AudioFlinger: more DTS passthrough support

Pass format for AC3 or DTS to SPDIF encoder.

Bug: 18292317
Change-Id: I6d80eb548cb431bac3390ebb3235146171344291
Signed-off-by: Phil Burk <philburk@google.com>

3 files changed, 29 insertions, 19 deletions

diff --git a/services/audioflinger/AudioHwDevice.cpp b/services/audioflinger/AudioHwDevice.cpp
index 09d86ea..3191598 100644
--- a/services/audioflinger/AudioHwDevice.cpp
+++ b/services/audioflinger/AudioHwDevice.cpp
@@ -44,7 +44,7 @@ status_t AudioHwDevice::openOutputStream(
     AudioStreamOut *outputStream = new AudioStreamOut(this, flags);
 
     // Try to open the HAL first using the current format.
-    ALOGV("AudioHwDevice::openOutputStream(), try "
+    ALOGV("openOutputStream(), try "
             " sampleRate %d, Format %#x, "
             "channelMask %#x",
             config->sample_rate,
@@ -59,7 +59,7 @@ status_t AudioHwDevice::openOutputStream(
         // FIXME Look at any modification to the config.
         // The HAL might modify the config to suggest a wrapped format.
         // Log this so we can see what the HALs are doing.
-        ALOGI("AudioHwDevice::openOutputStream(), HAL returned"
+        ALOGI("openOutputStream(), HAL returned"
             " sampleRate %d, Format %#x, "
             "channelMask %#x, status %d",
             config->sample_rate,
@@ -72,16 +72,19 @@ status_t AudioHwDevice::openOutputStream(
                 && ((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
                 && ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
 
-        // FIXME - Add isEncodingSupported() query to SPDIF wrapper then
-        // call it from here.
         if (wrapperNeeded) {
-            outputStream = new SpdifStreamOut(this, flags);
-            status = outputStream->open(handle, devices, &originalConfig, address);
-            if (status != NO_ERROR) {
-                ALOGE("ERROR - AudioHwDevice::openOutputStream(), SPDIF open returned %d",
-                    status);
-                delete outputStream;
-                outputStream = NULL;
+            if (SPDIFEncoder::isFormatSupported(originalConfig.format)) {
+                outputStream = new SpdifStreamOut(this, flags, originalConfig.format);
+                status = outputStream->open(handle, devices, &originalConfig, address);
+                if (status != NO_ERROR) {
+                    ALOGE("ERROR - openOutputStream(), SPDIF open returned %d",
+                        status);
+                    delete outputStream;
+                    outputStream = NULL;
+                }
+            } else {
+                ALOGE("ERROR - openOutputStream(), SPDIFEncoder does not support format 0x%08x",
+                    originalConfig.format);
             }
         }
     }
diff --git a/services/audioflinger/SpdifStreamOut.cpp b/services/audioflinger/SpdifStreamOut.cpp
index d23588e..45b541a 100644
--- a/services/audioflinger/SpdifStreamOut.cpp
+++ b/services/audioflinger/SpdifStreamOut.cpp
@@ -32,10 +32,12 @@ namespace android {
  * If the AudioFlinger is processing encoded data and the HAL expects
  * PCM then we need to wrap the data in an SPDIF wrapper.
  */
-SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags)
+SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev,
+            audio_output_flags_t flags,
+            audio_format_t format)
         : AudioStreamOut(dev,flags)
         , mRateMultiplier(1)
-        , mSpdifEncoder(this)
+        , mSpdifEncoder(this, format)
         , mRenderPositionHal(0)
         , mPreviousHalPosition32(0)
 {
@@ -49,15 +51,15 @@ status_t SpdifStreamOut::open(
 {
     struct audio_config customConfig = *config;
 
-    customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
-    customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
-
     // Some data bursts run at a higher sample rate.
+    // TODO Move this into the audio_utils as a static method.
     switch(config->format) {
         case AUDIO_FORMAT_E_AC3:
             mRateMultiplier = 4;
             break;
         case AUDIO_FORMAT_AC3:
+        case AUDIO_FORMAT_DTS:
+        case AUDIO_FORMAT_DTS_HD:
             mRateMultiplier = 1;
             break;
         default:
@@ -67,6 +69,9 @@ status_t SpdifStreamOut::open(
     }
     customConfig.sample_rate = config->sample_rate * mRateMultiplier;
 
+    customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+
     // Always print this because otherwise it could be very confusing if the
     // HAL and AudioFlinger are using different formats.
     // Print before open() because HAL may modify customConfig.
diff --git a/services/audioflinger/SpdifStreamOut.h b/services/audioflinger/SpdifStreamOut.h
index cb82ac7..d81c064 100644
--- a/services/audioflinger/SpdifStreamOut.h
+++ b/services/audioflinger/SpdifStreamOut.h
@@ -38,7 +38,8 @@ namespace android {
 class SpdifStreamOut : public AudioStreamOut {
 public:
 
-    SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags);
+    SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags,
+            audio_format_t format);
 
     virtual ~SpdifStreamOut() { }
 
@@ -77,8 +78,9 @@ private:
     class MySPDIFEncoder : public SPDIFEncoder
     {
     public:
-        MySPDIFEncoder(SpdifStreamOut *spdifStreamOut)
-          : mSpdifStreamOut(spdifStreamOut)
+        MySPDIFEncoder(SpdifStreamOut *spdifStreamOut, audio_format_t format)
+          :  SPDIFEncoder(format)
+          , mSpdifStreamOut(spdifStreamOut)
         {
         }