From 8edb8dc44b8a2f81bdb5db645b6b708548771a31 Mon Sep 17 00:00:00 2001
From: Andy Hung <hunga@google.com>
Date: Thu, 26 Mar 2015 19:13:55 -0700
Subject: Add playback rate to AudioTrack

Bug: 19196501
Change-Id: I6411e1d3ce652b711a71a6d9df020cb5f60d4714
---
 services/audioflinger/Threads.cpp | 45 +++++++++++++++++++++++++--------------
 services/audioflinger/Tracks.cpp  |  9 ++++++--
 2 files changed, 36 insertions(+), 18 deletions(-)

(limited to 'services')

diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 1a20fae..b30fd20 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1608,13 +1608,19 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
     // If you change this calculation, also review the start threshold which is related.
     if (!(*flags & IAudioFlinger::TRACK_FAST)
             && audio_is_linear_pcm(format) && sharedBuffer == 0) {
+        // this must match AudioTrack.cpp calculateMinFrameCount().
+        // TODO: Move to a common library
         uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
         uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
         if (minBufCount < 2) {
             minBufCount = 2;
         }
+        // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack
+        // or the client should compute and pass in a larger buffer request.
         size_t minFrameCount =
-                minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate);
+                minBufCount * sourceFramesNeededWithTimestretch(
+                        sampleRate, mNormalFrameCount,
+                        mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/);
         if (frameCount < minFrameCount) { // including frameCount == 0
             frameCount = minFrameCount;
         }
@@ -3592,21 +3598,17 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
         // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
         // during last round
         size_t desiredFrames;
-        uint32_t sr = track->sampleRate();
-        if (sr == mSampleRate) {
-            desiredFrames = mNormalFrameCount;
-        } else {
-            desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate);
-            // add frames already consumed but not yet released by the resampler
-            // because mAudioTrackServerProxy->framesReady() will include these frames
-            desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
-#if 0
-            // the minimum track buffer size is normally twice the number of frames necessary
-            // to fill one buffer and the resampler should not leave more than one buffer worth
-            // of unreleased frames after each pass, but just in case...
-            ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
-#endif
-        }
+        const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
+        float speed, pitch;
+        track->mAudioTrackServerProxy->getPlaybackRate(&speed, &pitch);
+
+        desiredFrames = sourceFramesNeededWithTimestretch(
+                sampleRate, mNormalFrameCount, mSampleRate, speed);
+        // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed.
+        // add frames already consumed but not yet released by the resampler
+        // because mAudioTrackServerProxy->framesReady() will include these frames
+        desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+
         uint32_t minFrames = 1;
         if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
                 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
@@ -3769,6 +3771,17 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                 AudioMixer::RESAMPLE,
                 AudioMixer::SAMPLE_RATE,
                 (void *)(uintptr_t)reqSampleRate);
+
+            // set the playback rate as an float array {speed, pitch}
+            float playbackRate[2];
+            track->mAudioTrackServerProxy->getPlaybackRate(
+                    &playbackRate[0] /*speed*/, &playbackRate[1] /*pitch*/);
+            mAudioMixer->setParameter(
+                name,
+                AudioMixer::TIMESTRETCH,
+                AudioMixer::PLAYBACK_RATE,
+                playbackRate);
+
             /*
              * Select the appropriate output buffer for the track.
              *
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 1566b1f..da2d634 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -903,9 +903,14 @@ status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& times
             mPreviousTimestampValid = false;
             return INVALID_OPERATION;
         }
+        // FIXME Not accurate under dynamic changes of sample rate and speed.
+        // Do not use track's mSampleRate as it is not current for mixer tracks.
+        uint32_t sampleRate = mAudioTrackServerProxy->getSampleRate();
+        float speed, pitch;
+        mAudioTrackServerProxy->getPlaybackRate(&speed, &pitch);
         uint32_t unpresentedFrames =
-                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
-                playbackThread->mSampleRate;
+                ((double) playbackThread->mLatchQ.mUnpresentedFrames * sampleRate * speed)
+                / playbackThread->mSampleRate;
         // FIXME Since we're using a raw pointer as the key, it is theoretically possible
         //       for a brand new track to share the same address as a recently destroyed
         //       track, and thus for us to get the frames released of the wrong track.
-- 
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