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| author | Glenn Kasten <gkasten@google.com> | 2013-07-16 17:18:58 -0700 |
|---|---|---|
| committer | Glenn Kasten <gkasten@google.com> | 2013-07-25 12:22:18 -0700 |
| commit | 7fc97ba08e2850f3f16db704b78ce78e3dbe1ff0 (patch) | |
| tree | ada9fda1321150cc75295eece5f7f82e35f0c9f3 /services/audioflinger/FastMixer.cpp | |
| parent | fc5ea08326b510c6f82f71845d95a8758a6ab698 (diff) | |
| download | frameworks_av-7fc97ba08e2850f3f16db704b78ce78e3dbe1ff0.zip frameworks_av-7fc97ba08e2850f3f16db704b78ce78e3dbe1ff0.tar.gz frameworks_av-7fc97ba08e2850f3f16db704b78ce78e3dbe1ff0.tar.bz2 | |
HAL stream format for mixer output threads must be stereo 16-bit PCM
Direct and tunnel output threads can support various HAL stream formats,
included encoded. But currently there are stereo 16-bit PCM assumptions
in several places for mixer and duplicating output threads:
- mMixBuffer and mixBuffer()
- AudioMixer including resampler
- FastMixer's mixBuffer
- effects
- NBAIO_Format
- anywhere FCC_2 is used
- possibly other places
Until those assumptions are removed, this CL enforces stereo 16-bit
PCM in mixer and duplicating threads at the place where the HAL format
is read. It was already being checked in checkForNewParameters_l(),
but not in readOutputParameters().
Change-Id: Ibe344cc922743da234299097aa1bb1f54795cc9b
Diffstat (limited to 'services/audioflinger/FastMixer.cpp')
| -rw-r--r-- | services/audioflinger/FastMixer.cpp | 8 |
1 files changed, 5 insertions, 3 deletions
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp index 5350e2c..ad9f4f2 100644 --- a/services/audioflinger/FastMixer.cpp +++ b/services/audioflinger/FastMixer.cpp @@ -45,6 +45,8 @@ #define MIN_WARMUP_CYCLES 2 // minimum number of loop cycles to wait for warmup #define MAX_WARMUP_CYCLES 10 // maximum number of loop cycles to wait for warmup +#define FCC_2 2 // fixed channel count assumption + namespace android { // Fast mixer thread @@ -225,7 +227,7 @@ bool FastMixer::threadLoop() } else { format = outputSink->format(); sampleRate = Format_sampleRate(format); - ALOG_ASSERT(Format_channelCount(format) == 2); + ALOG_ASSERT(Format_channelCount(format) == FCC_2); } dumpState->mSampleRate = sampleRate; } @@ -241,7 +243,7 @@ bool FastMixer::threadLoop() // implementation; it would be better to have normal mixer allocate for us // to avoid blocking here and to prevent possible priority inversion mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks); - mixBuffer = new short[frameCount * 2]; + mixBuffer = new short[frameCount * FCC_2]; periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00 underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75 overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50 @@ -438,7 +440,7 @@ bool FastMixer::threadLoop() //bool didFullWrite = false; // dumpsys could display a count of partial writes if ((command & FastMixerState::WRITE) && (outputSink != NULL) && (mixBuffer != NULL)) { if (mixBufferState == UNDEFINED) { - memset(mixBuffer, 0, frameCount * 2 * sizeof(short)); + memset(mixBuffer, 0, frameCount * FCC_2 * sizeof(short)); mixBufferState = ZEROED; } if (teeSink != NULL) { |