commit a41a34c6c4960eaf3ad05f9b2a626f5e766ee5bf
author qctecmdr <qctecmdr@localhost> Mon Jun 17 05:33:00 2019 -0700
committer Gerrit - the friendly Code Review server <code-review@localhost> Mon Jun 17 05:33:00 2019 -0700
tree e814d65c15c1f216ad5a169fbabbbc03337f62a7
parent 692619e67554fa0b27cd7a563c498506a90a742c
parent 8426d1f902738e83e9edd92a9064e82d8b506503

Merge "hal: Fix VoIP Rx no sound after openOutput if setVolume is not received"

hal/audio_hw.c

diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9691ce8..a5df951 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -93,6 +93,7 @@
 #define MMAP_PLAYBACK_VOLUME_MAX 0x2000
 #define PCM_PLAYBACK_VOLUME_MAX 0x2000
 #define DSD_VOLUME_MIN_DB (-110)
+#define INVALID_OUT_VOLUME -1
 
 #define RECORD_GAIN_MIN 0.0f
 #define RECORD_GAIN_MAX 1.0f
@@ -779,6 +780,11 @@
            (uc_id == USECASE_AUDIO_PLAYBACK_AFE_PROXY);
 }
 
+static inline bool is_valid_volume(float left, float right)
+{
+    return ((left >= 0.0f && right >= 0.0f) ? true : false);
+}
+
 static int enable_audio_route_for_voice_usecases(struct audio_device *adev,
                                                  struct audio_usecase *uc_info)
 {
@@ -4820,6 +4826,12 @@
     struct mixer_ctl *ctl;
     long set_values[4];
 
+    if (!is_valid_volume(left, right)) {
+        ALOGE("%s: Invalid stream volume for left=%f, right=%f",
+                   __func__, left, right);
+        return -EINVAL;
+    }
+
     ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
     if (!ctl) {
         ALOGE("%s: Could not get ctl for mixer cmd - %s",
@@ -6980,6 +6992,8 @@
                                         AUDIO_CHANNEL_OUT_MONO : AUDIO_CHANNEL_OUT_STEREO;
                 out->usecase = USECASE_AUDIO_PLAYBACK_VOIP;
                 out->format = AUDIO_FORMAT_PCM_16_BIT;
+                out->volume_l = INVALID_OUT_VOLUME;
+                out->volume_r = INVALID_OUT_VOLUME;
 
                 out->config = default_pcm_config_voip_copp;
                 out->config.period_size = VOIP_IO_BUF_SIZE(out->sample_rate, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;