Initial implementation of usb audio I/O
Change-Id: Ib82783f0b25887e2d34a24fde346cee5003d5b89
diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c
index 24a2d63..afe56b2 100644
--- a/modules/usbaudio/audio_hw.c
+++ b/modules/usbaudio/audio_hw.c
@@ -33,65 +33,270 @@
#include <tinyalsa/asoundlib.h>
-struct pcm_config pcm_config = {
+/* This is the default configuration to hand to The Framework on the initial
+ * adev_open_output_stream(). Actual device attributes will be used on the subsequent
+ * adev_open_output_stream() after the card and device number have been set in out_set_parameters()
+ */
+#define OUT_PERIOD_SIZE 1024
+#define OUT_PERIOD_COUNT 4
+#define OUT_SAMPLING_RATE 44100
+
+struct pcm_config default_alsa_out_config = {
.channels = 2,
- .rate = 44100,
- .period_size = 1024,
- .period_count = 4,
+ .rate = OUT_SAMPLING_RATE,
+ .period_size = OUT_PERIOD_SIZE,
+ .period_count = OUT_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
};
+/*
+ * Input defaults. See comment above.
+ */
+#define IN_PERIOD_SIZE 1024
+#define IN_PERIOD_COUNT 4
+#define IN_SAMPLING_RATE 44100
+
+struct pcm_config default_alsa_in_config = {
+ .channels = 2,
+ .rate = IN_SAMPLING_RATE,
+ .period_size = IN_PERIOD_SIZE,
+ .period_count = IN_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 1,
+ .stop_threshold = (IN_PERIOD_SIZE * IN_PERIOD_COUNT),
+};
+
struct audio_device {
struct audio_hw_device hw_device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
- int card;
- int device;
+
+ /* output */
+ int out_card;
+ int out_device;
+
+ /* input */
+ int in_card;
+ int in_device;
+
bool standby;
};
struct stream_out {
struct audio_stream_out stream;
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ struct pcm *pcm; /* state of the stream */
+ bool standby;
+
+ struct audio_device *dev; /* hardware information */
+
+ void * conversion_buffer; /* any conversions are put into here
+ * they could come from here too if
+ * there was a previous conversion */
+ size_t conversion_buffer_size; /* in bytes */
+};
+
+/*
+ * Output Configuration Cache
+ * FIXME(pmclean) This is not rentrant. Should probably be moved into the stream structure
+ * but that will involve changes in The Framework.
+ */
+static struct pcm_config cached_output_hardware_config;
+static bool output_hardware_config_is_cached = false;
+
+struct stream_in {
+ struct audio_stream_in stream;
+
pthread_mutex_t lock; /* see note below on mutex acquisition order */
struct pcm *pcm;
bool standby;
+ struct pcm_config alsa_pcm_config;
+
struct audio_device *dev;
+
+ struct audio_config hal_pcm_config;
+
+ unsigned int requested_rate;
+// struct resampler_itfe *resampler;
+// struct resampler_buffer_provider buf_provider;
+ int16_t *buffer;
+ size_t buffer_size;
+ size_t frames_in;
+ int read_status;
};
+/*
+ * Utility
+ */
+/*
+ * Translates from ALSA format ID to ANDROID_AUDIO_CORE format ID
+ * (see master/system/core/include/core/audio.h)
+ * TODO(pmclean) Replace with audio_format_from_pcm_format() (in hardware/audio_alsaops.h).
+ * post-integration.
+ */
+static audio_format_t alsa_to_fw_format_id(int alsa_fmt_id)
+{
+ switch (alsa_fmt_id) {
+ case PCM_FORMAT_S8:
+ return AUDIO_FORMAT_PCM_8_BIT;
+
+ case PCM_FORMAT_S24_3LE:
+ //TODO(pmclean) make sure this is the 'right' sort of 24-bit
+ return AUDIO_FORMAT_PCM_8_24_BIT;
+
+ case PCM_FORMAT_S32_LE:
+ case PCM_FORMAT_S24_LE:
+ return AUDIO_FORMAT_PCM_32_BIT;
+ }
+
+ return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+/*
+ * Data Conversions
+ */
+/*
+ * Convert a buffer of PCM16LE samples to packed (3-byte) PCM24LE samples.
+ * in_buff points to the buffer of PCM16 samples
+ * num_in_samples size of input buffer in SAMPLES
+ * out_buff points to the buffer to receive converted PCM24 LE samples.
+ * returns the number of BYTES of output data.
+ * We are doing this since we *always* present to The Framework as A PCM16LE device, but need to
+ * support PCM24_3LE (24-bit, packed).
+ * NOTE: we're just filling the low-order byte of the PCM24LE samples with 0.
+ * TODO(pmclean, hung) Move this to a utilities module.
+ */
+static size_t convert_16_to_24_3(unsigned short * in_buff,
+ size_t num_in_samples,
+ unsigned char * out_buff) {
+ /*
+ * Move from back to front so that the conversion can be done in-place
+ * i.e. in_buff == out_buff
+ */
+ int in_buff_size_in_bytes = num_in_samples * 2;
+ /* we need 3 bytes in the output for every 2 bytes in the input */
+ int out_buff_size_in_bytes = ((3 * in_buff_size_in_bytes) / 2);
+ unsigned char* dst_ptr = out_buff + out_buff_size_in_bytes - 1;
+ int src_smpl_index;
+ unsigned char* src_ptr = ((unsigned char *)in_buff) + in_buff_size_in_bytes - 1;
+ for (src_smpl_index = 0; src_smpl_index < num_in_samples; src_smpl_index++) {
+ *dst_ptr-- = *src_ptr--; /* hi-byte */
+ *dst_ptr-- = *src_ptr--; /* low-byte */
+ *dst_ptr-- = 0; /* zero-byte */
+ }
+
+ /* return number of *bytes* generated */
+ return out_buff_size_in_bytes;
+}
+
+/*
+ * Convert a buffer of 2-channel PCM16 samples to 4-channel PCM16 channels
+ * in_buff points to the buffer of PCM16 samples
+ * num_in_samples size of input buffer in SAMPLES
+ * out_buff points to the buffer to receive converted PCM16 samples.
+ * returns the number of BYTES of output data.
+ * NOTE channels 3 & 4 are filled with silence.
+ * We are doing this since we *always* present to The Framework as STEREO device, but need to
+ * support 4-channel devices.
+ * TODO(pmclean, hung) Move this to a utilities module.
+ */
+static size_t convert_2chan16_to_4chan16(unsigned short* in_buff,
+ size_t num_in_samples,
+ unsigned short* out_buff) {
+ /*
+ * Move from back to front so that the conversion can be done in-place
+ * i.e. in_buff == out_buff
+ */
+ int out_buff_size = num_in_samples * 2;
+ unsigned short* dst_ptr = out_buff + out_buff_size - 1;
+ int src_index;
+ unsigned short* src_ptr = in_buff + num_in_samples - 1;
+ for (src_index = 0; src_index < num_in_samples; src_index += 2) {
+ *dst_ptr-- = 0; /* chan 4 */
+ *dst_ptr-- = 0; /* chan 3 */
+ *dst_ptr-- = *src_ptr--; /* chan 2 */
+ *dst_ptr-- = *src_ptr--; /* chan 1 */
+ }
+
+ /* return number of *bytes* generated */
+ return out_buff_size * 2;
+}
+
+/*
+ * ALSA Utilities
+ */
+/*
+ * gets the ALSA bit-format flag from a bits-per-sample value.
+ * TODO(pmclean, hung) Move this to a utilities module.
+ */
+static int bits_to_alsa_format(int bits_per_sample, int default_format)
+{
+ enum pcm_format format;
+ for (format = PCM_FORMAT_S16_LE; format < PCM_FORMAT_MAX; format++) {
+ if (pcm_format_to_bits(format) == bits_per_sample) {
+ return format;
+ }
+ }
+ return default_format;
+}
+
+/*
+ * Reads and decodes configuration info from the specified ALSA card/device
+ */
+static int read_alsa_device_config(int card, int device, int io_type, struct pcm_config * config)
+{
+ ALOGV("usb:audio_hw - read_alsa_device_config(card:%d device:%d)", card, device);
+
+ if (card < 0 || device < 0) {
+ return -EINVAL;
+ }
+
+ struct pcm_params * alsa_hw_params = pcm_params_get(card, device, io_type);
+ if (alsa_hw_params == NULL) {
+ return -EINVAL;
+ }
+
+ /*
+ * This Logging will be useful when testing new USB devices.
+ */
+ /* ALOGV("usb:audio_hw - PCM_PARAM_SAMPLE_BITS min:%d, max:%d", 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:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_FRAME_BITS), pcm_params_get_max(alsa_hw_params, PCM_PARAM_FRAME_BITS)); */
+ /* ALOGV("usb:audio_hw - PCM_PARAM_CHANNELS min:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", 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:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_TICK_TIME), pcm_params_get_max(alsa_hw_params, PCM_PARAM_TICK_TIME)); */
+
+ config->channels = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+ config->rate = pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE);
+ config->period_size = pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIODS);
+ config->period_count = pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIODS);
+
+ int bits_per_sample = pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+ config->format = bits_to_alsa_format(bits_per_sample, PCM_FORMAT_S16_LE);
+
+ return 0;
+}
+
+/*
+ * HAl Functions
+ */
/**
* NOTE: when multiple mutexes have to be acquired, always respect the
* following order: hw device > out stream
*/
/* Helper functions */
-
-/* must be called with hw device and output stream mutexes locked */
-static int start_output_stream(struct stream_out *out)
-{
- struct audio_device *adev = out->dev;
- int i;
-
- if ((adev->card < 0) || (adev->device < 0))
- return -EINVAL;
-
- out->pcm = pcm_open(adev->card, adev->device, PCM_OUT, &pcm_config);
-
- if (out->pcm && !pcm_is_ready(out->pcm)) {
- ALOGE("pcm_open() failed: %s", pcm_get_error(out->pcm));
- pcm_close(out->pcm);
- return -ENOMEM;
- }
-
- return 0;
-}
-
-/* API functions */
-
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
- return pcm_config.rate;
+ return cached_output_hardware_config.rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
@@ -101,17 +306,22 @@
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
- return pcm_config.period_size *
- audio_stream_frame_size((struct audio_stream *)stream);
+ return cached_output_hardware_config.period_size * audio_stream_frame_size(stream);
}
static uint32_t out_get_channels(const struct audio_stream *stream)
{
+ // Always Stero for now. We will do *some* conversions in this HAL.
+ // TODO(pmclean) When AudioPolicyManager & AudioFlinger supports arbitrary channels
+ // rewrite this to return the ACTUAL channel format
return AUDIO_CHANNEL_OUT_STEREO;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
+ // Always return 16-bit PCM. We will do *some* conversions in this HAL.
+ // TODO(pmclean) When AudioPolicyManager & AudioFlinger supports arbitrary PCM formats
+ // rewrite this to return the ACTUAL data format
return AUDIO_FORMAT_PCM_16_BIT;
}
@@ -146,39 +356,122 @@
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
+ ALOGV("usb:audio_hw::out out_set_parameters() keys:%s", kvpairs);
+
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
struct str_parms *parms;
char value[32];
- int ret;
+ int param_val;
int routing = 0;
+ int ret_value = 0;
parms = str_parms_create_str(kvpairs);
pthread_mutex_lock(&adev->lock);
- ret = str_parms_get_str(parms, "card", value, sizeof(value));
- if (ret >= 0)
- adev->card = atoi(value);
+ bool recache_device_params = false;
+ param_val = str_parms_get_str(parms, "card", value, sizeof(value));
+ if (param_val >= 0) {
+ adev->out_card = atoi(value);
+ recache_device_params = true;
+ }
- ret = str_parms_get_str(parms, "device", value, sizeof(value));
- if (ret >= 0)
- adev->device = atoi(value);
+ param_val = str_parms_get_str(parms, "device", value, sizeof(value));
+ if (param_val >= 0) {
+ adev->out_device = atoi(value);
+ recache_device_params = true;
+ }
+
+ if (recache_device_params && adev->out_card >= 0 && adev->out_device >= 0) {
+ ret_value = read_alsa_device_config(adev->out_card, adev->out_device, PCM_OUT,
+ &(cached_output_hardware_config));
+ output_hardware_config_is_cached = (ret_value == 0);
+ }
pthread_mutex_unlock(&adev->lock);
str_parms_destroy(parms);
- return 0;
+ return ret_value;
}
-static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
-{
- return strdup("");
+//TODO(pmclean) it seems like both out_get_parameters() and in_get_parameters()
+// could be written in terms of a get_device_parameters(io_type)
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys) {
+ struct stream_out *out = (struct stream_out *) stream;
+ struct audio_device *adev = out->dev;
+
+ unsigned min, max;
+
+ struct str_parms *query = str_parms_create_str(keys);
+ struct str_parms *result = str_parms_create();
+
+ int num_written = 0;
+ char buffer[256];
+ int buffer_size = sizeof(buffer) / sizeof(buffer[0]);
+ char* result_str = NULL;
+
+ struct pcm_params * alsa_hw_params = pcm_params_get(adev->out_card, adev->out_device, PCM_OUT);
+
+ // These keys are from hardware/libhardware/include/audio.h
+ // supported sample rates
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
+ // pcm_hw_params doesn't have a list of supported samples rates, just a min and a max, so
+ // if they are different, return a list containing those two values, otherwise just the one.
+ min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE);
+ max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE);
+ num_written = snprintf(buffer, buffer_size, "%d", min);
+ if (min != max) {
+ snprintf(buffer + num_written, buffer_size - num_written, "|%d",
+ max);
+ }
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
+ buffer);
+ } // AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES
+
+ // supported channel counts
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
+ // Similarly for output channels count
+ min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+ max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS);
+ num_written = snprintf(buffer, buffer_size, "%d", min);
+ if (min != max) {
+ snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+ }
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, buffer);
+ } // AUDIO_PARAMETER_STREAM_SUP_CHANNELS
+
+ // supported sample formats
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+ // Similarly for output channels count
+ //TODO(pmclean): this is wrong.
+ min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+ max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+ num_written = snprintf(buffer, buffer_size, "%d", min);
+ if (min != max) {
+ snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+ }
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS, buffer);
+ } // AUDIO_PARAMETER_STREAM_SUP_FORMATS
+
+ result_str = str_parms_to_str(result);
+
+ // done with these...
+ str_parms_destroy(query);
+ str_parms_destroy(result);
+
+ return result_str;
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
- return (pcm_config.period_size * pcm_config.period_count * 1000) /
- out_get_sample_rate(&stream->common);
+ struct stream_out *out = (struct stream_out *)stream;
+
+ //TODO(pmclean): Do we need a term here for the USB latency
+ // (as reported in the USB descriptors)?
+ uint32_t latency = (cached_output_hardware_config.period_size *
+ cached_output_hardware_config.period_count * 1000) / out_get_sample_rate(&stream->common);
+ return latency;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
@@ -187,8 +480,41 @@
return -ENOSYS;
}
-static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
- size_t bytes)
+/* must be called with hw device and output stream mutexes locked */
+static int start_output_stream(struct stream_out *out)
+{
+ struct audio_device *adev = out->dev;
+ int return_val = 0;
+
+ ALOGV("usb:audio_hw::out start_output_stream(card:%d device:%d)",
+ adev->out_card, adev->out_device);
+
+ out->pcm = pcm_open(adev->out_card, adev->out_device, PCM_OUT, &cached_output_hardware_config);
+ if (out->pcm == NULL) {
+ return -ENOMEM;
+ }
+
+ if (out->pcm && !pcm_is_ready(out->pcm)) {
+ ALOGE("audio_hw audio_hw pcm_open() failed: %s", pcm_get_error(out->pcm));
+ pcm_close(out->pcm);
+ return -ENOMEM;
+ }
+
+ // Setup conversion buffer
+ size_t buffer_size = out_get_buffer_size(&(out->stream.common));
+
+ // computer maximum potential buffer size.
+ // * 2 for stereo -> quad conversion
+ // * 3/2 for 16bit -> 24 bit conversion
+ //TODO(pmclean) - remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
+ // (and do these conversions themselves)
+ out->conversion_buffer_size = (buffer_size * 3 * 2) / 2;
+ out->conversion_buffer = realloc(out->conversion_buffer, out->conversion_buffer_size);
+
+ return 0;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
{
int ret;
struct stream_out *out = (struct stream_out *)stream;
@@ -203,7 +529,45 @@
out->standby = false;
}
- pcm_write(out->pcm, (void *)buffer, bytes);
+ void * write_buff = buffer;
+ int num_write_buff_bytes = bytes;
+
+ /*
+ * Num Channels conversion
+ */
+ int num_device_channels = cached_output_hardware_config.channels;
+ int num_req_channels = 2; /* always, for now */
+ if (num_device_channels != num_req_channels && num_device_channels == 4) {
+ num_write_buff_bytes =
+ convert_2chan16_to_4chan16(write_buff, num_write_buff_bytes / 2,
+ out->conversion_buffer);
+ write_buff = out->conversion_buffer;
+ }
+
+ /*
+ * 16 vs 24-bit logic here
+ */
+ switch (cached_output_hardware_config.format) {
+ case PCM_FORMAT_S16_LE:
+ // the output format is the same as the input format, so just write it out
+ break;
+
+ case PCM_FORMAT_S24_3LE:
+ // 16-bit LE2 - 24-bit LE3
+ num_write_buff_bytes =
+ convert_16_to_24_3(write_buff, num_write_buff_bytes / 2, out->conversion_buffer);
+ write_buff = out->conversion_buffer;
+ break;
+
+ default:
+ // hmmmmm.....
+ ALOGV("usb:Unknown Format!!!");
+ break;
+ }
+
+ if (write_buff != NULL && num_write_buff_bytes != 0) {
+ pcm_write(out->pcm, write_buff, num_write_buff_bytes);
+ }
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
@@ -250,14 +614,18 @@
struct audio_config *config,
struct audio_stream_out **stream_out)
{
+ ALOGV("usb:audio_hw::out adev_open_output_stream() handle:0x%X, devices:0x%X, flags:0x%X",
+ handle, devices, flags);
+
struct audio_device *adev = (struct audio_device *)dev;
+
struct stream_out *out;
- int ret;
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
if (!out)
return -ENOMEM;
+ // setup function pointers
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
@@ -278,30 +646,64 @@
out->dev = adev;
- config->format = out_get_format(&out->stream.common);
- config->channel_mask = out_get_channels(&out->stream.common);
- config->sample_rate = out_get_sample_rate(&out->stream.common);
+ if (output_hardware_config_is_cached) {
+ config->sample_rate = cached_output_hardware_config.rate;
+
+ config->format = alsa_to_fw_format_id(cached_output_hardware_config.format);
+ if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+ // Always report PCM16 for now. AudioPolicyManagerBase/AudioFlinger dont' understand
+ // formats with more other format, so we won't get chosen (say with a 24bit DAC).
+ //TODO(pmclean) remove this when the above restriction is removed.
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ }
+
+ config->channel_mask =
+ audio_channel_out_mask_from_count(cached_output_hardware_config.channels);
+ if (config->channel_mask != AUDIO_CHANNEL_OUT_STEREO) {
+ // Always report STEREO for now. AudioPolicyManagerBase/AudioFlinger dont' understand
+ // formats with more channels, so we won't get chosen (say with a 4-channel DAC).
+ //TODO(pmclean) remove this when the above restriction is removed.
+ config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ }
+ } else {
+ cached_output_hardware_config = default_alsa_out_config;
+
+ config->format = out_get_format(&out->stream.common);
+ config->channel_mask = out_get_channels(&out->stream.common);
+ config->sample_rate = out_get_sample_rate(&out->stream.common);
+ }
+ ALOGV("usb:audio_hw config->sample_rate:%d", config->sample_rate);
+ ALOGV("usb:audio_hw config->format:0x%X", config->format);
+ ALOGV("usb:audio_hw config->channel_mask:0x%X", config->channel_mask);
+
+ out->conversion_buffer = NULL;
+ out->conversion_buffer_size = 0;
out->standby = true;
- adev->card = -1;
- adev->device = -1;
-
*stream_out = &out->stream;
return 0;
err_open:
free(out);
*stream_out = NULL;
- return ret;
+ return -ENOSYS;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
+ ALOGV("usb:audio_hw::out adev_close_output_stream()");
struct stream_out *out = (struct stream_out *)stream;
+ //TODO(pmclean) why are we doing this when stream get's freed at the end
+ // because it closes the pcm device
out_standby(&stream->common);
+
+ free(out->conversion_buffer);
+ out->conversion_buffer = NULL;
+ out->conversion_buffer_size = 0;
+
free(stream);
}
@@ -352,13 +754,264 @@
return 0;
}
+/* Helper functions */
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ return in->alsa_pcm_config.rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+ return -ENOSYS;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ size_t buff_size =
+ in->alsa_pcm_config.period_size
+ * audio_stream_frame_size((struct audio_stream *)stream);
+ return buff_size;
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ //TODO(pmclean) this should be done with a num_channels_to_alsa_channels()
+ return in->alsa_pcm_config.channels == 2
+ ? AUDIO_CHANNEL_IN_STEREO : AUDIO_CHANNEL_IN_MONO;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+ // just report 16-bit, pcm for now.
+ return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+ return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+ ALOGV("-pcm-audio_hw::in in_standby() [Not Implemented]");
+ return 0;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+ return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ ALOGV("Vaudio_hw::in in_set_parameters() keys:%s", kvpairs);
+
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = in->dev;
+ struct str_parms *parms;
+ char value[32];
+ int param_val;
+ int routing = 0;
+ int ret_value = 0;
+
+ parms = str_parms_create_str(kvpairs);
+ pthread_mutex_lock(&adev->lock);
+
+ // Card/Device
+ param_val = str_parms_get_str(parms, "card", value, sizeof(value));
+ if (param_val >= 0) {
+ adev->in_card = atoi(value);
+ }
+
+ param_val = str_parms_get_str(parms, "device", value, sizeof(value));
+ if (param_val >= 0) {
+ adev->in_device = atoi(value);
+ }
+
+ if (adev->in_card >= 0 && adev->in_device >= 0) {
+ ret_value = read_alsa_device_config(adev->in_card, adev->in_device, PCM_IN, &(in->alsa_pcm_config));
+ }
+
+ pthread_mutex_unlock(&adev->lock);
+ str_parms_destroy(parms);
+
+ return ret_value;
+}
+
+//TODO(pmclean) it seems like both out_get_parameters() and in_get_parameters()
+// could be written in terms of a get_device_parameters(io_type)
+
+static char * in_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+ ALOGV("usb:audio_hw::in in_get_parameters() keys:%s", keys);
+
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = in->dev;
+
+ struct pcm_params * alsa_hw_params = pcm_params_get(adev->in_card, adev->in_device, PCM_IN);
+ if (alsa_hw_params == NULL)
+ return strdup("");
+
+ struct str_parms *query = str_parms_create_str(keys);
+ struct str_parms *result = str_parms_create();
+
+ int num_written = 0;
+ char buffer[256];
+ int buffer_size = sizeof(buffer)/sizeof(buffer[0]);
+ char* result_str = NULL;
+
+ unsigned min, max;
+
+ // These keys are from hardware/libhardware/include/audio.h
+ // supported sample rates
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
+ // pcm_hw_params doesn't have a list of supported samples rates, just a min and a max, so
+ // if they are different, return a list containing those two values, otherwise just the one.
+ min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE);
+ max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE);
+ num_written = snprintf(buffer, buffer_size, "%d", min);
+ if (min != max) {
+ snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+ }
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SAMPLING_RATE, buffer);
+ } // AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES
+
+ // supported channel counts
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
+ // Similarly for output channels count
+ min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+ max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS);
+ num_written = snprintf(buffer, buffer_size, "%d", min);
+ if (min != max) {
+ snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+ }
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_CHANNELS, buffer);
+ } // AUDIO_PARAMETER_STREAM_SUP_CHANNELS
+
+ // supported sample formats
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+ //TODO(pmclean): this is wrong.
+ min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+ max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+ num_written = snprintf(buffer, buffer_size, "%d", min);
+ if (min != max) {
+ snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+ }
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS, buffer);
+ } // AUDIO_PARAMETER_STREAM_SUP_FORMATS
+
+ result_str = str_parms_to_str(result);
+
+ // done with these...
+ str_parms_destroy(query);
+ str_parms_destroy(result);
+
+ return result_str;
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+ return 0;
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+ return 0;
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain) {
+ return 0;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes) {
+ struct stream_in * in = (struct stream_in *)stream;
+
+ int err = pcm_read(in->pcm, buffer, bytes);
+
+ return err == 0 ? bytes : 0;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) {
+ return 0;
+}
+
static int adev_open_input_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
- struct audio_config *config,
+ struct audio_config *hal_config,
struct audio_stream_in **stream_in)
{
- return -ENOSYS;
+ ALOGV("usb:audio_hw::in adev_open_input_stream() rate:%d, chanMask:0x%X, fmt:%d",
+ hal_config->sample_rate,
+ hal_config->channel_mask,
+ hal_config->format);
+
+ struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
+ if (in == NULL)
+ return -ENOMEM;
+
+ // setup function pointers
+ in->stream.common.get_sample_rate = in_get_sample_rate;
+ in->stream.common.set_sample_rate = in_set_sample_rate;
+ in->stream.common.get_buffer_size = in_get_buffer_size;
+ in->stream.common.get_channels = in_get_channels;
+ in->stream.common.get_format = in_get_format;
+ in->stream.common.set_format = in_set_format;
+ in->stream.common.standby = in_standby;
+ in->stream.common.dump = in_dump;
+ in->stream.common.set_parameters = in_set_parameters;
+ in->stream.common.get_parameters = in_get_parameters;
+ in->stream.common.add_audio_effect = in_add_audio_effect;
+ in->stream.common.remove_audio_effect = in_remove_audio_effect;
+
+ in->stream.set_gain = in_set_gain;
+ in->stream.read = in_read;
+ in->stream.get_input_frames_lost = in_get_input_frames_lost;
+
+ struct audio_device *adev = (struct audio_device *)dev;
+ in->dev = adev;
+
+ in->standby = true;
+ in->requested_rate = hal_config->sample_rate;
+ in->alsa_pcm_config = default_alsa_in_config;
+
+ if (hal_config->sample_rate != 0)
+ in->alsa_pcm_config.rate = hal_config->sample_rate;
+
+ //TODO(pmclean) is this correct, or do we need to map from ALSA format?
+ // hal_config->format is an audio_format_t
+ // logical
+ // hal_config->format = default_alsa_in_config.format;
+ //TODO(pmclean) use audio_format_from_pcm_format() (in hardware/audio_alsaops.h)
+ switch (default_alsa_in_config.format) {
+ case PCM_FORMAT_S32_LE:
+ hal_config->format = AUDIO_FORMAT_PCM_32_BIT;
+ break;
+
+ case PCM_FORMAT_S8:
+ hal_config->format = AUDIO_FORMAT_PCM_8_BIT;
+ break;
+
+ case PCM_FORMAT_S24_LE:
+ hal_config->format = AUDIO_FORMAT_PCM_8_24_BIT;
+ break;
+
+ case PCM_FORMAT_S24_3LE:
+ hal_config->format = AUDIO_FORMAT_PCM_8_24_BIT;
+ break;
+
+ default:
+ case PCM_FORMAT_S16_LE:
+ hal_config->format = AUDIO_FORMAT_PCM_16_BIT;
+ break;
+ }
+
+ *stream_in = &in->stream;
+
+ return 0;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
@@ -373,22 +1026,25 @@
static int adev_close(hw_device_t *device)
{
- struct audio_device *adev = (struct audio_device *)device;
+ ALOGV("usb:audio_hw::adev_close()");
+ struct audio_device *adev = (struct audio_device *)device;
free(device);
+
+ output_hardware_config_is_cached = false;
+
return 0;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
- struct audio_device *adev;
- int ret;
+ // ALOGV("usb:audio_hw::adev_open(%s)", name);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
- adev = calloc(1, sizeof(struct audio_device));
+ struct audio_device *adev = calloc(1, sizeof(struct audio_device));
if (!adev)
return -ENOMEM;