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| author | Andy Hung <hunga@google.com> | 2014-06-02 17:27:26 +0000 |
|---|---|---|
| committer | Android (Google) Code Review <android-gerrit@google.com> | 2014-06-02 17:27:27 +0000 |
| commit | d28148537693b19320f3c3b5b34816d45542e9bd (patch) | |
| tree | 83024443aa56ab67e402b3af0c7bb822ce6ffa6d /services | |
| parent | 24c01a8417fe195e5ba2187dbbdf9bd1e3a6553f (diff) | |
| parent | f8a106ae5c40735de6c75e7c12d5e0e2e22f12cf (diff) | |
| download | frameworks_av-d28148537693b19320f3c3b5b34816d45542e9bd.zip frameworks_av-d28148537693b19320f3c3b5b34816d45542e9bd.tar.gz frameworks_av-d28148537693b19320f3c3b5b34816d45542e9bd.tar.bz2 | |
Merge "Fix floating point output from mixer"
Diffstat (limited to 'services')
| -rw-r--r-- | services/audioflinger/AudioMixer.cpp | 13 |
1 files changed, 7 insertions, 6 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index ace3bf1..a0f2b7d 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -1450,6 +1450,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, AudioBufferProvider::Buffer& b(t.buffer); int32_t* out = t.mainBuffer; + float *fout = reinterpret_cast<float*>(out); size_t numFrames = state->frameCount; const int16_t vl = t.volume[0]; @@ -1463,9 +1464,10 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, // in == NULL can happen if the track was flushed just after having // been enabled for mixing. - if (in == NULL || ((unsigned long)in & 3)) { - memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); - ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: " + if (in == NULL || (((uintptr_t)in) & 3)) { + memset(out, 0, numFrames + * MAX_NUM_CHANNELS * audio_bytes_per_sample(t.mMixerFormat)); + ALOGE_IF((((uintptr_t)in) & 3), "process stereo track: input buffer alignment pb: " "buffer %p track %d, channels %d, needs %08x", in, i, t.channelCount, t.needs); return; @@ -1473,8 +1475,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, size_t outFrames = b.frameCount; switch (t.mMixerFormat) { - case AUDIO_FORMAT_PCM_FLOAT: { - float *fout = reinterpret_cast<float*>(out); + case AUDIO_FORMAT_PCM_FLOAT: do { uint32_t rl = *reinterpret_cast<const uint32_t *>(in); in += 2; @@ -1485,7 +1486,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, // Note: In case of later int16_t sink output, // conversion and clamping is done by memcpy_to_i16_from_float(). } while (--outFrames); - } break; + break; case AUDIO_FORMAT_PCM_16_BIT: if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) { // volume is boosted, so we might need to clamp even though |