commit 685fadac731dd5e56c9bb91386cdcda862f560cc
author Eric Laurent <elaurent@google.com> Tue Jun 17 18:26:43 2014 +0000
committer Android (Google) Code Review <android-gerrit@google.com> Tue Jun 17 05:00:50 2014 +0000
tree c7d76c453198187f3d3c36675ccc51b6e7fba190
parent ace6d32dd7ce44cf66389ea5ad9b975a843f6064
parent 7661a484021f6e7c3b219bd21659118eef94e45b

Merge "usb audio: fix usb capture configuration"

modules/usbaudio/audio_hw.c

diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c
index 7b2a4f1..4b50d58 100644
--- a/modules/usbaudio/audio_hw.c
+++ b/modules/usbaudio/audio_hw.c
@@ -1128,7 +1128,7 @@
         num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels;
     }
 
-    if (cached_output_hardware_config.format == PCM_FORMAT_S24_3LE) {
+    if (cached_input_hardware_config.format == PCM_FORMAT_S24_3LE) {
         num_read_buff_bytes = (3 * num_read_buff_bytes) / 2;
     }
 
@@ -1148,7 +1148,7 @@
          * 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 (cached_input_hardware_config.format == PCM_FORMAT_S24_3LE) {
             if (num_device_channels != num_req_channels) {
                 out_buff = read_buff;
             }
@@ -1197,6 +1197,8 @@
           config->sample_rate, config->channel_mask, config->format);
 
     struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
+    int ret = 0;
+
     if (in == NULL)
         return -ENOMEM;
 
@@ -1220,10 +1222,13 @@
 
     in->dev = (struct audio_device *)dev;
 
-    if (output_hardware_config_is_cached) {
-        config->sample_rate = cached_output_hardware_config.rate;
+    if (config->channel_mask != AUDIO_CHANNEL_IN_STEREO)
+        ret = -EINVAL;
 
-        config->format = alsa_to_fw_format_id(cached_output_hardware_config.format);
+    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);
         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).
@@ -1231,20 +1236,20 @@
             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) {
+        config->channel_mask = audio_channel_in_mask_from_count(
+                cached_input_hardware_config.channels);
+        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.
-            config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+            config->channel_mask = AUDIO_CHANNEL_IN_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);
+        config->format = in_get_format(&in->stream.common);
+        config->channel_mask = in_get_channels(&in->stream.common);
+        config->sample_rate = in_get_sample_rate(&in->stream.common);
     }
 
     in->standby = true;
@@ -1254,7 +1259,7 @@
 
     *stream_in = &in->stream;
 
-    return 0;
+    return ret;
 }
 
 static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream)