Merge "Lock input and output sample rates" into nyc-dev

1 files changed, 24 insertions, 6 deletions

diff --git a/modules/usbaudio/audio_hal.c b/modules/usbaudio/audio_hal.c
index d899d6b9..4b3305c7 100644
--- a/modules/usbaudio/audio_hal.c
+++ b/modules/usbaudio/audio_hal.c
@@ -55,6 +55,9 @@ static const unsigned k_force_channels = 0;
 
 #define DEFAULT_INPUT_BUFFER_SIZE_MS 20
 
+/* Lock play & record samples rates at or above this threshold */
+#define RATELOCK_THRESHOLD 96000
+
 struct audio_device {
     struct audio_hw_device hw_device;
 
@@ -66,6 +69,10 @@ struct audio_device {
     /* input */
     alsa_device_profile in_profile;
 
+    /* lock input & output sample rates */
+    /*FIXME - How do we address multiple output streams? */
+    uint32_t device_sample_rate;
+
     bool mic_muted;
 
     bool standby;
@@ -517,8 +524,6 @@ static int adev_open_output_stream(struct audio_hw_device *dev,
 
     profile_read_device_info(out->profile);
 
-    pthread_mutex_unlock(&adev->lock);
-
     int ret = 0;
 
     /* Rate */
@@ -531,6 +536,9 @@ static int adev_open_output_stream(struct audio_hw_device *dev,
         ret = -EINVAL;
     }
 
+    out->dev->device_sample_rate = config->sample_rate;
+    pthread_mutex_unlock(&adev->lock);
+
     /* Format */
     if (config->format == AUDIO_FORMAT_DEFAULT) {
         proxy_config.format = profile_get_default_format(out->profile);
@@ -608,6 +616,10 @@ static void adev_close_output_stream(struct audio_hw_device *dev,
     out->conversion_buffer = NULL;
     out->conversion_buffer_size = 0;
 
+    pthread_mutex_lock(&out->dev->lock);
+    out->dev->device_sample_rate = 0;
+    pthread_mutex_unlock(&out->dev->lock);
+
     free(stream);
 }
 
@@ -759,7 +771,7 @@ static int in_set_gain(struct audio_stream_in *stream, float gain)
 /* must be called with hw device and output stream mutexes locked */
 static int start_input_stream(struct stream_in *in)
 {
-    ALOGV("ustart_input_stream(card:%d device:%d)", in->profile->card, in->profile->device);
+    ALOGV("start_input_stream(card:%d device:%d)", in->profile->card, in->profile->device);
 
     return proxy_open(&in->proxy);
 }
@@ -855,7 +867,7 @@ static int adev_open_input_stream(struct audio_hw_device *dev,
                                   const char *address /*__unused*/,
                                   audio_source_t source __unused)
 {
-    ALOGV("in adev_open_input_stream() rate:%" PRIu32 ", chanMask:0x%" PRIX32 ", fmt:%" PRIu8,
+    ALOGV("adev_open_input_stream() rate:%" PRIu32 ", chanMask:0x%" PRIX32 ", fmt:%" PRIu8,
           config->sample_rate, config->channel_mask, config->format);
 
     struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
@@ -897,17 +909,23 @@ static int adev_open_input_stream(struct audio_hw_device *dev,
     parse_card_device_params(address, &(in->profile->card), &(in->profile->device));
 
     profile_read_device_info(in->profile);
-    pthread_mutex_unlock(&in->dev->lock);
 
     /* Rate */
     if (config->sample_rate == 0) {
         proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
+    } else if (in->dev->device_sample_rate != 0 &&      /* we are playing, so lock the rate */
+               in->dev->device_sample_rate >= RATELOCK_THRESHOLD) {  /* but only for high
+                                                                        sample rates */
+        // Lock the rate to the output rate
+        ret = config->sample_rate != in->dev->device_sample_rate ? -EINVAL : 0;
+        proxy_config.rate = config->sample_rate = in->dev->device_sample_rate;
     } else if (profile_is_sample_rate_valid(in->profile, config->sample_rate)) {
-        proxy_config.rate = config->sample_rate;
+        in->dev->device_sample_rate = proxy_config.rate = config->sample_rate;
     } else {
         proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
         ret = -EINVAL;
     }
+    pthread_mutex_unlock(&in->dev->lock);
 
     /* Format */
     if (config->format == AUDIO_FORMAT_DEFAULT) {