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-rw-r--r--services/audioflinger/AudioFlinger.cpp72
1 files changed, 69 insertions, 3 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index fab1ef5..23215dd 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -13,6 +13,25 @@
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
+**
+** This file was modified by DTS, Inc. The portions of the
+** code that are surrounded by "DTS..." are copyrighted and
+** licensed separately, as follows:
+**
+** (C) 2015 DTS, Inc.
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+**
*/
@@ -64,6 +83,9 @@
#include <media/nbaio/PipeReader.h>
#include <media/AudioParameter.h>
#include <private/android_filesystem_config.h>
+#ifdef SRS_PROCESSING
+#include "postpro_patch.h"
+#endif
// ----------------------------------------------------------------------------
@@ -131,6 +153,14 @@ const char *formatToString(audio_format_t format) {
case AUDIO_FORMAT_OPUS: return "opus";
case AUDIO_FORMAT_AC3: return "ac-3";
case AUDIO_FORMAT_E_AC3: return "e-ac-3";
+ case AUDIO_FORMAT_PCM_OFFLOAD:
+ switch (format) {
+ case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD: return "pcm-16bit-offload";
+ case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD: return "pcm-24bit-offload";
+ default:
+ break;
+ }
+ break;
default:
break;
}
@@ -1043,6 +1073,13 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
if (ioHandle == AUDIO_IO_HANDLE_NONE) {
Mutex::Autolock _l(mLock);
status_t final_result = NO_ERROR;
+#ifdef SRS_PROCESSING
+ POSTPRO_PATCH_PARAMS_SET(keyValuePairs);
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ thread->setPostPro();
+ }
+#endif
{
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_PARAMETER;
@@ -1053,9 +1090,24 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
}
mHardwareStatus = AUDIO_HW_IDLE;
}
- // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
+
AudioParameter param = AudioParameter(keyValuePairs);
- String8 value;
+ String8 value, key;
+ key = String8("SND_CARD_STATUS");
+ if (param.get(key, value) == NO_ERROR) {
+ ALOGV("Set keySoundCardStatus:%s", value.string());
+ if ((value.find("OFFLINE", 0) != -1) ) {
+ ALOGV("OFFLINE detected - call InvalidateTracks()");
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ if( thread->getOutput()->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ){
+ thread->invalidateTracks(AUDIO_STREAM_MUSIC);
+ }
+ }
+ }
+ }
+
+ // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
if (mBtNrecIsOff != btNrecIsOff) {
@@ -1122,6 +1174,9 @@ String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& k
if (ioHandle == AUDIO_IO_HANDLE_NONE) {
String8 out_s8;
+#ifdef SRS_PROCESSING
+ POSTPRO_PATCH_PARAMS_GET(keys, out_s8);
+#endif
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
char *s;
@@ -1347,6 +1402,12 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId,
+void AudioFlinger::PlaybackThread::setPostPro()
+{
+ Mutex::Autolock _l(mLock);
+ if (mType == OFFLOAD)
+ broadcast_l();
+}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
@@ -1822,7 +1883,11 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
- ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
+ ALOGV("openOutput_l() created direct output: ID %d thread %p ", *output, thread);
+ //Check if this is DirectPCM, if so
+ if (flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ thread->mIsDirectPcm = true;
+ }
} else {
thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
@@ -2964,6 +3029,7 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
bool firstRead = true;
#define TEE_SINK_READ 1024 // frames per I/O operation
void *buffer = malloc(TEE_SINK_READ * frameSize);
+ ALOG_ASSERT(buffer != NULL);
for (;;) {
size_t count = TEE_SINK_READ;
ssize_t actual = teeSource->read(buffer, count,