diff --git a/audio/alsa_utils/Android.bp b/audio/alsa_utils/Android.bp new file mode 100644 index 0000000..402b76f --- /dev/null +++ b/audio/alsa_utils/Android.bp @@ -0,0 +1,65 @@ +// 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. + +package { + // http://go/android-license-faq + // A large-scale-change added 'default_applicable_licenses' to import + // the below license kinds from "system_media_license": + // SPDX-license-identifier-Apache-2.0 + default_applicable_licenses: ["system_media_license"], +} + +cc_defaults { + name: "libalsautils_defaults", + vendor_available: true, + srcs: [ + "alsa_device_profile.c", + "alsa_device_proxy.c", + "alsa_format.c", + "alsa_logging.c", + ], + export_include_dirs: ["include"], + header_libs: [ + "libaudio_system_headers", + ], + export_header_lib_headers: [ + "libaudio_system_headers", + ], + shared_libs: [ + "libaudioutils", + "libcutils", + "liblog", + ], + cflags: [ + "-Wall", + "-Werror", + "-Wno-unused-parameter", + ], +} + +cc_library { + name: "libalsautils", + defaults: ["libalsautils_defaults"], + shared_libs: [ + "libtinyalsa", + ], +} + +cc_library { + name: "libalsautilsv2", + defaults: ["libalsautils_defaults"], + shared_libs: [ + "libtinyalsav2", + ], +} diff --git a/audio/alsa_utils/alsa_device_profile.c b/audio/alsa_utils/alsa_device_profile.c new file mode 100644 index 0000000..f47ee09 --- /dev/null +++ b/audio/alsa_utils/alsa_device_profile.c @@ -0,0 +1,748 @@ +/* + * 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 +#include +#include +#include +#include + +#include + +#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; + profile->extra_latency_ms = 0; + + /* 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; + // This method tests whether a sample rate is supported by the USB device + // by attempting to open it. + // + // The profile default_config currently contains the minimum channel count. + // As some usb devices cannot sustain the sample rate across all its supported + // channel counts, we try the largest usable channel count. This is + // bounded by FCC_LIMIT. + // + // If config.channels > FCC_LIMIT then we still test it for sample rate compatibility. + // It is possible that the USB device does not support less than a certain number + // of channels, and that minimum number is > FCC_LIMIT. Then the default_config + // channels will be > FCC_LIMIT (and we still proceed with the test). + // + // For example, the FocusRite Scarlett 18i20 supports between 16 to 20 playback + // channels and between 14 to 18 capture channels. + // If FCC_LIMIT is 8, we still need to use and test 16 output channels for playback + // and 14 input channels for capture, as that will be the ALSA opening configuration. + // The Android USB audio HAL layer will automatically zero pad to accommodate the + // 16 playback or 14 capture channel configuration from the (up to FCC_LIMIT) + // channels delivered by AudioFlinger. + if (config.channels < FCC_LIMIT) { + config.channels = profile->max_channel_count; + if (config.channels > FCC_LIMIT) config.channels = FCC_LIMIT; + } + 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, + const 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. */ + // The order of this array controls the order for channel mask generation. + // In general, this is just counting from max to min not skipping anything, + // but need not be that way. + static const unsigned std_channel_counts[FCC_24] = { + 24, 23, 22, 21, 20, 19, 18, 17, + 16, 15, 14, 13, 12, 11, 10, 9, + 8, 7, 6, 5, 4, 3, 2, 1 + }; + + unsigned num_counts = 0; + unsigned index; + int max_allowed_index = -1; // index of maximum allowed channel count reported by device. + /* 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++) { + const unsigned test_count = std_channel_counts[index]; + /* TODO Do we want a channel counts test? */ + if (test_count <= FCC_LIMIT) { + if (test_count >= min && test_count <= max /* && + profile_test_channel_count(profile, channel_counts[index])*/) { + profile->channel_counts[num_counts++] = test_count; + } + if (max_allowed_index < 0 || + std_channel_counts[max_allowed_index] < test_count) { + max_allowed_index = index; + } + } + } + // if we have no match with the standard counts, we use the largest (preferred) std count. + // Note: the usb hal will adjust channel data properly to fit. + if (num_counts == 0 && max_allowed_index >= 0) { + ALOGW("usb device does not match std channel counts, setting to %d", + std_channel_counts[max_allowed_index]); + profile->channel_counts[num_counts++] = std_channel_counts[max_allowed_index]; + } + 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_fill_builtin_device_info(alsa_device_profile* profile, struct pcm_config* config, + unsigned buffer_frame_count) { + if (!profile_is_initialized(profile)) { + return false; + } + profile->extra_latency_ms = property_get_int32( + "ro.hardware.audio.tinyalsa.host_latency_ms", 0); + profile->default_config.channels = config->channels; + profile->default_config.rate = config->rate; + profile->default_config.format = config->format; + int period_count = property_get_int32( + "ro.hardware.audio.tinyalsa.period_count", DEFAULT_PERIOD_COUNT); + if (period_count <= 0) period_count = DEFAULT_PERIOD_COUNT; + profile->default_config.period_count = period_count; + int period_size_multiplier = property_get_int32( + "ro.hardware.audio.tinyalsa.period_size_multiplier", 1); + if (period_size_multiplier <= 0) period_size_multiplier = 1; + profile->default_config.period_size = + period_size_multiplier * buffer_frame_count / period_count; + profile->min_period_size = profile->max_period_size = profile->default_config.period_size; + profile->formats[0] = config->format; + profile->formats[1] = PCM_FORMAT_INVALID; + profile->channel_counts[0] = config->channels; + profile->channel_counts[1] = 0; + profile->min_channel_count = profile->max_channel_count = config->channels; + profile->sample_rates[0] = config->rate; + profile->sample_rates[1] = 0; + profile->is_valid = true; + return true; +} + +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 */ + const 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) +{ + // 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", + [4] = "AUDIO_CHANNEL_OUT_QUAD", + [6] = "AUDIO_CHANNEL_OUT_5POINT1", + [FCC_8] = "AUDIO_CHANNEL_OUT_7POINT1", + [FCC_12] = "AUDIO_CHANNEL_OUT_7POINT1POINT4", + [FCC_24] = "AUDIO_CHANNEL_OUT_22POINT2", + }; + + 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", + + [9] = "AUDIO_CHANNEL_INDEX_MASK_9", + [10] = "AUDIO_CHANNEL_INDEX_MASK_10", + [11] = "AUDIO_CHANNEL_INDEX_MASK_11", + [12] = "AUDIO_CHANNEL_INDEX_MASK_12", + [13] = "AUDIO_CHANNEL_INDEX_MASK_13", + [14] = "AUDIO_CHANNEL_INDEX_MASK_14", + [15] = "AUDIO_CHANNEL_INDEX_MASK_15", + [16] = "AUDIO_CHANNEL_INDEX_MASK_16", + + [17] = "AUDIO_CHANNEL_INDEX_MASK_17", + [18] = "AUDIO_CHANNEL_INDEX_MASK_18", + [19] = "AUDIO_CHANNEL_INDEX_MASK_19", + [20] = "AUDIO_CHANNEL_INDEX_MASK_20", + [21] = "AUDIO_CHANNEL_INDEX_MASK_21", + [22] = "AUDIO_CHANNEL_INDEX_MASK_22", + [23] = "AUDIO_CHANNEL_INDEX_MASK_23", + [24] = "AUDIO_CHANNEL_INDEX_MASK_24", + }; + + const bool isOutProfile = profile->direction == PCM_OUT; + + const char * const * const chans_strs = isOutProfile ? out_chans_strs : in_chans_strs; + size_t chans_strs_size = + isOutProfile ? ARRAY_SIZE(out_chans_strs) : ARRAY_SIZE(in_chans_strs); + if (chans_strs_size > FCC_LIMIT + 1) chans_strs_size = FCC_LIMIT + 1; // starts with 0. + + /* + * MAX_CHANNEL_NAME_LEN: The longest channel name so far is "AUDIO_CHANNEL_OUT_7POINT1POINT4" + * at 31 chars, add 1 for the "|" delimiter, so we allocate 48 chars to be safe. + * + * We allocate room for channel index and channel position strings (2x). + */ + const size_t MAX_CHANNEL_NAME_LEN = 48; + char buffer[MAX_CHANNEL_NAME_LEN * (FCC_LIMIT * 2) + 1]; + 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++) { + + if (channel_count > FCC_LIMIT) continue; + + /* 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; + } + + if (curStrLen != 0) 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; + } + + if (curStrLen != 0) 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; + fmtIndex < MAX_PROFILE_FORMATS && profile->formats[fmtIndex] != PCM_FORMAT_INVALID; + fmtIndex++) { + dprintf(fd, "%d ", profile->formats[fmtIndex]); + } + dprintf(fd, "\n"); + + /* sample rates */ + dprintf(fd, " Rates: "); + for (int rateIndex = 0; + rateIndex < MAX_PROFILE_SAMPLE_RATES && profile->sample_rates[rateIndex] != 0; + rateIndex++) { + dprintf(fd, "%u ", profile->sample_rates[rateIndex]); + } + dprintf(fd, "\n"); + + // channel counts + dprintf(fd, " Channel Counts: "); + for (int cntIndex = 0; + cntIndex < MAX_PROFILE_CHANNEL_COUNTS && profile->channel_counts[cntIndex] != 0; + 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/audio/alsa_utils/alsa_device_proxy.c b/audio/alsa_utils/alsa_device_proxy.c new file mode 100644 index 0000000..434914a --- /dev/null +++ b/audio/alsa_utils/alsa_device_proxy.c @@ -0,0 +1,409 @@ +/* + * 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 + +#include +#include +#include +#include + +#include + +#include "include/alsa_device_proxy.h" + +#include "include/alsa_logging.h" + +#define DEFAULT_PERIOD_SIZE 1024 + +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) + +// These must use the same clock. If we change ALSA clock to real time, the system +// clock must be updated, too. +#define ALSA_CLOCK_TYPE PCM_MONOTONIC +#define SYSTEM_CLOCK_TYPE CLOCK_MONOTONIC + +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, bool require_exact_match) +{ + int ret = 0; + + ALOGD("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 if (require_exact_match) { + ret = -EINVAL; + } 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 if (require_exact_match) { + ret = -EINVAL; + } 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 if (require_exact_match) { + ret = -EINVAL; + } 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 = DEFAULT_PERIOD_COUNT; + } + + 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, require_exact_match); + 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_prepare_from_default_config(alsa_device_proxy * proxy, + const alsa_device_profile * profile) +{ + ALOGD("proxy_prepare_from_default_config(c:%d, d:%d)", profile->card, profile->device); + + proxy->profile = profile; + +#ifdef LOG_PCM_PARAMS + log_pcm_config(&profile->default_config, "proxy_prepare_from_default_config()"); +#endif + + proxy->alsa_config.format = profile->default_config.format; + proxy->alsa_config.rate = profile->default_config.rate; + proxy->alsa_config.channels = profile->default_config.channels; + proxy->alsa_config.period_count = profile->default_config.period_count; + proxy->alsa_config.period_size = profile->default_config.period_size; + proxy->pcm = NULL; + enum pcm_format format = profile->default_config.format; + if (format >= 0 && (size_t)format < ARRAY_SIZE(format_byte_size_map)) { + proxy->frame_size = format_byte_size_map[format] * proxy->alsa_config.channels; + } else { + proxy->frame_size = 1; + } + + return 0; +} + +int proxy_open(alsa_device_proxy * proxy) +{ + const alsa_device_profile* profile = proxy->profile; + ALOGD("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 | ALSA_CLOCK_TYPE, &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) +{ + ALOGD("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; +} + +static unsigned int proxy_get_extra_latency_ms(const alsa_device_proxy * proxy) +{ + return proxy->profile->extra_latency_ms; +} + +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) + proxy_get_extra_latency_ms(proxy); +} + +int proxy_get_presentation_position(const alsa_device_proxy * proxy, + uint64_t *frames, struct timespec *timestamp) +{ + int ret = -EPERM; // -1 + unsigned int avail; + struct timespec alsaTs; + if (proxy->pcm != NULL + && pcm_get_htimestamp(proxy->pcm, &avail, &alsaTs) == 0) { + const size_t kernel_buffer_size = pcm_get_buffer_size(proxy->pcm); + if (avail > kernel_buffer_size) { + // pcm_get_htimestamp() computes the available frames by comparing the ALSA driver + // hw_ptr and the appl_ptr levels. In underrun, the hw_ptr may keep running and report + // an excessively large number available number. + ALOGW("available frames(%u) > buffer size(%zu), clamped", avail, kernel_buffer_size); + avail = kernel_buffer_size; + } + if (alsaTs.tv_sec != 0 || alsaTs.tv_nsec != 0) { + *timestamp = alsaTs; + } else { // If ALSA returned a zero timestamp, do not use it. + clock_gettime(SYSTEM_CLOCK_TYPE, timestamp); + } + 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; + if (proxy->pcm != NULL + && pcm_get_htimestamp(proxy->pcm, &avail, ×tamp) == 0) { + if (timestamp.tv_sec == 0 && timestamp.tv_nsec == 0) { + // If ALSA returned a zero timestamp, do not use it. + clock_gettime(SYSTEM_CLOCK_TYPE, ×tamp); + } + uint64_t framesTemp = proxy->transferred + avail; + if (framesTemp > INT64_MAX) { + framesTemp -= INT64_MAX; + } + *frames = framesTemp; + *time = audio_utils_ns_from_timespec(×tamp); + ret = 0; + } + return ret; +} + +int proxy_stop(alsa_device_proxy * proxy) +{ + int ret = -ENOSYS; + if (proxy->pcm != NULL) ret = pcm_stop(proxy->pcm); + return ret; +} + +/* + * I/O + */ +int proxy_write(alsa_device_proxy * proxy, const void *data, unsigned int count) +{ + return proxy_write_with_retries(proxy, data, count, 1); +} + +int proxy_write_with_retries( + alsa_device_proxy * proxy, const void *data, unsigned int count, int tries) +{ + while (true) { + --tries; + const int ret = pcm_write(proxy->pcm, data, count); + if (ret == 0) { + proxy->transferred += count / proxy->frame_size; + return 0; + } else if (tries > 0 && (ret == -EIO || ret == -EAGAIN)) { + continue; + } + return ret; + } +} + +int proxy_read(alsa_device_proxy * proxy, void *data, unsigned int count) +{ + return proxy_read_with_retries(proxy, data, count, 1); +} + +int proxy_read_with_retries(alsa_device_proxy * proxy, void *data, unsigned int count, int tries) +{ + while (true) { + --tries; + const int ret = pcm_read(proxy->pcm, data, count); + if (ret == 0) { + proxy->transferred += count / proxy->frame_size; + return 0; + } else if (tries > 0 && (ret == -EIO || ret == -EAGAIN)) { + continue; + } + 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[], + bool require_exact_match) { + 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) { + if (require_exact_match && alsa_config.rate != sample_rates[rate_index]) { + rate_index++; + continue; + } + alsa_config.rate = sample_rates[rate_index]; + alsa_pcm = pcm_open(profile->card, profile->device, + profile->direction | ALSA_CLOCK_TYPE, &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/audio/alsa_utils/alsa_format.c b/audio/alsa_utils/alsa_format.c new file mode 100644 index 0000000..ef4d7a8 --- /dev/null +++ b/audio/alsa_utils/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 + +#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(const 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/audio/alsa_utils/alsa_logging.c b/audio/alsa_utils/alsa_logging.c new file mode 100644 index 0000000..d609867 --- /dev/null +++ b/audio/alsa_utils/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 + +#include + +#include "include/alsa_logging.h" + +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) + +/* + * Logging + */ +void log_pcm_mask(const char* mask_name, const 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(const 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(const 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/audio/alsa_utils/include/alsa_device_profile.h b/audio/alsa_utils/include/alsa_device_profile.h new file mode 100644 index 0000000..24b6665 --- /dev/null +++ b/audio/alsa_utils/include/alsa_device_profile.h @@ -0,0 +1,101 @@ +/* + * 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 +#include +#include + +#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 (FCC_LIMIT + 1) + /* 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 +#define DEFAULT_PERIOD_COUNT 4 + +typedef struct { + int card; + int device; + int direction; /* PCM_OUT or PCM_IN */ + int extra_latency_ms; /* any extra latency in addition to the buffer */ + + 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_fill_builtin_device_info(alsa_device_profile* profile, struct pcm_config* config, + unsigned buffer_frame_count); +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/audio/alsa_utils/include/alsa_device_proxy.h b/audio/alsa_utils/include/alsa_device_proxy.h new file mode 100644 index 0000000..eba9f05 --- /dev/null +++ b/audio/alsa_utils/include/alsa_device_proxy.h @@ -0,0 +1,76 @@ +/* + * 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 + +#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, bool require_exact_match); +int proxy_prepare_from_default_config( + alsa_device_proxy * proxy, const alsa_device_profile * profile); +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); +int proxy_stop(alsa_device_proxy * proxy); + +/* 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[], + bool require_exact_match); + +/* I/O */ +int proxy_write(alsa_device_proxy * proxy, const void *data, unsigned int count); +int proxy_write_with_retries( + alsa_device_proxy * proxy, const void *data, unsigned int count, int tries); +int proxy_read(alsa_device_proxy * proxy, void *data, unsigned int count); +int proxy_read_with_retries( + alsa_device_proxy * proxy, void *data, unsigned int count, int tries); + +/* Debugging */ +void proxy_dump(const alsa_device_proxy * proxy, int fd); + +#endif /* ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_DEVICE_PROXY_H */ diff --git a/audio/alsa_utils/include/alsa_format.h b/audio/alsa_utils/include/alsa_format.h new file mode 100644 index 0000000..fb39f53 --- /dev/null +++ b/audio/alsa_utils/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 + +#include + +enum pcm_format get_pcm_format_for_mask(const struct pcm_mask* mask); + +#endif /* ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_FORMAT_H */ diff --git a/audio/alsa_utils/include/alsa_logging.h b/audio/alsa_utils/include/alsa_logging.h new file mode 100644 index 0000000..1f58184 --- /dev/null +++ b/audio/alsa_utils/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 + +void log_pcm_mask(const char* mask_name, const struct pcm_mask* mask); +void log_pcm_params(const struct pcm_params * alsa_hw_params); +void log_pcm_config(const struct pcm_config * config, const char* label); + +#endif /* ANDROID_SYSTEM_MEDIA_ALSA_UTILS_ALSA_LOGGING_H */