Multi-format USB Audio suport
Change-Id: I2815e1dc38d39a3d88fb0affd52b1b9f50a73b92
diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c
index 3a6c5c9..7b65014 100644
--- a/modules/usbaudio/audio_hw.c
+++ b/modules/usbaudio/audio_hw.c
@@ -28,9 +28,11 @@
#include <cutils/str_parms.h>
#include <cutils/properties.h>
-#include <hardware/hardware.h>
-#include <system/audio.h>
#include <hardware/audio.h>
+#include <hardware/audio_alsaops.h>
+#include <hardware/hardware.h>
+
+#include <system/audio.h>
#include <tinyalsa/asoundlib.h>
@@ -141,69 +143,9 @@
* 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.
- * This conversion is safe to do in-place (in_buff == out_buff).
- * TODO(pmclean, hung) Move this to a utilities module.
- */
-static size_t convert_16_to_24_3(const 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;
- size_t src_smpl_index;
- const unsigned char* src_ptr = ((const 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 */
- /*TODO(pmclean) - we might want to consider dithering the lowest byte. */
- *dst_ptr-- = 0; /* zero-byte */
- }
-
- /* return number of *bytes* generated */
- return out_buff_size_in_bytes;
-}
-
-/*
* Convert a buffer of packed (3-byte) PCM24LE samples to PCM16LE samples.
* in_buff points to the buffer of PCM24LE samples
* num_in_samples size of input buffer in SAMPLES
@@ -215,9 +157,11 @@
* NOTE:
* We're just filling the low-order byte of the PCM24LE samples with 0.
* This conversion is safe to do in-place (in_buff == out_buff).
- * TODO(pmclean, hung) Move this to a utilities module.
+ * TODO Move this to a utilities module.
*/
-static size_t convert_24_3_to_16(const unsigned char * in_buff, size_t num_in_samples, short * out_buff) {
+static size_t convert_24_3_to_16(const unsigned char * in_buff, size_t num_in_samples,
+ short * out_buff)
+{
/*
* Move from front to back so that the conversion can be done in-place
* i.e. in_buff == out_buff
@@ -251,11 +195,12 @@
* This conversion is safe to do in-place (in_buff == out_buff)
* 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.
+ * TODO Move this to a utilities module.
*/
static size_t expand_channels_16(const short* in_buff, int in_buff_chans,
short* out_buff, int out_buff_chans,
- size_t num_in_samples) {
+ size_t num_in_samples)
+{
/*
* Move from back to front so that the conversion can be done in-place
* i.e. in_buff == out_buff
@@ -269,10 +214,10 @@
int num_zero_chans = out_buff_chans - in_buff_chans;
for (src_index = 0; src_index < num_in_samples; src_index += in_buff_chans) {
int dst_offset;
- for(dst_offset = 0; dst_offset < num_zero_chans; dst_offset++) {
+ for (dst_offset = 0; dst_offset < num_zero_chans; dst_offset++) {
*dst_ptr-- = 0;
}
- for(; dst_offset < out_buff_chans; dst_offset++) {
+ for (; dst_offset < out_buff_chans; dst_offset++) {
*dst_ptr-- = *src_ptr--;
}
}
@@ -296,11 +241,12 @@
* This conversion is safe to do in-place (in_buff == out_buff)
* 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.
+ * TODO Move this to a utilities module.
*/
static size_t contract_channels_16(short* in_buff, int in_buff_chans,
short* out_buff, int out_buff_chans,
- size_t num_in_samples) {
+ size_t num_in_samples)
+{
/*
* Move from front to back so that the conversion can be done in-place
* i.e. in_buff == out_buff
@@ -315,7 +261,7 @@
size_t src_index;
for (src_index = 0; src_index < num_in_samples; src_index += in_buff_chans) {
int dst_offset;
- for(dst_offset = 0; dst_offset < out_buff_chans; dst_offset++) {
+ for (dst_offset = 0; dst_offset < out_buff_chans; dst_offset++) {
*dst_ptr++ = *src_ptr++;
}
src_ptr += num_skip_samples;
@@ -328,19 +274,292 @@
/*
* ALSA Utilities
*/
+/*TODO This table and the function that uses it should be moved to a utilities module (probably) */
/*
- * gets the ALSA bit-format flag from a bits-per-sample value.
- * TODO(pmclean, hung) Move this to a utilities module.
+ * Maps bit-positions in a pcm_mask to the corresponding AUDIO_ format string.
*/
-static int bits_to_alsa_format(unsigned int bits_per_sample, int default_format)
+static const char * const format_string_map[] = {
+ "AUDIO_FORMAT_PCM_8_BIT", /* 00 - SNDRV_PCM_FORMAT_S8 */
+ "AUDIO_FORMAT_PCM_8_BIT", /* 01 - SNDRV_PCM_FORMAT_U8 */
+ "AUDIO_FORMAT_PCM_16_BIT", /* 02 - SNDRV_PCM_FORMAT_S16_LE */
+ NULL, /* 03 - SNDRV_PCM_FORMAT_S16_BE */
+ NULL, /* 04 - SNDRV_PCM_FORMAT_U16_LE */
+ NULL, /* 05 - SNDRV_PCM_FORMAT_U16_BE */
+ "AUDIO_FORMAT_PCM_24_BIT_PACKED", /* 06 - SNDRV_PCM_FORMAT_S24_LE */
+ NULL, /* 07 - SNDRV_PCM_FORMAT_S24_BE */
+ NULL, /* 08 - SNDRV_PCM_FORMAT_U24_LE */
+ NULL, /* 09 - SNDRV_PCM_FORMAT_U24_BE */
+ "AUDIO_FORMAT_PCM_32_BIT", /* 10 - SNDRV_PCM_FORMAT_S32_LE */
+ NULL, /* 11 - SNDRV_PCM_FORMAT_S32_BE */
+ NULL, /* 12 - SNDRV_PCM_FORMAT_U32_LE */
+ NULL, /* 13 - SNDRV_PCM_FORMAT_U32_BE */
+ "AUDIO_FORMAT_PCM_FLOAT", /* 14 - SNDRV_PCM_FORMAT_FLOAT_LE */
+ NULL, /* 15 - SNDRV_PCM_FORMAT_FLOAT_BE */
+ NULL, /* 16 - SNDRV_PCM_FORMAT_FLOAT64_LE */
+ NULL, /* 17 - SNDRV_PCM_FORMAT_FLOAT64_BE */
+ NULL, /* 18 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE */
+ NULL, /* 19 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE */
+ NULL, /* 20 - SNDRV_PCM_FORMAT_MU_LAW */
+ NULL, /* 21 - SNDRV_PCM_FORMAT_A_LAW */
+ NULL, /* 22 - SNDRV_PCM_FORMAT_IMA_ADPCM */
+ NULL, /* 23 - SNDRV_PCM_FORMAT_MPEG */
+ NULL, /* 24 - SNDRV_PCM_FORMAT_GSM */
+ NULL, NULL, NULL, NULL, NULL, NULL, /* 25 -> 30 (not assigned) */
+ NULL, /* 31 - SNDRV_PCM_FORMAT_SPECIAL */
+ "AUDIO_FORMAT_PCM_24_BIT_PACKED", /* 32 - SNDRV_PCM_FORMAT_S24_3LE */ /* ??? */
+ NULL, /* 33 - SNDRV_PCM_FORMAT_S24_3BE */
+ NULL, /* 34 - SNDRV_PCM_FORMAT_U24_3LE */
+ NULL, /* 35 - SNDRV_PCM_FORMAT_U24_3BE */
+ NULL, /* 36 - SNDRV_PCM_FORMAT_S20_3LE */
+ NULL, /* 37 - SNDRV_PCM_FORMAT_S20_3BE */
+ NULL, /* 38 - SNDRV_PCM_FORMAT_U20_3LE */
+ NULL, /* 39 - SNDRV_PCM_FORMAT_U20_3BE */
+ NULL, /* 40 - SNDRV_PCM_FORMAT_S18_3LE */
+ NULL, /* 41 - SNDRV_PCM_FORMAT_S18_3BE */
+ NULL, /* 42 - SNDRV_PCM_FORMAT_U18_3LE */
+ NULL, /* 43 - SNDRV_PCM_FORMAT_U18_3BE */
+ NULL, /* 44 - SNDRV_PCM_FORMAT_G723_24 */
+ NULL, /* 45 - SNDRV_PCM_FORMAT_G723_24_1B */
+ NULL, /* 46 - SNDRV_PCM_FORMAT_G723_40 */
+ NULL, /* 47 - SNDRV_PCM_FORMAT_G723_40_1B */
+ NULL, /* 48 - SNDRV_PCM_FORMAT_DSD_U8 */
+ NULL /* 49 - SNDRV_PCM_FORMAT_DSD_U16_LE */
+};
+
+/*
+ * Generate string containing a bar ("|") delimited list of AUDIO_ formats specified in
+ * the mask parameter.
+ *
+ */
+static char* get_format_str_for_mask(struct pcm_mask* mask)
{
- 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;
+ char buffer[256];
+ int buffer_size = sizeof(buffer) / sizeof(buffer[0]);
+ buffer[0] = '\0';
+
+ int num_slots = sizeof(mask->bits) / sizeof(mask->bits[0]);
+ int bits_per_slot = sizeof(mask->bits[0]) * 8;
+
+ const char* format_str = NULL;
+ int table_size = sizeof(format_string_map)/sizeof(format_string_map[0]);
+
+ int slot_index, bit_index, table_index;
+ table_index = 0;
+ int num_written = 0;
+ 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++) {
+ if ((mask->bits[slot_index] & bit_mask) != 0) {
+ format_str = table_index < table_size
+ ? format_string_map[table_index]
+ : NULL;
+ if (format_str != NULL) {
+ if (num_written != 0) {
+ num_written += snprintf(buffer + num_written,
+ buffer_size - num_written, "|");
+ }
+ num_written += snprintf(buffer + num_written, buffer_size - num_written,
+ "%s", format_str);
+ }
+ }
+ bit_mask <<= 1;
+ table_index++;
}
}
- return default_format;
+
+ return strdup(buffer);
+}
+
+/*
+ * Maps from bit position in pcm_mask to AUDIO_ format constants.
+ */
+static int const format_value_map[] = {
+ AUDIO_FORMAT_PCM_8_BIT, /* 00 - SNDRV_PCM_FORMAT_S8 */
+ AUDIO_FORMAT_PCM_8_BIT, /* 01 - SNDRV_PCM_FORMAT_U8 */
+ AUDIO_FORMAT_PCM_16_BIT, /* 02 - SNDRV_PCM_FORMAT_S16_LE */
+ AUDIO_FORMAT_INVALID, /* 03 - SNDRV_PCM_FORMAT_S16_BE */
+ AUDIO_FORMAT_INVALID, /* 04 - SNDRV_PCM_FORMAT_U16_LE */
+ AUDIO_FORMAT_INVALID, /* 05 - SNDRV_PCM_FORMAT_U16_BE */
+ AUDIO_FORMAT_PCM_24_BIT_PACKED, /* 06 - SNDRV_PCM_FORMAT_S24_LE */
+ AUDIO_FORMAT_INVALID, /* 07 - SNDRV_PCM_FORMAT_S24_BE */
+ AUDIO_FORMAT_INVALID, /* 08 - SNDRV_PCM_FORMAT_U24_LE */
+ AUDIO_FORMAT_INVALID, /* 09 - SNDRV_PCM_FORMAT_U24_BE */
+ AUDIO_FORMAT_PCM_32_BIT, /* 10 - SNDRV_PCM_FORMAT_S32_LE */
+ AUDIO_FORMAT_INVALID, /* 11 - SNDRV_PCM_FORMAT_S32_BE */
+ AUDIO_FORMAT_INVALID, /* 12 - SNDRV_PCM_FORMAT_U32_LE */
+ AUDIO_FORMAT_INVALID, /* 13 - SNDRV_PCM_FORMAT_U32_BE */
+ AUDIO_FORMAT_PCM_FLOAT, /* 14 - SNDRV_PCM_FORMAT_FLOAT_LE */
+ AUDIO_FORMAT_INVALID, /* 15 - SNDRV_PCM_FORMAT_FLOAT_BE */
+ AUDIO_FORMAT_INVALID, /* 16 - SNDRV_PCM_FORMAT_FLOAT64_LE */
+ AUDIO_FORMAT_INVALID, /* 17 - SNDRV_PCM_FORMAT_FLOAT64_BE */
+ AUDIO_FORMAT_INVALID, /* 18 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE */
+ AUDIO_FORMAT_INVALID, /* 19 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE */
+ AUDIO_FORMAT_INVALID, /* 20 - SNDRV_PCM_FORMAT_MU_LAW */
+ AUDIO_FORMAT_INVALID, /* 21 - SNDRV_PCM_FORMAT_A_LAW */
+ AUDIO_FORMAT_INVALID, /* 22 - SNDRV_PCM_FORMAT_IMA_ADPCM */
+ AUDIO_FORMAT_INVALID, /* 23 - SNDRV_PCM_FORMAT_MPEG */
+ AUDIO_FORMAT_INVALID, /* 24 - SNDRV_PCM_FORMAT_GSM */
+ AUDIO_FORMAT_INVALID, /* 25 -> 30 (not assigned) */
+ AUDIO_FORMAT_INVALID,
+ AUDIO_FORMAT_INVALID,
+ AUDIO_FORMAT_INVALID,
+ AUDIO_FORMAT_INVALID,
+ AUDIO_FORMAT_INVALID,
+ AUDIO_FORMAT_INVALID, /* 31 - SNDRV_PCM_FORMAT_SPECIAL */
+ AUDIO_FORMAT_PCM_24_BIT_PACKED, /* 32 - SNDRV_PCM_FORMAT_S24_3LE */ /* ??? */
+ AUDIO_FORMAT_INVALID, /* 33 - SNDRV_PCM_FORMAT_S24_3BE */
+ AUDIO_FORMAT_INVALID, /* 34 - SNDRV_PCM_FORMAT_U24_3LE */
+ AUDIO_FORMAT_INVALID, /* 35 - SNDRV_PCM_FORMAT_U24_3BE */
+ AUDIO_FORMAT_INVALID, /* 36 - SNDRV_PCM_FORMAT_S20_3LE */
+ AUDIO_FORMAT_INVALID, /* 37 - SNDRV_PCM_FORMAT_S20_3BE */
+ AUDIO_FORMAT_INVALID, /* 38 - SNDRV_PCM_FORMAT_U20_3LE */
+ AUDIO_FORMAT_INVALID, /* 39 - SNDRV_PCM_FORMAT_U20_3BE */
+ AUDIO_FORMAT_INVALID, /* 40 - SNDRV_PCM_FORMAT_S18_3LE */
+ AUDIO_FORMAT_INVALID, /* 41 - SNDRV_PCM_FORMAT_S18_3BE */
+ AUDIO_FORMAT_INVALID, /* 42 - SNDRV_PCM_FORMAT_U18_3LE */
+ AUDIO_FORMAT_INVALID, /* 43 - SNDRV_PCM_FORMAT_U18_3BE */
+ AUDIO_FORMAT_INVALID, /* 44 - SNDRV_PCM_FORMAT_G723_24 */
+ AUDIO_FORMAT_INVALID, /* 45 - SNDRV_PCM_FORMAT_G723_24_1B */
+ AUDIO_FORMAT_INVALID, /* 46 - SNDRV_PCM_FORMAT_G723_40 */
+ AUDIO_FORMAT_INVALID, /* 47 - SNDRV_PCM_FORMAT_G723_40_1B */
+ AUDIO_FORMAT_INVALID, /* 48 - SNDRV_PCM_FORMAT_DSD_U8 */
+ AUDIO_FORMAT_INVALID /* 49 - SNDRV_PCM_FORMAT_DSD_U16_LE */
+};
+
+static int get_format_for_mask(struct pcm_mask* mask)
+{
+ int num_slots = sizeof(mask->bits)/ sizeof(mask->bits[0]);
+ int bits_per_slot = sizeof(mask->bits[0]) * 8;
+
+ int table_size = sizeof(format_value_map) / sizeof(format_value_map[0]);
+
+ int slot_index, bit_index, table_index;
+ table_index = 0;
+ int num_written = 0;
+ 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++) {
+ if ((mask->bits[slot_index] & bit_mask) != 0) {
+ /* just return the first one */
+ return table_index < table_size
+ ? format_value_map[table_index]
+ : AUDIO_FORMAT_INVALID;
+ }
+ bit_mask <<= 1;
+ table_index++;
+ }
+ }
+
+ return AUDIO_FORMAT_INVALID;
+}
+
+/*
+ * Maps from bit position in pcm_mask to AUDIO_ format constants.
+ */
+static int const pcm_format_value_map[] = {
+ PCM_FORMAT_S8, /* 00 - SNDRV_PCM_FORMAT_S8 */
+ 0, /* 01 - SNDRV_PCM_FORMAT_U8 */
+ PCM_FORMAT_S16_LE, /* 02 - SNDRV_PCM_FORMAT_S16_LE */
+ 0, /* 03 - SNDRV_PCM_FORMAT_S16_BE */
+ 0, /* 04 - SNDRV_PCM_FORMAT_U16_LE */
+ 0, /* 05 - SNDRV_PCM_FORMAT_U16_BE */
+ PCM_FORMAT_S24_3LE, /* 06 - SNDRV_PCM_FORMAT_S24_LE */
+ 0, /* 07 - SNDRV_PCM_FORMAT_S24_BE */
+ 0, /* 08 - SNDRV_PCM_FORMAT_U24_LE */
+ 0, /* 09 - SNDRV_PCM_FORMAT_U24_BE */
+ PCM_FORMAT_S32_LE, /* 10 - SNDRV_PCM_FORMAT_S32_LE */
+ 0, /* 11 - SNDRV_PCM_FORMAT_S32_BE */
+ 0, /* 12 - SNDRV_PCM_FORMAT_U32_LE */
+ 0, /* 13 - SNDRV_PCM_FORMAT_U32_BE */
+ 0, /* 14 - SNDRV_PCM_FORMAT_FLOAT_LE */
+ 0, /* 15 - SNDRV_PCM_FORMAT_FLOAT_BE */
+ 0, /* 16 - SNDRV_PCM_FORMAT_FLOAT64_LE */
+ 0, /* 17 - SNDRV_PCM_FORMAT_FLOAT64_BE */
+ 0, /* 18 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE */
+ 0, /* 19 - SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE */
+ 0, /* 20 - SNDRV_PCM_FORMAT_MU_LAW */
+ 0, /* 21 - SNDRV_PCM_FORMAT_A_LAW */
+ 0, /* 22 - SNDRV_PCM_FORMAT_IMA_ADPCM */
+ 0, /* 23 - SNDRV_PCM_FORMAT_MPEG */
+ 0, /* 24 - SNDRV_PCM_FORMAT_GSM */
+ 0, /* 25 -> 30 (not assigned) */
+ 0,
+ 0,
+ 0,
+ 0,
+ 0,
+ 0, /* 31 - SNDRV_PCM_FORMAT_SPECIAL */
+ PCM_FORMAT_S24_3LE, /* 32 - SNDRV_PCM_FORMAT_S24_3LE */ /* ??? */
+ 0, /* 33 - SNDRV_PCM_FORMAT_S24_3BE */
+ 0, /* 34 - SNDRV_PCM_FORMAT_U24_3LE */
+ 0, /* 35 - SNDRV_PCM_FORMAT_U24_3BE */
+ 0, /* 36 - SNDRV_PCM_FORMAT_S20_3LE */
+ 0, /* 37 - SNDRV_PCM_FORMAT_S20_3BE */
+ 0, /* 38 - SNDRV_PCM_FORMAT_U20_3LE */
+ 0, /* 39 - SNDRV_PCM_FORMAT_U20_3BE */
+ 0, /* 40 - SNDRV_PCM_FORMAT_S18_3LE */
+ 0, /* 41 - SNDRV_PCM_FORMAT_S18_3BE */
+ 0, /* 42 - SNDRV_PCM_FORMAT_U18_3LE */
+ 0, /* 43 - SNDRV_PCM_FORMAT_U18_3BE */
+ 0, /* 44 - SNDRV_PCM_FORMAT_G723_24 */
+ 0, /* 45 - SNDRV_PCM_FORMAT_G723_24_1B */
+ 0, /* 46 - SNDRV_PCM_FORMAT_G723_40 */
+ 0, /* 47 - SNDRV_PCM_FORMAT_G723_40_1B */
+ 0, /* 48 - SNDRV_PCM_FORMAT_DSD_U8 */
+ 0 /* 49 - SNDRV_PCM_FORMAT_DSD_U16_LE */
+};
+
+static int get_pcm_format_for_mask(struct pcm_mask* mask) {
+ int num_slots = sizeof(mask->bits)/ sizeof(mask->bits[0]);
+ int bits_per_slot = sizeof(mask->bits[0]) * 8;
+
+ int table_size = sizeof(pcm_format_value_map) / sizeof(pcm_format_value_map[0]);
+
+ int slot_index, bit_index, table_index;
+ table_index = 0;
+ int num_written = 0;
+ 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++) {
+ if ((mask->bits[slot_index] & bit_mask) != 0) {
+ /* just return the first one */
+ return table_index < table_size
+ ? pcm_format_value_map[table_index]
+ : AUDIO_FORMAT_INVALID;
+ }
+ bit_mask <<= 1;
+ table_index++;
+ }
+ }
+
+ return 0; // is this right?
+}
+
+static void log_pcm_mask(const char* mask_name, struct pcm_mask* mask) {
+ char buff[512];
+ char bit_buff[32];
+ int buffSize = sizeof(buff)/sizeof(buff[0]);
+
+ buff[0] = '\0';
+
+ int num_slots = sizeof(mask->bits) / sizeof(mask->bits[0]);
+ int bits_per_slot = sizeof(mask->bits[0]) * 8;
+
+ int 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("usb:audio_hw - %s mask:%s", mask_name, buff);
}
static void log_pcm_params(struct pcm_params * alsa_hw_params) {
@@ -350,6 +569,8 @@
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));
@@ -389,7 +610,7 @@
return (size + 15) & 0xFFFFFFF0;
}
-/*TODO(pmclean) - Evaluate if this value should/can be retrieved from a device-specific property */
+/*TODO - Evaluate if this value should/can be retrieved from a device-specific property */
#define MIN_BUFF_TIME 5 /* milliseconds */
/*
@@ -433,9 +654,7 @@
}
config->period_count = pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIODS);
- unsigned 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);
-
+ config->format = get_pcm_format_for_mask(pcm_params_get_mask(alsa_hw_params, PCM_PARAM_FORMAT));
return 0;
}
@@ -466,21 +685,19 @@
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
+ /* TODO 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;
+ return audio_format_from_pcm_format(cached_output_hardware_config.format);
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
+ cached_output_hardware_config.format = pcm_format_from_audio_format(format);
return 0;
}
@@ -548,8 +765,8 @@
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)
+/*TODO 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)
{
@@ -591,6 +808,7 @@
// supported channel counts
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
// Similarly for output channels count
+ /* TODO - This is wrong, we need format strings, not numbers (another CL) */
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, "%u", min);
@@ -602,15 +820,10 @@
// 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, "%u", min);
- if (min != max) {
- snprintf(buffer + num_written, buffer_size - num_written, "|%u", max);
- }
- str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS, buffer);
+ char * format_params =
+ get_format_str_for_mask(pcm_params_get_mask(alsa_hw_params, PCM_PARAM_FORMAT));
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS, format_params);
+ free(format_params);
} // AUDIO_PARAMETER_STREAM_SUP_FORMATS
result_str = str_parms_to_str(result);
@@ -619,6 +832,8 @@
str_parms_destroy(query);
str_parms_destroy(result);
+ ALOGV("usb:audio_hw::out out_get_parameters() = %s", result_str);
+
return result_str;
}
@@ -626,10 +841,10 @@
{
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)?
+ /*TODO 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);
+ * cached_output_hardware_config.period_count * 1000)
+ / out_get_sample_rate(&stream->common);
return latency;
}
@@ -648,6 +863,7 @@
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;
}
@@ -682,8 +898,8 @@
// * 3/2 for 16bit -> 24 bit conversion
size_t required_conversion_buffer_size = (bytes * 3 * 2) / 2;
if (required_conversion_buffer_size > out->conversion_buffer_size) {
- //TODO(pmclean) - remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
- // (and do these conversions themselves)
+ /* TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
+ (and do these conversions themselves) */
out->conversion_buffer_size = required_conversion_buffer_size;
out->conversion_buffer = realloc(out->conversion_buffer, out->conversion_buffer_size);
}
@@ -704,28 +920,6 @@
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 / sizeof(short),
- 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);
}
@@ -808,20 +1002,14 @@
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->format = audio_format_from_pcm_format(cached_output_hardware_config.format);
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.
+ /*TODO remove this when the above restriction is removed. */
config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
}
} else {
@@ -852,8 +1040,7 @@
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
+ // Close the pcm device
out_standby(&stream->common);
free(out->conversion_buffer);
@@ -925,7 +1112,6 @@
ALOGV("usb: in_get_buffer_size() = %zu",
cached_input_hardware_config.period_size * audio_stream_frame_size(stream));
return cached_input_hardware_config.period_size * audio_stream_frame_size(stream);
-
}
static uint32_t in_get_channels(const struct audio_stream *stream)
@@ -936,8 +1122,7 @@
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;
+ return audio_format_from_pcm_format(cached_input_hardware_config.format);
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
@@ -1011,8 +1196,8 @@
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)
+/*TODO 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);
@@ -1054,6 +1239,7 @@
// supported channel counts
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
// Similarly for output channels count
+ // TODO This is wrong, we need format strings, not numbers (another CL)
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, "%u", min);
@@ -1065,7 +1251,8 @@
// supported sample formats
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
- //TODO(pmclean): this is wrong.
+ /*TODO This is wrong. It needs to return AUDIO_ format constants (as strings)
+ as in audio_policy.conf */
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, "%u", min);
@@ -1122,7 +1309,7 @@
return 0;
}
-//TODO(pmclean) mutex stuff here (see out_write)
+/* TODO mutex stuff here (see out_write) */
static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
{
size_t num_read_buff_bytes = 0;
@@ -1131,8 +1318,6 @@
struct stream_in * in = (struct stream_in *) stream;
- ALOGV("usb: in_read(%d)", bytes);
-
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
@@ -1160,8 +1345,8 @@
// Setup/Realloc the conversion buffer (if necessary).
if (num_read_buff_bytes != bytes) {
if (num_read_buff_bytes > in->conversion_buffer_size) {
- //TODO(pmclean) - remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
- // (and do these conversions themselves)
+ /*TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
+ (and do these conversions themselves) */
in->conversion_buffer_size = num_read_buff_bytes;
in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
}
@@ -1253,11 +1438,11 @@
if (input_hardware_config_is_cached) {
config->sample_rate = cached_input_hardware_config.rate;
- config->format = alsa_to_fw_format_id(cached_input_hardware_config.format);
+ config->format = audio_format_from_pcm_format(cached_input_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.
+ /* TODO Remove this when the above restriction is removed. */
config->format = AUDIO_FORMAT_PCM_16_BIT;
}
@@ -1266,7 +1451,7 @@
if (config->channel_mask != AUDIO_CHANNEL_IN_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.
+ /* TODO Remove this when the above restriction is removed. */
config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
}
} else {
@@ -1291,8 +1476,7 @@
{
struct stream_in *in = (struct stream_in *)stream;
- //TODO(pmclean) why are we doing this when stream get's freed at the end
- // because it closes the pcm device
+ // Close the pcm device
in_standby(&stream->common);
free(in->conversion_buffer);