From 0d6db582f2ccc9f8943c5f3965e2994b7d137158 Mon Sep 17 00:00:00 2001
From: Eric Laurent <elaurent@google.com>
Date: Wed, 12 Nov 2014 18:39:44 -0800
Subject: remove AudioSystem::getOutputSamplingRateForAttr()

It is safer to query the output sampling rate once the output
have been acquired by calling getOutputForAttr().

Bug: 16009464.
Change-Id: Ib561facd19fba5359a6b837d75f1a1cd2dc51b29
---
 include/media/AudioSystem.h    |  2 --
 media/libmedia/AudioSystem.cpp | 13 -------------
 media/libmedia/AudioTrack.cpp  | 30 +++++++++++++++---------------
 3 files changed, 15 insertions(+), 30 deletions(-)

diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 6a0f2a6..f3b7fbb 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -99,8 +99,6 @@ public:
     // to be non-zero if status == NO_ERROR
     static status_t getOutputSamplingRate(uint32_t* samplingRate,
             audio_stream_type_t stream);
-    static status_t getOutputSamplingRateForAttr(uint32_t* samplingRate,
-                const audio_attributes_t *attr);
     static status_t getOutputFrameCount(size_t* frameCount,
             audio_stream_type_t stream);
     static status_t getOutputLatency(uint32_t* latency,
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index f5c3383..fce4389 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -247,19 +247,6 @@ status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream
     return getSamplingRate(output, samplingRate);
 }
 
-status_t AudioSystem::getOutputSamplingRateForAttr(uint32_t* samplingRate,
-        const audio_attributes_t *attr)
-{
-    if (attr == NULL) {
-        return BAD_VALUE;
-    }
-    audio_io_handle_t output = getOutputForAttr(attr);
-    if (output == 0) {
-        return PERMISSION_DENIED;
-    }
-    return getSamplingRate(output, samplingRate);
-}
-
 status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
                                       uint32_t* samplingRate)
 {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index cd493f6..2f57b9d 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -304,17 +304,6 @@ status_t AudioTrack::set(
                 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
     }
 
-    status_t status;
-    if (sampleRate == 0) {
-        status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes);
-        if (status != NO_ERROR) {
-            ALOGE("Could not get output sample rate for stream type %d; status %d",
-                    mStreamType, status);
-            return status;
-        }
-    }
-    mSampleRate = sampleRate;
-
     // these below should probably come from the audioFlinger too...
     if (format == AUDIO_FORMAT_DEFAULT) {
         format = AUDIO_FORMAT_PCM_16_BIT;
@@ -373,6 +362,12 @@ status_t AudioTrack::set(
         // so no need to check for specific PCM formats here
     }
 
+    // sampling rate must be specified for direct outputs
+    if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+        return BAD_VALUE;
+    }
+    mSampleRate = sampleRate;
+
     // Make copy of input parameter offloadInfo so that in the future:
     //  (a) createTrack_l doesn't need it as an input parameter
     //  (b) we can support re-creation of offloaded tracks
@@ -413,7 +408,7 @@ status_t AudioTrack::set(
     }
 
     // create the IAudioTrack
-    status = createTrack_l();
+    status_t status = createTrack_l();
 
     if (status != NO_ERROR) {
         if (mAudioTrackThread != 0) {
@@ -680,15 +675,18 @@ status_t AudioTrack::setSampleRate(uint32_t rate)
         return INVALID_OPERATION;
     }
 
+    AutoMutex lock(mLock);
+    if (mOutput == AUDIO_IO_HANDLE_NONE) {
+        return NO_INIT;
+    }
     uint32_t afSamplingRate;
-    if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {
+    if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
         return NO_INIT;
     }
     if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
         return BAD_VALUE;
     }
 
-    AutoMutex lock(mLock);
     mSampleRate = rate;
     mProxy->setSampleRate(rate);
 
@@ -963,7 +961,9 @@ status_t AudioTrack::createTrack_l()
         ALOGE("getSamplingRate(output=%d) status %d", output, status);
         goto release;
     }
-
+    if (mSampleRate == 0) {
+        mSampleRate = afSampleRate;
+    }
     // Client decides whether the track is TIMED (see below), but can only express a preference
     // for FAST.  Server will perform additional tests.
     if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !((
-- 
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