USB HAL Capture Implementation
Change-Id: Ic04f7e1b33b423e5169667700145b9cbdd9704a6
diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c
index afe56b2..4cb0071 100644
--- a/modules/usbaudio/audio_hw.c
+++ b/modules/usbaudio/audio_hw.c
@@ -15,7 +15,7 @@
*/
#define LOG_TAG "usb_audio_hw"
-/*#define LOG_NDEBUG 0*/
+/* #define LOG_NDEBUG 0 */
#include <errno.h>
#include <pthread.h>
@@ -99,7 +99,7 @@
/*
* Output Configuration Cache
- * FIXME(pmclean) This is not rentrant. Should probably be moved into the stream structure
+ * FIXME(pmclean) This is not reentrant. Should probably be moved into the stream structure
* but that will involve changes in The Framework.
*/
static struct pcm_config cached_output_hardware_config;
@@ -112,22 +112,31 @@
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;
+
+ // We may need to read more data from the device in order to data reduce to 16bit, 4chan */
+ 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 */
};
/*
+ * Input Configuration Cache
+ * FIXME(pmclean) This is not reentrant. Should probably be moved into the stream structure
+ * but that will involve changes in The Framework.
+ */
+static struct pcm_config cached_input_hardware_config;
+static bool input_hardware_config_is_cached = false;
+
+/*
* Utility
*/
/*
@@ -162,15 +171,16 @@
* 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.
+ * 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.
+ * 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(unsigned short * in_buff,
- size_t num_in_samples,
- unsigned char * out_buff) {
+static size_t convert_16_to_24_3(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
@@ -179,11 +189,12 @@
/* 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;
+ size_t 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 */
+ /*TODO(pmclean) - we might want to consider dithering the lowest byte. */
*dst_ptr-- = 0; /* zero-byte */
}
@@ -192,36 +203,125 @@
}
/*
- * Convert a buffer of 2-channel PCM16 samples to 4-channel PCM16 channels
- * in_buff points to the buffer of PCM16 samples
+ * 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
+ * out_buff points to the buffer to receive converted PCM16LE 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_24_3_to_16(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
+ */
+ /* we need 2 bytes in the output for every 3 bytes in the input */
+ unsigned char* dst_ptr = (unsigned char*)out_buff;
+ unsigned char* src_ptr = in_buff;
+ size_t src_smpl_index;
+ for (src_smpl_index = 0; src_smpl_index < num_in_samples; src_smpl_index++) {
+ src_ptr++; /* lowest-(skip)-byte */
+ *dst_ptr++ = *src_ptr++; /* low-byte */
+ *dst_ptr++ = *src_ptr++; /* high-byte */
+ }
+
+ /* return number of *bytes* generated: */
+ return num_in_samples * 2;
+}
+
+/*
+ * Convert a buffer of N-channel, interleaved PCM16 samples to M-channel PCM16 channels
+ * (where N < M).
+ * in_buff points to the buffer of PCM16 samples
+ * in_buff_channels Specifies the number of channels in the input buffer.
* 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.
+ * out_buff_channels Specifies the number of channels in the output buffer.
+ * num_in_samples size of input buffer in SAMPLES
+ * returns
+ * the number of BYTES of output data.
+ * NOTE
+ * channels > N are filled with silence.
+ * 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.
*/
-static size_t convert_2chan16_to_4chan16(unsigned short* in_buff,
- size_t num_in_samples,
- unsigned short* out_buff) {
+static size_t expand_channels_16(short* in_buff, int in_buff_chans,
+ short* out_buff, int out_buff_chans,
+ 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
+ * NOTE: num_in_samples * out_buff_channels must be an even multiple of in_buff_chans
*/
- int out_buff_size = num_in_samples * 2;
- unsigned short* dst_ptr = out_buff + out_buff_size - 1;
+ int num_out_samples = (num_in_samples * out_buff_chans)/in_buff_chans;
+
+ short* dst_ptr = out_buff + num_out_samples - 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 */
+ short* src_ptr = in_buff + num_in_samples - 1;
+ 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++) {
+ *dst_ptr-- = 0;
+ }
+ for(; dst_offset < out_buff_chans; dst_offset++) {
+ *dst_ptr-- = *src_ptr--;
+ }
}
/* return number of *bytes* generated */
- return out_buff_size * 2;
+ return num_out_samples * sizeof(short);
+}
+
+/*
+ * Convert a buffer of N-channel, interleaved PCM16 samples to M-channel PCM16 channels
+ * (where N > M).
+ * in_buff points to the buffer of PCM16 samples
+ * in_buff_channels Specifies the number of channels in the input buffer.
+ * out_buff points to the buffer to receive converted PCM16 samples.
+ * out_buff_channels Specifies the number of channels in the output buffer.
+ * num_in_samples size of input buffer in SAMPLES
+ * returns
+ * the number of BYTES of output data.
+ * NOTE
+ * channels > N are thrown away.
+ * 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.
+ */
+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) {
+ /*
+ * Move from front to back so that the conversion can be done in-place
+ * i.e. in_buff == out_buff
+ * NOTE: num_in_samples * out_buff_channels must be an even multiple of in_buff_chans
+ */
+ int num_out_samples = (num_in_samples * out_buff_chans)/in_buff_chans;
+
+ int num_skip_samples = in_buff_chans - out_buff_chans;
+
+ short* dst_ptr = out_buff;
+ short* src_ptr = in_buff;
+ int 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++) {
+ *dst_ptr++ = *src_ptr++;
+ }
+ src_ptr += num_skip_samples;
+ }
+
+ /* return number of *bytes* generated */
+ return num_out_samples * sizeof(short);
}
/*
@@ -236,8 +336,8 @@
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 format;
+ }
}
return default_format;
}
@@ -247,7 +347,7 @@
*/
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);
+ ALOGV("usb:audio_hw - read_alsa_device_config(c:%d d:%d t:0x%X)",card, device, io_type);
if (card < 0 || device < 0) {
return -EINVAL;
@@ -384,7 +484,7 @@
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));
+ &cached_output_hardware_config);
output_hardware_config_is_cached = (ret_value == 0);
}
@@ -397,10 +497,16 @@
//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) {
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+ ALOGV("usb:audio_hw::out out_get_parameters() keys:%s", keys);
+
struct stream_out *out = (struct stream_out *) stream;
struct audio_device *adev = out->dev;
+ if (adev->out_card < 0 || adev->out_device < 0)
+ return strdup("");
+
unsigned min, max;
struct str_parms *query = str_parms_create_str(keys);
@@ -422,8 +528,7 @@
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);
+ snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
}
str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
buffer);
@@ -465,17 +570,16 @@
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
- struct stream_out *out = (struct stream_out *)stream;
+ 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);
+ 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,
- float right)
+static int out_set_volume(struct audio_stream_out *stream, float left, float right)
{
return -ENOSYS;
}
@@ -486,8 +590,8 @@
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);
+ 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) {
@@ -500,17 +604,6 @@
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;
}
@@ -529,6 +622,18 @@
out->standby = false;
}
+ // Setup conversion buffer
+ // compute maximum potential buffer size.
+ // * 2 for stereo -> quad conversion
+ // * 3/2 for 16bit -> 24 bit conversion
+ int 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)
+ out->conversion_buffer_size = required_conversion_buffer_size;
+ out->conversion_buffer = realloc(out->conversion_buffer, out->conversion_buffer_size);
+ }
+
void * write_buff = buffer;
int num_write_buff_bytes = bytes;
@@ -537,10 +642,11 @@
*/
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) {
+ if (num_device_channels != num_req_channels) {
num_write_buff_bytes =
- convert_2chan16_to_4chan16(write_buff, num_write_buff_bytes / 2,
- out->conversion_buffer);
+ expand_channels_16(write_buff, num_req_channels,
+ out->conversion_buffer, num_device_channels,
+ num_write_buff_bytes / sizeof(short));
write_buff = out->conversion_buffer;
}
@@ -554,8 +660,9 @@
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);
+ 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;
@@ -585,8 +692,7 @@
return bytes;
}
-static int out_get_render_position(const struct audio_stream_out *stream,
- uint32_t *dsp_frames)
+static int out_get_render_position(const struct audio_stream_out *stream, uint32_t *dsp_frames)
{
return -EINVAL;
}
@@ -601,8 +707,7 @@
return 0;
}
-static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
- int64_t *timestamp)
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream, int64_t *timestamp)
{
return -EINVAL;
}
@@ -614,7 +719,7 @@
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",
+ ALOGV("usb:audio_hw::out adev_open_output_stream() handle:0x%X, device:0x%X, flags:0x%X",
handle, devices, flags);
struct audio_device *adev = (struct audio_device *)dev;
@@ -672,9 +777,6 @@
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;
@@ -712,8 +814,7 @@
return 0;
}
-static char * adev_get_parameters(const struct audio_hw_device *dev,
- const char *keys)
+static char * adev_get_parameters(const struct audio_hw_device *dev, const char *keys)
{
return strdup("");
}
@@ -757,8 +858,7 @@
/* 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;
+ return cached_input_hardware_config.rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
@@ -768,19 +868,16 @@
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;
+ ALOGV("usb: in_get_buffer_size() = %d",
+ 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)
{
- 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;
+ // just report stereo for now
+ return AUDIO_CHANNEL_IN_STEREO;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
@@ -796,7 +893,20 @@
static int in_standby(struct audio_stream *stream)
{
- ALOGV("-pcm-audio_hw::in in_standby() [Not Implemented]");
+ struct stream_in *in = (struct stream_in *) stream;
+
+ pthread_mutex_lock(&in->dev->lock);
+ pthread_mutex_lock(&in->lock);
+
+ if (!in->standby) {
+ pcm_close(in->pcm);
+ in->pcm = NULL;
+ in->standby = true;
+ }
+
+ pthread_mutex_unlock(&in->lock);
+ pthread_mutex_unlock(&in->dev->lock);
+
return 0;
}
@@ -807,7 +917,7 @@
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
- ALOGV("Vaudio_hw::in in_set_parameters() keys:%s", kvpairs);
+ ALOGV("usb: audio_hw::in in_set_parameters() keys:%s", kvpairs);
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
@@ -820,19 +930,25 @@
parms = str_parms_create_str(kvpairs);
pthread_mutex_lock(&adev->lock);
+ bool recache_device_params = false;
+
// Card/Device
param_val = str_parms_get_str(parms, "card", value, sizeof(value));
if (param_val >= 0) {
adev->in_card = atoi(value);
+ recache_device_params = true;
}
param_val = str_parms_get_str(parms, "device", value, sizeof(value));
if (param_val >= 0) {
adev->in_device = atoi(value);
+ recache_device_params = true;
}
- 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));
+ if (recache_device_params && adev->in_card >= 0 && adev->in_device >= 0) {
+ ret_value = read_alsa_device_config(adev->in_card, adev->in_device,
+ PCM_IN, &(cached_input_hardware_config));
+ input_hardware_config_is_cached = (ret_value == 0);
}
pthread_mutex_unlock(&adev->lock);
@@ -844,72 +960,74 @@
//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);
+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 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("");
+ if (adev->in_card < 0 || adev->in_device < 0)
+ return strdup("");
- struct str_parms *query = str_parms_create_str(keys);
- struct str_parms *result = str_parms_create();
+ struct pcm_params * alsa_hw_params = pcm_params_get(adev->in_card, adev->in_device, PCM_IN);
+ if (alsa_hw_params == NULL)
+ return strdup("");
- int num_written = 0;
- char buffer[256];
- int buffer_size = sizeof(buffer)/sizeof(buffer[0]);
- char* result_str = NULL;
+ struct str_parms *query = str_parms_create_str(keys);
+ struct str_parms *result = str_parms_create();
- unsigned min, max;
+ int num_written = 0;
+ char buffer[256];
+ int buffer_size = sizeof(buffer) / sizeof(buffer[0]);
+ char* result_str = NULL;
- // 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
+ unsigned min, max;
- // 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
+ // 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 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
+ // 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
- result_str = str_parms_to_str(result);
+ // 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
- // done with these...
- str_parms_destroy(query);
- str_parms_destroy(result);
+ result_str = str_parms_to_str(result);
- return result_str;
+ // 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)
@@ -922,32 +1040,123 @@
return 0;
}
-static int in_set_gain(struct audio_stream_in *stream, float gain) {
+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;
+/* must be called with hw device and output stream mutexes locked */
+static int start_input_stream(struct stream_in *in) {
+ struct audio_device *adev = in->dev;
+ int return_val = 0;
- int err = pcm_read(in->pcm, buffer, bytes);
+ ALOGV("usb:audio_hw::start_input_stream(card:%d device:%d)",
+ adev->in_card, adev->in_device);
- return err == 0 ? bytes : 0;
+ in->pcm = pcm_open(adev->in_card, adev->in_device, PCM_IN, &cached_input_hardware_config);
+ if (in->pcm == NULL) {
+ ALOGE("usb:audio_hw pcm_open() in->pcm == NULL");
+ return -ENOMEM;
+ }
+
+ if (in->pcm && !pcm_is_ready(in->pcm)) {
+ ALOGE("usb:audio_hw audio_hw pcm_open() failed: %s", pcm_get_error(in->pcm));
+ pcm_close(in->pcm);
+ return -ENOMEM;
+ }
+
+ return 0;
}
-static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) {
+//TODO(pmclean) mutex stuff here (see out_write)
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
+{
+ int num_read_buff_bytes = 0;
+ void * read_buff = buffer;
+ void * out_buff = buffer;
+
+ struct stream_in * in = (struct stream_in *) stream;
+
+ pthread_mutex_lock(&in->dev->lock);
+ pthread_mutex_lock(&in->lock);
+
+ if (in->standby) {
+ if (start_input_stream(in) != 0) {
+ goto err;
+ }
+ in->standby = false;
+ }
+
+ // OK, we need to figure out how much data to read to be able to output the requested
+ // number of bytes in the HAL format (16-bit, stereo).
+ num_read_buff_bytes = bytes;
+ int num_device_channels = cached_input_hardware_config.channels;
+ int num_req_channels = 2; /* always, for now */
+
+ if (num_device_channels != num_req_channels) {
+ num_read_buff_bytes *= num_device_channels/num_req_channels;
+ }
+
+ if (cached_output_hardware_config.format == PCM_FORMAT_S24_3LE) {
+ num_read_buff_bytes = (3 * num_read_buff_bytes) / 2;
+ }
+
+ // 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)
+ in->conversion_buffer_size = num_read_buff_bytes;
+ in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
+ }
+ read_buff = in->conversion_buffer;
+ }
+
+ if (pcm_read(in->pcm, read_buff, num_read_buff_bytes) == 0) {
+ /*
+ * Do any conversions necessary to send the data in the format specified to/by the HAL
+ * (but different from the ALSA format), such as 24bit ->16bit, or 4chan -> 2chan.
+ */
+ if (cached_output_hardware_config.format == PCM_FORMAT_S24_3LE) {
+ if (num_device_channels != num_req_channels) {
+ out_buff = read_buff;
+ }
+
+ /* Bit Format Conversion */
+ num_read_buff_bytes =
+ convert_24_3_to_16(read_buff, num_read_buff_bytes / 3, out_buff);
+ }
+
+ if (num_device_channels != num_req_channels) {
+ out_buff = buffer;
+ /* Num Channels conversion */
+ num_read_buff_bytes =
+ contract_channels_16(read_buff, num_device_channels,
+ out_buff, num_req_channels,
+ num_read_buff_bytes / sizeof(short));
+ }
+ }
+
+err:
+ pthread_mutex_unlock(&in->lock);
+ pthread_mutex_unlock(&in->dev->lock);
+
+ return num_read_buff_bytes;
+}
+
+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 *hal_config,
+ struct audio_config *config,
struct audio_stream_in **stream_in)
{
- 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);
+ ALOGV("usb: in adev_open_input_stream() rate:%d, chanMask:0x%X, fmt:%d",
+ config->sample_rate, config->channel_mask, config->format);
struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
if (in == NULL)
@@ -971,52 +1180,56 @@
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->dev = (struct audio_device *)dev;
+
+ 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_input_hardware_config = default_alsa_in_config;
+
+ config->format = out_get_format(&in->stream.common);
+ config->channel_mask = out_get_channels(&in->stream.common);
+ config->sample_rate = out_get_sample_rate(&in->stream.common);
+ }
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;
- }
+ in->conversion_buffer = NULL;
+ in->conversion_buffer_size = 0;
*stream_in = &in->stream;
return 0;
}
-static void adev_close_input_stream(struct audio_hw_device *dev,
- struct audio_stream_in *stream)
+static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream)
{
+ 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
+ in_standby(&stream->common);
+
+ free(in->conversion_buffer);
+
+ free(stream);
}
static int adev_dump(const audio_hw_device_t *device, int fd)
@@ -1026,21 +1239,17 @@
static int adev_close(hw_device_t *device)
{
- ALOGV("usb:audio_hw::adev_close()");
-
struct audio_device *adev = (struct audio_device *)device;
free(device);
output_hardware_config_is_cached = false;
+ input_hardware_config_is_cached = false;
return 0;
}
-static int adev_open(const hw_module_t* module, const char* name,
- hw_device_t** device)
+static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device)
{
- // ALOGV("usb:audio_hw::adev_open(%s)", name);
-
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;