/include/alsa_device_proxy.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_DEVICE_PROXY_H
+#define ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_DEVICE_PROXY_H
+
+#include <tinyalsa/asoundlib.h>
+
+#include "alsa_device_profile.h"
+
+typedef struct {
+ const alsa_device_profile* profile;
+
+ struct pcm_config alsa_config;
+
+ struct pcm * pcm;
+
+ size_t frame_size; /* valid after proxy_prepare(), the frame size in bytes */
+ uint64_t transferred; /* the total frames transferred, not cleared on standby */
+} alsa_device_proxy;
+
+
+/* State */
+int proxy_prepare(alsa_device_proxy * proxy, const alsa_device_profile * profile,
+ struct pcm_config * config);
+int proxy_open(alsa_device_proxy * proxy);
+void proxy_close(alsa_device_proxy * proxy);
+int proxy_get_presentation_position(const alsa_device_proxy * proxy,
+ uint64_t *frames, struct timespec *timestamp);
+int proxy_get_capture_position(const alsa_device_proxy * proxy,
+ int64_t *frames, int64_t *time);
+
+/* Attributes */
+unsigned proxy_get_sample_rate(const alsa_device_proxy * proxy);
+enum pcm_format proxy_get_format(const alsa_device_proxy * proxy);
+unsigned proxy_get_channel_count(const alsa_device_proxy * proxy);
+unsigned int proxy_get_period_size(const alsa_device_proxy * proxy);
+unsigned proxy_get_latency(const alsa_device_proxy * proxy);
+
+/*
+ * Scans the provided list of sample rates and finds the first one that works.
+ *
+ * returns the index of the first rate for which the ALSA device can be opened.
+ * return negative value if none work or an error occurs.
+ */
+int proxy_scan_rates(alsa_device_proxy * proxy, const unsigned sample_rates[]);
+
+/* I/O */
+int proxy_write(alsa_device_proxy * proxy, const void *data, unsigned int count);
+int proxy_read(alsa_device_proxy * proxy, void *data, unsigned int count);
+
+/* Debugging */
+void proxy_dump(const alsa_device_proxy * proxy, int fd);
+
+#endif /* ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_DEVICE_PROXY_H */
diff --git a/alsa_utils_legacy/include/alsa_format.h b/alsa_utils_legacy/include/alsa_format.h
new file mode 100644
index 0000000..e07f836
--- /dev/null
+++ b/alsa_utils_legacy/include/alsa_format.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_FORMAT_H
+#define ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_FORMAT_H
+
+#include <system/audio.h>
+
+#include <tinyalsa/asoundlib.h>
+
+enum pcm_format get_pcm_format_for_mask(struct pcm_mask* mask);
+
+#endif /* ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_FORMAT_H */
diff --git a/alsa_utils_legacy/include/alsa_logging.h b/alsa_utils_legacy/include/alsa_logging.h
new file mode 100644
index 0000000..1b0731e
--- /dev/null
+++ b/alsa_utils_legacy/include/alsa_logging.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_LOGGING_H
+#define ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_LOGGING_H
+
+#include <tinyalsa/asoundlib.h>
+
+void log_pcm_mask(const char* mask_name, struct pcm_mask* mask);
+void log_pcm_params(struct pcm_params * alsa_hw_params);
+void log_pcm_config(struct pcm_config * config, const char* label);
+
+#endif /* ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_LOGGING_H */