hal: Fix Glitch issue during concurrent playback
- VOIP Playback and VOIP Record streams are opened with 8k for whatsapp
call. Then resume of compress offload playback happens.
- In this scenario, Compress offload backend would be configured to 48k
even though clip is of different sample rate.
- This results in backend reconfiguration for next Low Latency playback
which would be heard as glitch.
- Fix this by avoiding default sample rate selection based on usecase
sample rate. Instead check the condition for backend sample rate
configuration.
Change-Id: If9b1825628dee4717740b442431fb17487e503c1
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index ef23b15..2d7bc91 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -5983,7 +5983,11 @@
bit_width = out->bit_width;
if (sample_rate < out->sample_rate)
sample_rate = out->sample_rate;
- if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
+ /*
+ * TODO: Add Support for Backend configuration for devices which support
+ * sample rate less than 44.1
+ */
+ if (sample_rate < OUTPUT_SAMPLING_RATE_44100)
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
if (channels < out_channels)
channels = out_channels;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index a86c200..86e8a42 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -5945,7 +5945,11 @@
bit_width = out->bit_width;
if (sample_rate < out->sample_rate)
sample_rate = out->sample_rate;
- if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
+ /*
+ * TODO: Add Support for Backend configuration for devices which support
+ * sample rate less than 44.1
+ */
+ if (sample_rate < OUTPUT_SAMPLING_RATE_44100)
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
if (channels < out_channels)
channels = out_channels;