diff options
Diffstat (limited to 'services')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 72 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.h | 10 | ||||
| -rw-r--r-- | services/audioflinger/AudioMixer.cpp | 100 | ||||
| -rw-r--r-- | services/audioflinger/AudioMixer.h | 26 | ||||
| -rw-r--r-- | services/audioflinger/AudioPolicyService.cpp | 36 | ||||
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 6 | ||||
| -rw-r--r-- | services/audioflinger/AudioResamplerCubic.cpp | 12 | ||||
| -rw-r--r-- | services/audioflinger/Effects.cpp | 3 | ||||
| -rw-r--r-- | services/audioflinger/FastMixer.cpp | 3 | ||||
| -rw-r--r-- | services/audioflinger/PlaybackTracks.h | 1 | ||||
| -rw-r--r-- | services/audioflinger/RecordTracks.h | 1 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 656 | ||||
| -rw-r--r-- | services/audioflinger/Threads.h | 37 | ||||
| -rw-r--r-- | services/audioflinger/TrackBase.h | 10 | ||||
| -rw-r--r-- | services/audioflinger/Tracks.cpp | 19 |
15 files changed, 554 insertions, 438 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 3132e54..c9c9f8a 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -162,12 +162,15 @@ AudioFlinger::AudioFlinger() (void) property_get("af.tee", value, "0"); teeEnabled = atoi(value); } - if (teeEnabled & 1) + if (teeEnabled & 1) { mTeeSinkInputEnabled = true; - if (teeEnabled & 2) + } + if (teeEnabled & 2) { mTeeSinkOutputEnabled = true; - if (teeEnabled & 4) + } + if (teeEnabled & 4) { mTeeSinkTrackEnabled = true; + } #endif } @@ -513,10 +516,12 @@ sp<IAudioTrack> AudioFlinger::createTrack( track = thread->createTrack_l(client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); + // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless // move effect chain to this output thread if an effect on same session was waiting // for a track to be created if (lStatus == NO_ERROR && effectThread != NULL) { + // no risk of deadlock because AudioFlinger::mLock is held Mutex::Autolock _dl(thread->mLock); Mutex::Autolock _sl(effectThread->mLock); moveEffectChain_l(lSessionId, effectThread, thread, true); @@ -536,7 +541,9 @@ sp<IAudioTrack> AudioFlinger::createTrack( } } } + } + if (lStatus == NO_ERROR) { // s for server's pid, n for normal mixer name, f for fast index name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, @@ -550,9 +557,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( } Exit: - if (status != NULL) { - *status = lStatus; - } + *status = lStatus; return trackHandle; } @@ -1293,6 +1298,7 @@ sp<IAudioRecord> AudioFlinger::openRecord( flags, tid, &lStatus); LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR)); } + if (lStatus != NO_ERROR) { // remove local strong reference to Client before deleting the RecordTrack so that the // Client destructor is called by the TrackBase destructor with mLock held @@ -1301,14 +1307,11 @@ sp<IAudioRecord> AudioFlinger::openRecord( goto Exit; } - // return to handle to client + // return handle to client recordHandle = new RecordHandle(recordTrack); - lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return recordHandle; } @@ -1449,18 +1452,15 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { - PlaybackThread *thread = NULL; struct audio_config config; + memset(&config, 0, sizeof(config)); config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; - if (offloadInfo) { + if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } - audio_stream_out_t *outStream = NULL; - AudioHwDevice *outHwDev; - ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", module, (pDevices != NULL) ? *pDevices : 0, @@ -1469,7 +1469,7 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, config.channel_mask, flags); ALOGV("openOutput(), offloadInfo %p version 0x%04x", - offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); + offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); if (pDevices == NULL || *pDevices == 0) { return 0; @@ -1477,15 +1477,17 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, Mutex::Autolock _l(mLock); - outHwDev = findSuitableHwDev_l(module, *pDevices); - if (outHwDev == NULL) + AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); + if (outHwDev == NULL) { return 0; + } audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); audio_io_handle_t id = nextUniqueId(); mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; + audio_stream_out_t *outStream = NULL; status_t status = hwDevHal->open_output_stream(hwDevHal, id, *pDevices, @@ -1505,6 +1507,7 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, if (status == NO_ERROR && outStream != NULL) { AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); + PlaybackThread *thread; if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { thread = new OffloadThread(this, output, id, *pDevices); ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); @@ -1672,18 +1675,15 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask) { - status_t status; - RecordThread *thread = NULL; struct audio_config config; + memset(&config, 0, sizeof(config)); config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; uint32_t reqSamplingRate = config.sample_rate; audio_format_t reqFormat = config.format; - audio_channel_mask_t reqChannels = config.channel_mask; - audio_stream_in_t *inStream = NULL; - AudioHwDevice *inHwDev; + audio_channel_mask_t reqChannelMask = config.channel_mask; if (pDevices == NULL || *pDevices == 0) { return 0; @@ -1691,14 +1691,16 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, Mutex::Autolock _l(mLock); - inHwDev = findSuitableHwDev_l(module, *pDevices); - if (inHwDev == NULL) + AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); + if (inHwDev == NULL) { return 0; + } audio_hw_device_t *inHwHal = inHwDev->hwDevice(); audio_io_handle_t id = nextUniqueId(); - status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, + audio_stream_in_t *inStream = NULL; + status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " "status %d", @@ -1714,10 +1716,12 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, if (status == BAD_VALUE && reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && (config.sample_rate <= 2 * reqSamplingRate) && - (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { + (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) { + // FIXME describe the change proposed by HAL (save old values so we can log them here) ALOGV("openInput() reopening with proposed sampling rate and channel mask"); inStream = NULL; status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); + // FIXME log this new status; HAL should not propose any further changes } if (status == NO_ERROR && inStream != NULL) { @@ -1776,10 +1780,10 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, // Start record thread // RecordThread requires both input and output device indication to forward to audio // pre processing modules - thread = new RecordThread(this, + RecordThread *thread = new RecordThread(this, input, reqSamplingRate, - reqChannels, + reqChannelMask, id, primaryOutputDevice_l(), *pDevices @@ -1796,7 +1800,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, *pFormat = config.format; } if (pChannelMask != NULL) { - *pChannelMask = reqChannels; + *pChannelMask = reqChannelMask; } // notify client processes of the new input creation @@ -1954,7 +1958,7 @@ void AudioFlinger::purgeStaleEffects_l() { } } if (!found) { - Mutex::Autolock _l (t->mLock); + Mutex::Autolock _l(t->mLock); // remove all effects from the chain while (ec->mEffects.size()) { sp<EffectModule> effect = ec->mEffects[0]; @@ -2249,9 +2253,7 @@ sp<IEffect> AudioFlinger::createEffect( } Exit: - if (status != NULL) { - *status = lStatus; - } + *status = lStatus; return handle; } diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 53e238e..9137040 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -110,7 +110,7 @@ public: int *sessionId, String8& name, int clientUid, - status_t *status); + status_t *status /*non-NULL*/); virtual sp<IAudioRecord> openRecord( audio_io_handle_t input, @@ -121,7 +121,7 @@ public: IAudioFlinger::track_flags_t *flags, pid_t tid, int *sessionId, - status_t *status); + status_t *status /*non-NULL*/); virtual uint32_t sampleRate(audio_io_handle_t output) const; virtual int channelCount(audio_io_handle_t output) const; @@ -210,7 +210,7 @@ public: int32_t priority, audio_io_handle_t io, int sessionId, - status_t *status, + status_t *status /*non-NULL*/, int *id, int *enabled); @@ -499,7 +499,7 @@ private: private: const char * const mModuleName; audio_hw_device_t * const mHwDevice; - Flags mFlags; + const Flags mFlags; }; // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. @@ -509,7 +509,7 @@ private: struct AudioStreamOut { AudioHwDevice* const audioHwDev; audio_stream_out_t* const stream; - audio_output_flags_t flags; + const audio_output_flags_t flags; audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index df4e029..8bea752 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -58,7 +58,7 @@ AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider() status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer, int64_t pts) { //ALOGV("DownmixerBufferProvider::getNextBuffer()"); - if (this->mTrackBufferProvider != NULL) { + if (mTrackBufferProvider != NULL) { status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts); if (res == OK) { mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount; @@ -81,7 +81,7 @@ status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider: void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) { //ALOGV("DownmixerBufferProvider::releaseBuffer()"); - if (this->mTrackBufferProvider != NULL) { + if (mTrackBufferProvider != NULL) { mTrackBufferProvider->releaseBuffer(pBuffer); } else { ALOGE("DownmixerBufferProvider::releaseBuffer() error: NULL track buffer provider"); @@ -90,9 +90,9 @@ void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buf // ---------------------------------------------------------------------------- -bool AudioMixer::isMultichannelCapable = false; +bool AudioMixer::sIsMultichannelCapable = false; -effect_descriptor_t AudioMixer::dwnmFxDesc; +effect_descriptor_t AudioMixer::sDwnmFxDesc; // Ensure mConfiguredNames bitmask is initialized properly on all architectures. // The value of 1 << x is undefined in C when x >= 32. @@ -113,8 +113,6 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr // AudioMixer is not yet capable of multi-channel output beyond stereo ALOG_ASSERT(2 == MAX_NUM_CHANNELS, "bad MAX_NUM_CHANNELS %d", MAX_NUM_CHANNELS); - LocalClock lc; - pthread_once(&sOnceControl, &sInitRoutine); mState.enabledTracks= 0; @@ -136,27 +134,6 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr t++; } - // find multichannel downmix effect if we have to play multichannel content - uint32_t numEffects = 0; - int ret = EffectQueryNumberEffects(&numEffects); - if (ret != 0) { - ALOGE("AudioMixer() error %d querying number of effects", ret); - return; - } - ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects); - - for (uint32_t i = 0 ; i < numEffects ; i++) { - if (EffectQueryEffect(i, &dwnmFxDesc) == 0) { - ALOGV("effect %d is called %s", i, dwnmFxDesc.name); - if (memcmp(&dwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) { - ALOGI("found effect \"%s\" from %s", - dwnmFxDesc.name, dwnmFxDesc.implementor); - isMultichannelCapable = true; - break; - } - } - } - ALOGE_IF(!isMultichannelCapable, "unable to find downmix effect"); } AudioMixer::~AudioMixer() @@ -229,7 +206,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) void AudioMixer::invalidateState(uint32_t mask) { - if (mask) { + if (mask != 0) { mState.needsChanged |= mask; mState.hook = process__validate; } @@ -276,13 +253,13 @@ status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName) DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(); int32_t status; - if (!isMultichannelCapable) { + if (!sIsMultichannelCapable) { ALOGE("prepareTrackForDownmix(%d) fails: mixer doesn't support multichannel content", trackName); goto noDownmixForActiveTrack; } - if (EffectCreate(&dwnmFxDesc.uuid, + if (EffectCreate(&sDwnmFxDesc.uuid, pTrack->sessionId /*sessionId*/, -2 /*ioId not relevant here, using random value*/, &pDbp->mDownmixHandle/*pHandle*/) != 0) { ALOGE("prepareTrackForDownmix(%d) fails: error creating downmixer effect", trackName); @@ -566,7 +543,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) resampler = AudioResampler::create( format, // the resampler sees the number of channels after the downmixer, if any - downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount, + (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount), devSampleRate, quality); resampler->setLocalTimeFreq(sLocalTimeFreq); } @@ -667,27 +644,29 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) countActiveTracks++; track_t& t = state->tracks[i]; uint32_t n = 0; + // FIXME can overflow (mask is only 3 bits) n |= NEEDS_CHANNEL_1 + t.channelCount - 1; - n |= NEEDS_FORMAT_16; - n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED; + if (t.doesResample()) { + n |= NEEDS_RESAMPLE; + } if (t.auxLevel != 0 && t.auxBuffer != NULL) { - n |= NEEDS_AUX_ENABLED; + n |= NEEDS_AUX; } if (t.volumeInc[0]|t.volumeInc[1]) { volumeRamp = true; } else if (!t.doesResample() && t.volumeRL == 0) { - n |= NEEDS_MUTE_ENABLED; + n |= NEEDS_MUTE; } t.needs = n; - if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) { + if (n & NEEDS_MUTE) { t.hook = track__nop; } else { - if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) { + if (n & NEEDS_AUX) { all16BitsStereoNoResample = false; } - if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { + if (n & NEEDS_RESAMPLE) { all16BitsStereoNoResample = false; resampling = true; t.hook = track__genericResample; @@ -709,7 +688,7 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) // select the processing hooks state->hook = process__nop; - if (countActiveTracks) { + if (countActiveTracks > 0) { if (resampling) { if (!state->outputTemp) { state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; @@ -745,16 +724,15 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) // Now that the volume ramp has been done, set optimal state and // track hooks for subsequent mixer process - if (countActiveTracks) { + if (countActiveTracks > 0) { bool allMuted = true; uint32_t en = state->enabledTracks; while (en) { const int i = 31 - __builtin_clz(en); en &= ~(1<<i); track_t& t = state->tracks[i]; - if (!t.doesResample() && t.volumeRL == 0) - { - t.needs |= NEEDS_MUTE_ENABLED; + if (!t.doesResample() && t.volumeRL == 0) { + t.needs |= NEEDS_MUTE; t.hook = track__nop; } else { allMuted = false; @@ -1124,8 +1102,9 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) t.in = t.buffer.raw; // t.in == NULL can happen if the track was flushed just after having // been enabled for mixing. - if (t.in == NULL) + if (t.in == NULL) { enabledTracks &= ~(1<<i); + } } e0 = enabledTracks; @@ -1157,12 +1136,12 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) track_t& t = state->tracks[i]; size_t outFrames = BLOCKSIZE; int32_t *aux = NULL; - if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) { + if (CC_UNLIKELY(t.needs & NEEDS_AUX)) { aux = t.auxBuffer + numFrames; } while (outFrames) { size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; - if (inFrames) { + if (inFrames > 0) { t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux); t.frameCount -= inFrames; @@ -1238,14 +1217,14 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) e1 &= ~(1<<i); track_t& t = state->tracks[i]; int32_t *aux = NULL; - if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) { + if (CC_UNLIKELY(t.needs & NEEDS_AUX)) { aux = t.auxBuffer; } // this is a little goofy, on the resampling case we don't // acquire/release the buffers because it's done by // the resampler. - if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { + if (t.needs & NEEDS_RESAMPLE) { t.resampler->setPTS(pts); t.hook(&t, outTemp, numFrames, state->resampleTemp, aux); } else { @@ -1445,8 +1424,9 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS, int outputFrameIndex) { - if (AudioBufferProvider::kInvalidPTS == basePTS) + if (AudioBufferProvider::kInvalidPTS == basePTS) { return AudioBufferProvider::kInvalidPTS; + } return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate); } @@ -1458,6 +1438,28 @@ int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS, { LocalClock lc; sLocalTimeFreq = lc.getLocalFreq(); + + // find multichannel downmix effect if we have to play multichannel content + uint32_t numEffects = 0; + int ret = EffectQueryNumberEffects(&numEffects); + if (ret != 0) { + ALOGE("AudioMixer() error %d querying number of effects", ret); + return; + } + ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects); + + for (uint32_t i = 0 ; i < numEffects ; i++) { + if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) { + ALOGV("effect %d is called %s", i, sDwnmFxDesc.name); + if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) { + ALOGI("found effect \"%s\" from %s", + sDwnmFxDesc.name, sDwnmFxDesc.implementor); + sIsMultichannelCapable = true; + break; + } + } + } + ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect"); } // ---------------------------------------------------------------------------- diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h index 43aeb86..d5c9da7 100644 --- a/services/audioflinger/AudioMixer.h +++ b/services/audioflinger/AudioMixer.h @@ -120,27 +120,19 @@ public: private: enum { + // FIXME this representation permits up to 8 channels NEEDS_CHANNEL_COUNT__MASK = 0x00000007, - NEEDS_FORMAT__MASK = 0x000000F0, - NEEDS_MUTE__MASK = 0x00000100, - NEEDS_RESAMPLE__MASK = 0x00001000, - NEEDS_AUX__MASK = 0x00010000, }; enum { - NEEDS_CHANNEL_1 = 0x00000000, - NEEDS_CHANNEL_2 = 0x00000001, + NEEDS_CHANNEL_1 = 0x00000000, // mono + NEEDS_CHANNEL_2 = 0x00000001, // stereo - NEEDS_FORMAT_16 = 0x00000010, + // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT - NEEDS_MUTE_DISABLED = 0x00000000, - NEEDS_MUTE_ENABLED = 0x00000100, - - NEEDS_RESAMPLE_DISABLED = 0x00000000, - NEEDS_RESAMPLE_ENABLED = 0x00001000, - - NEEDS_AUX_DISABLED = 0x00000000, - NEEDS_AUX_ENABLED = 0x00010000, + NEEDS_MUTE = 0x00000100, + NEEDS_RESAMPLE = 0x00001000, + NEEDS_AUX = 0x00010000, }; struct state_t; @@ -256,9 +248,9 @@ private: state_t mState __attribute__((aligned(32))); // effect descriptor for the downmixer used by the mixer - static effect_descriptor_t dwnmFxDesc; + static effect_descriptor_t sDwnmFxDesc; // indicates whether a downmix effect has been found and is usable by this mixer - static bool isMultichannelCapable; + static bool sIsMultichannelCapable; // Call after changing either the enabled status of a track, or parameters of an enabled track. // OK to call more often than that, but unnecessary. diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index 35e816b..c5ad2c0 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -77,24 +77,28 @@ AudioPolicyService::AudioPolicyService() mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this); /* instantiate the audio policy manager */ rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module); - if (rc) + if (rc) { return; + } rc = audio_policy_dev_open(module, &mpAudioPolicyDev); ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc)); - if (rc) + if (rc) { return; + } rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this, &mpAudioPolicy); ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc)); - if (rc) + if (rc) { return; + } rc = mpAudioPolicy->init_check(mpAudioPolicy); ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc)); - if (rc) + if (rc) { return; + } ALOGI("Loaded audio policy from %s (%s)", module->name, module->id); @@ -126,10 +130,12 @@ AudioPolicyService::~AudioPolicyService() } mInputs.clear(); - if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) + if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) { mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy); - if (mpAudioPolicyDev != NULL) + } + if (mpAudioPolicyDev != NULL) { audio_policy_dev_close(mpAudioPolicyDev); + } } status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, @@ -1114,11 +1120,13 @@ int AudioPolicyService::setStreamVolume(audio_stream_type_t stream, int AudioPolicyService::startTone(audio_policy_tone_t tone, audio_stream_type_t stream) { - if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) + if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) { ALOGE("startTone: illegal tone requested (%d)", tone); - if (stream != AUDIO_STREAM_VOICE_CALL) + } + if (stream != AUDIO_STREAM_VOICE_CALL) { ALOGE("startTone: illegal stream (%d) requested for tone %d", stream, tone); + } mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING, AUDIO_STREAM_VOICE_CALL); return 0; @@ -1509,8 +1517,9 @@ static audio_io_handle_t aps_open_dup_output(void *service, static int aps_close_output(void *service, audio_io_handle_t output) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->closeOutput(output); } @@ -1573,8 +1582,9 @@ static audio_io_handle_t aps_open_input_on_module(void *service, static int aps_close_input(void *service, audio_io_handle_t input) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->closeInput(input); } @@ -1583,8 +1593,9 @@ static int aps_set_stream_output(void *service, audio_stream_type_t stream, audio_io_handle_t output) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->setStreamOutput(stream, output); } @@ -1594,8 +1605,9 @@ static int aps_move_effects(void *service, int session, audio_io_handle_t dst_output) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->moveEffects(session, src_output, dst_output); } diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index 2c3c719..323f1a4 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -339,8 +339,9 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction); out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction); Advance(&inputIndex, &phaseFraction, phaseIncrement); - if (outputIndex == outputSampleCount) + if (outputIndex == outputSampleCount) { break; + } } // process input samples @@ -434,8 +435,9 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, out[outputIndex++] += vl * sample; out[outputIndex++] += vr * sample; Advance(&inputIndex, &phaseFraction, phaseIncrement); - if (outputIndex == outputSampleCount) + if (outputIndex == outputSampleCount) { break; + } } // process input samples diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp index 18e59e9..1f9714b 100644 --- a/services/audioflinger/AudioResamplerCubic.cpp +++ b/services/audioflinger/AudioResamplerCubic.cpp @@ -66,8 +66,9 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, if (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { return; + } // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); } int16_t *in = mBuffer.i16; @@ -97,8 +98,9 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, calculateOutputPTS(outputIndex / 2)); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { goto save_state; // ugly, but efficient + } in = mBuffer.i16; // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); } @@ -132,8 +134,9 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, if (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { return; + } // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); } int16_t *in = mBuffer.i16; @@ -163,8 +166,9 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, calculateOutputPTS(outputIndex / 2)); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { goto save_state; // ugly, but efficient + } // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); in = mBuffer.i16; } diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp index a8a5169..bb98a35 100644 --- a/services/audioflinger/Effects.cpp +++ b/services/audioflinger/Effects.cpp @@ -116,8 +116,9 @@ status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) continue; } // first non destroyed handle is considered in control - if (controlHandle == NULL) + if (controlHandle == NULL) { controlHandle = h; + } if (h->priority() <= priority) { break; } diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp index f27ea17..7126e92 100644 --- a/services/audioflinger/FastMixer.cpp +++ b/services/audioflinger/FastMixer.cpp @@ -459,8 +459,9 @@ bool FastMixer::threadLoop() } int64_t pts; - if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts))) + if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts))) { pts = AudioBufferProvider::kInvalidPTS; + } // process() is CPU-bound mixer->process(pts); diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h index 43b77f3..4b6c74d 100644 --- a/services/audioflinger/PlaybackTracks.h +++ b/services/audioflinger/PlaybackTracks.h @@ -34,6 +34,7 @@ public: int uid, IAudioFlinger::track_flags_t flags); virtual ~Track(); + virtual status_t initCheck() const; static void appendDumpHeader(String8& result); void dump(char* buffer, size_t size); diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h index 57de568..5ef6f58 100644 --- a/services/audioflinger/RecordTracks.h +++ b/services/audioflinger/RecordTracks.h @@ -59,5 +59,4 @@ private: // releaseBuffer() not overridden bool mOverflow; // overflow on most recent attempt to fill client buffer - AudioRecordServerProxy* mAudioRecordServerProxy; }; diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index b8c1632..b893625 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -269,8 +269,8 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio : Thread(false /*canCallJava*/), mType(type), mAudioFlinger(audioFlinger), - // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are - // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() + // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize + // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() mParamStatus(NO_ERROR), //FIXME: mStandby should be true here. Is this some kind of hack? mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), @@ -297,6 +297,17 @@ AudioFlinger::ThreadBase::~ThreadBase() } } +status_t AudioFlinger::ThreadBase::readyToRun() +{ + status_t status = initCheck(); + if (status == NO_ERROR) { + ALOGI("AudioFlinger's thread %p ready to run", this); + } else { + ALOGE("No working audio driver found."); + } + return status; +} + void AudioFlinger::ThreadBase::exit() { ALOGV("ThreadBase::exit"); @@ -369,7 +380,13 @@ void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32 void AudioFlinger::ThreadBase::processConfigEvents() { - mLock.lock(); + Mutex::Autolock _l(mLock); + processConfigEvents_l(); +} + +// post condition: mConfigEvents.isEmpty() +void AudioFlinger::ThreadBase::processConfigEvents_l() +{ while (!mConfigEvents.isEmpty()) { ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); ConfigEvent *event = mConfigEvents[0]; @@ -377,32 +394,31 @@ void AudioFlinger::ThreadBase::processConfigEvents() // release mLock before locking AudioFlinger mLock: lock order is always // AudioFlinger then ThreadBase to avoid cross deadlock mLock.unlock(); - switch(event->type()) { - case CFG_EVENT_PRIO: { - PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); - // FIXME Need to understand why this has be done asynchronously - int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), - true /*asynchronous*/); - if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " - "error %d", - prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); - } - } break; - case CFG_EVENT_IO: { - IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); - mAudioFlinger->mLock.lock(); + switch (event->type()) { + case CFG_EVENT_PRIO: { + PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); + // FIXME Need to understand why this has be done asynchronously + int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), + true /*asynchronous*/); + if (err != 0) { + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); + } + } break; + case CFG_EVENT_IO: { + IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); + { + Mutex::Autolock _l(mAudioFlinger->mLock); audioConfigChanged_l(ioEvent->event(), ioEvent->param()); - mAudioFlinger->mLock.unlock(); - } break; - default: - ALOGE("processConfigEvents() unknown event type %d", event->type()); - break; + } + } break; + default: + ALOGE("processConfigEvents() unknown event type %d", event->type()); + break; } delete event; mLock.lock(); } - mLock.unlock(); } void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) @@ -427,6 +443,8 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) result.append(buffer); snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); result.append(buffer); + snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); + result.append(buffer); snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); @@ -739,8 +757,7 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( int sessionId, effect_descriptor_t *desc, int *enabled, - status_t *status - ) + status_t *status) { sp<EffectModule> effect; sp<EffectHandle> handle; @@ -850,9 +867,7 @@ Exit: handle.clear(); } - if (status != NULL) { - *status = lStatus; - } + *status = lStatus; return handle; } @@ -1002,7 +1017,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge type_t type) : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), mNormalFrameCount(0), mMixBuffer(NULL), - mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), + mSuspended(0), mBytesWritten(0), mActiveTracksGeneration(0), // mStreamTypes[] initialized in constructor body mOutput(output), @@ -1060,7 +1075,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge AudioFlinger::PlaybackThread::~PlaybackThread() { mAudioFlinger->unregisterWriter(mNBLogWriter); - delete [] mAllocMixBuffer; + delete[] mMixBuffer; } void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) @@ -1150,16 +1165,6 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& } // Thread virtuals -status_t AudioFlinger::PlaybackThread::readyToRun() -{ - status_t status = initCheck(); - if (status == NO_ERROR) { - ALOGI("AudioFlinger's thread %p ready to run", this); - } else { - ALOGE("No working audio driver found."); - } - return status; -} void AudioFlinger::PlaybackThread::onFirstRef() { @@ -1326,8 +1331,12 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac track = TimedTrack::create(this, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId, uid); } - if (track == 0 || track->getCblk() == NULL || track->name() < 0) { - lStatus = NO_MEMORY; + + // new Track always returns non-NULL, + // but TimedTrack::create() is a factory that could fail by returning NULL + lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; + if (lStatus != NO_ERROR) { + track.clear(); goto Exit; } @@ -1352,9 +1361,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return track; } @@ -1642,7 +1649,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mFormat); } mFrameSize = audio_stream_frame_size(&mOutput->stream->common); - mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; + mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); + mFrameCount = mBufferSize / mFrameSize; if (mFrameCount & 15) { ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", mFrameCount); @@ -1699,11 +1707,11 @@ void AudioFlinger::PlaybackThread::readOutputParameters() ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); - delete[] mAllocMixBuffer; - size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; - mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; - mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); - memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); + delete[] mMixBuffer; + size_t normalBufferSize = mNormalFrameCount * mFrameSize; + // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) + mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; + memset(mMixBuffer, 0, normalBufferSize); // force reconfiguration of effect chains and engines to take new buffer size and audio // parameters into account @@ -1837,7 +1845,7 @@ void AudioFlinger::PlaybackThread::threadLoop_removeTracks( const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); - if (count) { + if (count > 0) { for (size_t i = 0 ; i < count ; i++) { const sp<Track>& track = tracksToRemove.itemAt(i); if (!track->isOutputTrack()) { @@ -2405,7 +2413,7 @@ if (mType == MIXER) { void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); - if (count) { + if (count > 0) { for (size_t i=0 ; i<count ; i++) { const sp<Track>& track = tracksToRemove.itemAt(i); mActiveTracks.remove(track); @@ -2798,7 +2806,7 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime() sleepTime = idleSleepTime; } } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { - memset (mMixBuffer, 0, mixBufferSize); + memset(mMixBuffer, 0, mixBufferSize); sleepTime = 0; ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), "anticipated start"); @@ -3024,7 +3032,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // +1 for rounding and +1 for additional sample needed for interpolation desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; // add frames already consumed but not yet released by the resampler - // because cblk->framesReady() will include these frames + // because mAudioTrackServerProxy->framesReady() will include these frames desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); // 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 @@ -3362,6 +3370,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { status = BAD_VALUE; } else { + // no need to save value, since it's constant reconfig = true; } } @@ -3369,6 +3378,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { status = BAD_VALUE; } else { + // no need to save value, since it's constant reconfig = true; } } @@ -4156,15 +4166,15 @@ bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() // must be called with thread mutex locked bool AudioFlinger::OffloadThread::shouldStandby_l() { - bool TrackPaused = false; + bool trackPaused = false; // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack // after a timeout and we will enter standby then. if (mTracks.size() > 0) { - TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); + trackPaused = mTracks[mTracks.size() - 1]->isPaused(); } - return !mStandby && !TrackPaused; + return !mStandby && !trackPaused; } @@ -4372,7 +4382,9 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, ) : ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), - // mRsmpInIndex and mBufferSize set by readInputParameters() + // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear + // are set by readInputParameters() + // mRsmpInIndex LEGACY mReqChannelCount(popcount(channelMask)), mReqSampleRate(sampleRate) // mBytesRead is only meaningful while active, and so is cleared in start() @@ -4399,22 +4411,12 @@ void AudioFlinger::RecordThread::onFirstRef() run(mName, PRIORITY_URGENT_AUDIO); } -status_t AudioFlinger::RecordThread::readyToRun() -{ - status_t status = initCheck(); - ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); - return status; -} - bool AudioFlinger::RecordThread::threadLoop() { - AudioBufferProvider::Buffer buffer; - sp<RecordTrack> activeTrack; - Vector< sp<EffectChain> > effectChains; - nsecs_t lastWarning = 0; inputStandBy(); + sp<RecordTrack> activeTrack; { Mutex::Autolock _l(mLock); activeTrack = mActiveTrack; @@ -4424,27 +4426,38 @@ bool AudioFlinger::RecordThread::threadLoop() // used to verify we've read at least once before evaluating how many bytes were read bool readOnce = false; + // used to request a deferred sleep, to be executed later while mutex is unlocked + bool doSleep = false; + // start recording - while (!exitPending()) { + for (;;) { + TrackBase::track_state activeTrackState; + Vector< sp<EffectChain> > effectChains; - processConfigEvents(); + // sleep with mutex unlocked + if (doSleep) { + doSleep = false; + usleep(kRecordThreadSleepUs); + } { // scope for mLock Mutex::Autolock _l(mLock); - checkForNewParameters_l(); + if (exitPending()) { + break; + } + processConfigEvents_l(); + // return value 'reconfig' is currently unused + bool reconfig = checkForNewParameters_l(); if (mActiveTrack != 0 && activeTrack != mActiveTrack) { SortedVector<int> tmp; tmp.add(mActiveTrack->uid()); updateWakeLockUids_l(tmp); } + // make a stable copy of mActiveTrack activeTrack = mActiveTrack; - if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { - standby(); - - if (exitPending()) { - break; - } - + if (activeTrack == 0) { + standbyIfNotAlreadyInStandby(); + // exitPending() can't become true here releaseWakeLock_l(); ALOGV("RecordThread: loop stopping"); // go to sleep @@ -4453,176 +4466,238 @@ bool AudioFlinger::RecordThread::threadLoop() acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); continue; } - if (mActiveTrack != 0) { - if (mActiveTrack->isTerminated()) { - removeTrack_l(mActiveTrack); - mActiveTrack.clear(); - } else if (mActiveTrack->mState == TrackBase::PAUSING) { - standby(); + + if (activeTrack->isTerminated()) { + removeTrack_l(activeTrack); + mActiveTrack.clear(); + continue; + } + + activeTrackState = activeTrack->mState; + switch (activeTrackState) { + case TrackBase::PAUSING: + standbyIfNotAlreadyInStandby(); + mActiveTrack.clear(); + mStartStopCond.broadcast(); + doSleep = true; + continue; + + case TrackBase::RESUMING: + mStandby = false; + if (mReqChannelCount != activeTrack->channelCount()) { mActiveTrack.clear(); mStartStopCond.broadcast(); - } else if (mActiveTrack->mState == TrackBase::RESUMING) { - if (mReqChannelCount != mActiveTrack->channelCount()) { + continue; + } + if (readOnce) { + mStartStopCond.broadcast(); + // record start succeeds only if first read from audio input succeeds + if (mBytesRead < 0) { mActiveTrack.clear(); - mStartStopCond.broadcast(); - } else if (readOnce) { - // record start succeeds only if first read from audio input - // succeeds - if (mBytesRead >= 0) { - mActiveTrack->mState = TrackBase::ACTIVE; - } else { - mActiveTrack.clear(); - } - mStartStopCond.broadcast(); + continue; } - mStandby = false; + activeTrack->mState = TrackBase::ACTIVE; } + break; + + case TrackBase::ACTIVE: + break; + + case TrackBase::IDLE: + doSleep = true; + continue; + + default: + LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); } lockEffectChains_l(effectChains); } - if (mActiveTrack != 0) { - if (mActiveTrack->mState != TrackBase::ACTIVE && - mActiveTrack->mState != TrackBase::RESUMING) { - unlockEffectChains(effectChains); - usleep(kRecordThreadSleepUs); - continue; - } - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } + // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable + // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING - buffer.frameCount = mFrameCount; - status_t status = mActiveTrack->getNextBuffer(&buffer); - if (status == NO_ERROR) { - readOnce = true; - size_t framesOut = buffer.frameCount; - if (mResampler == NULL) { - // no resampling - while (framesOut) { - size_t framesIn = mFrameCount - mRsmpInIndex; - if (framesIn) { - int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; - int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * - mActiveTrack->mFrameSize; - if (framesIn > framesOut) - framesIn = framesOut; - mRsmpInIndex += framesIn; - framesOut -= framesIn; - if (mChannelCount == mReqChannelCount) { - memcpy(dst, src, framesIn * mFrameSize); - } else { - if (mChannelCount == 1) { - upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } else { - downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } - } + for (size_t i = 0; i < effectChains.size(); i ++) { + // thread mutex is not locked, but effect chain is locked + effectChains[i]->process_l(); + } + + AudioBufferProvider::Buffer buffer; + buffer.frameCount = mFrameCount; + status_t status = activeTrack->getNextBuffer(&buffer); + if (status == NO_ERROR) { + readOnce = true; + size_t framesOut = buffer.frameCount; + if (mResampler == NULL) { + // no resampling + while (framesOut) { + size_t framesIn = mFrameCount - mRsmpInIndex; + if (framesIn > 0) { + int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; + int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * + activeTrack->mFrameSize; + if (framesIn > framesOut) { + framesIn = framesOut; } - if (framesOut && mFrameCount == mRsmpInIndex) { - void *readInto; - if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { - readInto = buffer.raw; - framesOut = 0; + mRsmpInIndex += framesIn; + framesOut -= framesIn; + if (mChannelCount == mReqChannelCount) { + memcpy(dst, src, framesIn * mFrameSize); + } else { + if (mChannelCount == 1) { + upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, + (int16_t *)src, framesIn); } else { - readInto = mRsmpInBuffer; - mRsmpInIndex = 0; + downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, + (int16_t *)src, framesIn); } - mBytesRead = mInput->stream->read(mInput->stream, readInto, - mBufferSize); - if (mBytesRead <= 0) { - if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) - { - ALOGE("Error reading audio input"); - // Force input into standby so that it tries to - // recover at next read attempt - inputStandBy(); - usleep(kRecordThreadSleepUs); - } - mRsmpInIndex = mFrameCount; - framesOut = 0; - buffer.frameCount = 0; + } + } + if (framesOut > 0 && mFrameCount == mRsmpInIndex) { + void *readInto; + if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { + readInto = buffer.raw; + framesOut = 0; + } else { + readInto = mRsmpInBuffer; + mRsmpInIndex = 0; + } + mBytesRead = mInput->stream->read(mInput->stream, readInto, + mBufferSize); + if (mBytesRead <= 0) { + // TODO: verify that it's benign to use a stale track state + if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) + { + ALOGE("Error reading audio input"); + // Force input into standby so that it tries to + // recover at next read attempt + inputStandBy(); + doSleep = true; } + mRsmpInIndex = mFrameCount; + framesOut = 0; + buffer.frameCount = 0; + } #ifdef TEE_SINK - else if (mTeeSink != 0) { - (void) mTeeSink->write(readInto, - mBytesRead >> Format_frameBitShift(mTeeSink->format())); - } -#endif + else if (mTeeSink != 0) { + (void) mTeeSink->write(readInto, + mBytesRead >> Format_frameBitShift(mTeeSink->format())); } +#endif } - } else { - // resampling - - // resampler accumulates, but we only have one source track - memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); - // alter output frame count as if we were expecting stereo samples - if (mChannelCount == 1 && mReqChannelCount == 1) { - framesOut >>= 1; + } + } else { + // resampling + + // avoid busy-waiting if client doesn't keep up + bool madeProgress = false; + + // keep mRsmpInBuffer full so resampler always has sufficient input + for (;;) { + int32_t rear = mRsmpInRear; + ssize_t filled = rear - mRsmpInFront; + ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); + // exit once there is enough data in buffer for resampler + if ((size_t) filled >= mRsmpInFrames) { + break; } - mResampler->resample(mRsmpOutBuffer, framesOut, - this /* AudioBufferProvider* */); - // ditherAndClamp() works as long as all buffers returned by - // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. - if (mChannelCount == 2 && mReqChannelCount == 1) { - // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t - ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: - // do post stereo to mono conversion - downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, - framesOut); - } else { - ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); + size_t avail = mRsmpInFramesP2 - filled; + // Only try to read full HAL buffers. + // But if the HAL read returns a partial buffer, use it. + if (avail < mFrameCount) { + ALOGE("insufficient space to read: avail %d < mFrameCount %d", + avail, mFrameCount); + break; + } + // If 'avail' is non-contiguous, first read past the nominal end of buffer, then + // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. + rear &= mRsmpInFramesP2 - 1; + mBytesRead = mInput->stream->read(mInput->stream, + &mRsmpInBuffer[rear * mChannelCount], mBufferSize); + if (mBytesRead <= 0) { + ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); + break; + } + ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); + size_t framesRead = mBytesRead / mFrameSize; + ALOG_ASSERT(framesRead > 0); + madeProgress = true; + // If 'avail' was non-contiguous, we now correct for reading past end of buffer. + size_t part1 = mRsmpInFramesP2 - rear; + if (framesRead > part1) { + memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], + (framesRead - part1) * mFrameSize); } - // now done with mRsmpOutBuffer + mRsmpInRear += framesRead; + } + if (!madeProgress) { + ALOGV("Did not make progress"); + usleep(((mFrameCount * 1000) / mSampleRate) * 1000); } - if (mFramestoDrop == 0) { - mActiveTrack->releaseBuffer(&buffer); + + // resampler accumulates, but we only have one source track + memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); + mResampler->resample(mRsmpOutBuffer, framesOut, + this /* AudioBufferProvider* */); + // ditherAndClamp() works as long as all buffers returned by + // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. + if (mReqChannelCount == 1) { + // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t + ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); + // the resampler always outputs stereo samples: + // do post stereo to mono conversion + downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, + framesOut); } else { - if (mFramestoDrop > 0) { - mFramestoDrop -= buffer.frameCount; - if (mFramestoDrop <= 0) { - clearSyncStartEvent(); - } - } else { - mFramestoDrop += buffer.frameCount; - if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || - mSyncStartEvent->isCancelled()) { - ALOGW("Synced record %s, session %d, trigger session %d", - (mFramestoDrop >= 0) ? "timed out" : "cancelled", - mActiveTrack->sessionId(), - (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); - clearSyncStartEvent(); - } - } + ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); } - mActiveTrack->clearOverflow(); + // now done with mRsmpOutBuffer + } - // client isn't retrieving buffers fast enough - else { - if (!mActiveTrack->setOverflow()) { - nsecs_t now = systemTime(); - if ((now - lastWarning) > kWarningThrottleNs) { - ALOGW("RecordThread: buffer overflow"); - lastWarning = now; + if (mFramestoDrop == 0) { + activeTrack->releaseBuffer(&buffer); + } else { + if (mFramestoDrop > 0) { + mFramestoDrop -= buffer.frameCount; + if (mFramestoDrop <= 0) { + clearSyncStartEvent(); } + } else { + mFramestoDrop += buffer.frameCount; + if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || + mSyncStartEvent->isCancelled()) { + ALOGW("Synced record %s, session %d, trigger session %d", + (mFramestoDrop >= 0) ? "timed out" : "cancelled", + activeTrack->sessionId(), + (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); + clearSyncStartEvent(); + } + } + } + activeTrack->clearOverflow(); + } + // client isn't retrieving buffers fast enough + else { + if (!activeTrack->setOverflow()) { + nsecs_t now = systemTime(); + if ((now - lastWarning) > kWarningThrottleNs) { + ALOGW("RecordThread: buffer overflow"); + lastWarning = now; } - // Release the processor for a while before asking for a new buffer. - // This will give the application more chance to read from the buffer and - // clear the overflow. - usleep(kRecordThreadSleepUs); } + // Release the processor for a while before asking for a new buffer. + // This will give the application more chance to read from the buffer and + // clear the overflow. + doSleep = true; } + // enable changes in effect chain unlockEffectChains(effectChains); - effectChains.clear(); + // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end } - standby(); + standbyIfNotAlreadyInStandby(); { Mutex::Autolock _l(mLock); @@ -4640,7 +4715,7 @@ bool AudioFlinger::RecordThread::threadLoop() return false; } -void AudioFlinger::RecordThread::standby() +void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() { if (!mStandby) { inputStandBy(); @@ -4653,7 +4728,7 @@ void AudioFlinger::RecordThread::inputStandBy() mInput->stream->common.standby(&mInput->stream->common); } -sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( +sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, audio_format_t format, @@ -4732,9 +4807,9 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR track = new RecordTrack(this, client, sampleRate, format, channelMask, frameCount, sessionId, uid); - if (track->getCblk() == 0) { - ALOGE("createRecordTrack_l() no control block"); - lStatus = NO_MEMORY; + lStatus = track->initCheck(); + if (lStatus != NO_ERROR) { + ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); track.clear(); goto Exit; } @@ -4756,9 +4831,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return track; } @@ -4789,6 +4862,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac } { + // This section is a rendezvous between binder thread executing start() and RecordThread AutoMutex lock(mLock); if (mActiveTrack != 0) { if (recordTrack != mActiveTrack.get()) { @@ -4799,21 +4873,29 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac return status; } + // FIXME why? already set in constructor, 'STARTING_1' would be more accurate recordTrack->mState = TrackBase::IDLE; mActiveTrack = recordTrack; mLock.unlock(); status_t status = AudioSystem::startInput(mId); mLock.lock(); + // FIXME should verify that mActiveTrack is still == recordTrack if (status != NO_ERROR) { mActiveTrack.clear(); clearSyncStartEvent(); return status; } + // FIXME LEGACY mRsmpInIndex = mFrameCount; + mRsmpInFront = 0; + mRsmpInRear = 0; + mRsmpInUnrel = 0; mBytesRead = 0; if (mResampler != NULL) { mResampler->reset(); } + // FIXME hijacking a playback track state name which was intended for start after pause; + // here 'STARTING_2' would be more accurate mActiveTrack->mState = TrackBase::RESUMING; // signal thread to start ALOGV("Signal record thread"); @@ -4824,6 +4906,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac status = INVALID_OPERATION; goto startError; } + // FIXME incorrect usage of wait: no explicit predicate or loop mStartStopCond.wait(mLock); if (mActiveTrack == 0) { ALOGV("Record failed to start"); @@ -4874,11 +4957,13 @@ bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { return false; } + // note that threadLoop may still be processing the track at this point [without lock] recordTrack->mState = TrackBase::PAUSING; // do not wait for mStartStopCond if exiting if (exitPending()) { return true; } + // FIXME incorrect usage of wait: no explicit predicate or loop mStartStopCond.wait(mLock); // if we have been restarted, recordTrack == mActiveTrack.get() here if (exitPending() || recordTrack != mActiveTrack.get()) { @@ -5002,46 +5087,47 @@ void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args // AudioBufferProvider interface status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) { - size_t framesReq = buffer->frameCount; - size_t framesReady = mFrameCount - mRsmpInIndex; - int channelCount; - - if (framesReady == 0) { - mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); - if (mBytesRead <= 0) { - if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { - ALOGE("RecordThread::getNextBuffer() Error reading audio input"); - // Force input into standby so that it tries to - // recover at next read attempt - inputStandBy(); - usleep(kRecordThreadSleepUs); - } - buffer->raw = NULL; - buffer->frameCount = 0; - return NOT_ENOUGH_DATA; - } - mRsmpInIndex = 0; - framesReady = mFrameCount; - } - - if (framesReq > framesReady) { - framesReq = framesReady; - } - - if (mChannelCount == 1 && mReqChannelCount == 2) { - channelCount = 1; - } else { - channelCount = 2; - } - buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; - buffer->frameCount = framesReq; + int32_t rear = mRsmpInRear; + int32_t front = mRsmpInFront; + ssize_t filled = rear - front; + ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); + // 'filled' may be non-contiguous, so return only the first contiguous chunk + front &= mRsmpInFramesP2 - 1; + size_t part1 = mRsmpInFramesP2 - front; + if (part1 > (size_t) filled) { + part1 = filled; + } + size_t ask = buffer->frameCount; + ALOG_ASSERT(ask > 0); + if (part1 > ask) { + part1 = ask; + } + if (part1 == 0) { + // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty + ALOGE("RecordThread::getNextBuffer() starved"); + buffer->raw = NULL; + buffer->frameCount = 0; + mRsmpInUnrel = 0; + return NOT_ENOUGH_DATA; + } + + buffer->raw = mRsmpInBuffer + front * mChannelCount; + buffer->frameCount = part1; + mRsmpInUnrel = part1; return NO_ERROR; } // AudioBufferProvider interface void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) { - mRsmpInIndex += buffer->frameCount; + size_t stepCount = buffer->frameCount; + if (stepCount == 0) { + return; + } + ALOG_ASSERT(stepCount <= mRsmpInUnrel); + mRsmpInUnrel -= stepCount; + mRsmpInFront += stepCount; + buffer->raw = NULL; buffer->frameCount = 0; } @@ -5056,7 +5142,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() int value; audio_format_t reqFormat = mFormat; uint32_t reqSamplingRate = mReqSampleRate; - uint32_t reqChannelCount = mReqChannelCount; + audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { reqSamplingRate = value; @@ -5071,8 +5157,13 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() } } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - reqChannelCount = popcount(value); - reconfig = true; + audio_channel_mask_t mask = (audio_channel_mask_t) value; + if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { + status = BAD_VALUE; + } else { + reqChannelMask = mask; + reconfig = true; + } } if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { // do not accept frame count changes if tracks are open as the track buffer @@ -5121,6 +5212,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() } mAudioSource = (audio_source_t)value; } + if (status == NO_ERROR) { status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); @@ -5137,7 +5229,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() <= (2 * reqSamplingRate)) && popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && - (reqChannelCount <= FCC_2)) { + (reqChannelMask == AUDIO_CHANNEL_IN_MONO || + reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { status = NO_ERROR; } if (status == NO_ERROR) { @@ -5173,7 +5266,7 @@ String8 AudioFlinger::RecordThread::getParameters(const String8& keys) void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; - void *param2 = NULL; + const void *param2 = NULL; switch (event) { case AudioSystem::INPUT_OPENED: @@ -5212,29 +5305,22 @@ void AudioFlinger::RecordThread::readInputParameters() mFrameSize = audio_stream_frame_size(&mInput->stream->common); mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); mFrameCount = mBufferSize / mFrameSize; - mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; - - if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) - { - int channelCount; - // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid - // stereo to mono post process as the resampler always outputs stereo. - if (mChannelCount == 1 && mReqChannelCount == 2) { - channelCount = 1; - } else { - channelCount = 2; - } - mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); + // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to + // 1 full output buffer, regardless of the alignment of the available input. + mRsmpInFrames = mFrameCount * 3; + mRsmpInFramesP2 = roundup(mRsmpInFrames); + // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer + mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; + mRsmpInFront = 0; + mRsmpInRear = 0; + mRsmpInUnrel = 0; + + if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { + mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); mResampler->setSampleRate(mSampleRate); mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); + // resampler always outputs stereo mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; - - // optmization: if mono to mono, alter input frame count as if we were inputing - // stereo samples - if (mChannelCount == 1 && mReqChannelCount == 1) { - mFrameCount >>= 1; - } - } mRsmpInIndex = mFrameCount; } diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index a0b53cb..43e335d 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -36,6 +36,8 @@ public: audio_devices_t outDevice, audio_devices_t inDevice, type_t type); virtual ~ThreadBase(); + virtual status_t readyToRun(); + void dumpBase(int fd, const Vector<String16>& args); void dumpEffectChains(int fd, const Vector<String16>& args); @@ -141,6 +143,7 @@ public: void sendIoConfigEvent_l(int event, int param = 0); void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); void processConfigEvents(); + void processConfigEvents_l(); // see note at declaration of mStandby, mOutDevice and mInDevice bool standby() const { return mStandby; } @@ -156,7 +159,7 @@ public: int sessionId, effect_descriptor_t *desc, int *enabled, - status_t *status); + status_t *status /*non-NULL*/); void disconnectEffect(const sp< EffectModule>& effect, EffectHandle *handle, bool unpinIfLast); @@ -275,6 +278,7 @@ protected: uint32_t mChannelCount; size_t mFrameSize; audio_format_t mFormat; + size_t mBufferSize; // HAL buffer size for read() or write() // Parameter sequence by client: binder thread calling setParameters(): // 1. Lock mLock @@ -358,7 +362,6 @@ public: void dump(int fd, const Vector<String16>& args); // Thread virtuals - virtual status_t readyToRun(); virtual bool threadLoop(); // RefBase @@ -425,7 +428,7 @@ public: IAudioFlinger::track_flags_t *flags, pid_t tid, int uid, - status_t *status); + status_t *status /*non-NULL*/); AudioStreamOut* getOutput() const; AudioStreamOut* clearOutput(); @@ -479,7 +482,6 @@ protected: size_t mNormalFrameCount; // normal mixer and effects int16_t* mMixBuffer; // frame size aligned mix buffer - int8_t* mAllocMixBuffer; // mixer buffer allocation address // suspend count, > 0 means suspended. While suspended, the thread continues to pull from // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle @@ -867,12 +869,12 @@ public: // Thread virtuals virtual bool threadLoop(); - virtual status_t readyToRun(); // RefBase virtual void onFirstRef(); virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } + sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, @@ -883,7 +885,7 @@ public: int uid, IAudioFlinger::track_flags_t *flags, pid_t tid, - status_t *status); + status_t *status /*non-NULL*/); status_t start(RecordTrack* recordTrack, AudioSystem::sync_event_t event, @@ -926,13 +928,13 @@ public: bool hasFastRecorder() const { return false; } private: - void clearSyncStartEvent(); + void clearSyncStartEvent(); // Enter standby if not already in standby, and set mStandby flag - void standby(); + void standbyIfNotAlreadyInStandby(); // Call the HAL standby method unconditionally, and don't change mStandby flag - void inputStandBy(); + void inputStandBy(); AudioStreamIn *mInput; SortedVector < sp<RecordTrack> > mTracks; @@ -945,11 +947,22 @@ private: AudioResampler *mResampler; // interleaved stereo pairs of fixed-point signed Q19.12 int32_t *mRsmpOutBuffer; - int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount] - size_t mRsmpInIndex; - size_t mBufferSize; // stream buffer size for read() + + // resampler converts input at HAL Hz to output at AudioRecord client Hz + int16_t *mRsmpInBuffer; // see new[] for details on the size + size_t mRsmpInFrames; // size of resampler input in frames + size_t mRsmpInFramesP2;// size rounded up to a power-of-2 + size_t mRsmpInUnrel; // unreleased frames remaining from + // most recent getNextBuffer + // these are rolling counters that are never cleared + int32_t mRsmpInFront; // next available frame + int32_t mRsmpInRear; // last filled frame + 1 + size_t mRsmpInIndex; // FIXME legacy + + // client's requested configuration, which may differ from the HAL configuration const uint32_t mReqChannelCount; const uint32_t mReqSampleRate; + ssize_t mBytesRead; // sync event triggering actual audio capture. Frames read before this event will // be dropped and therefore not read by the application. diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h index cd201d9..05fde7c 100644 --- a/services/audioflinger/TrackBase.h +++ b/services/audioflinger/TrackBase.h @@ -48,6 +48,7 @@ public: int uid, bool isOut); virtual ~TrackBase(); + virtual status_t initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; } virtual status_t start(AudioSystem::sync_event_t event, int triggerSession) = 0; @@ -78,15 +79,6 @@ protected: virtual uint32_t sampleRate() const { return mSampleRate; } - // Return a pointer to the start of a contiguous slice of the track buffer. - // Parameter 'offset' is the requested start position, expressed in - // monotonically increasing frame units relative to the track epoch. - // Parameter 'frames' is the requested length, also in frame units. - // Always returns non-NULL. It is the caller's responsibility to - // verify that this will be successful; the result of calling this - // function with invalid 'offset' or 'frames' is undefined. - void* getBuffer(uint32_t offset, uint32_t frames) const; - bool isStopped() const { return (mState == STOPPED || mState == FLUSHED); } diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index af04ce7..d5178b1 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -396,6 +396,15 @@ AudioFlinger::PlaybackThread::Track::~Track() } } +status_t AudioFlinger::PlaybackThread::Track::initCheck() const +{ + status_t status = TrackBase::initCheck(); + if (status == NO_ERROR && mName < 0) { + status = NO_MEMORY; + } + return status; +} + void AudioFlinger::PlaybackThread::Track::destroy() { // NOTE: destroyTrack_l() can remove a strong reference to this Track @@ -1045,15 +1054,17 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, "AudioFlingerTimed"); - if (mTimedMemoryDealer == NULL) + if (mTimedMemoryDealer == NULL) { return NO_MEMORY; + } } sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); if (newBuffer == NULL) { newBuffer = mTimedMemoryDealer->allocate(size); - if (newBuffer == NULL) + if (newBuffer == NULL) { return NO_MEMORY; + } } *buffer = newBuffer; @@ -1764,9 +1775,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( { ALOGV("RecordTrack constructor"); if (mCblk != NULL) { - mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, - mFrameSize); - mServerProxy = mAudioRecordServerProxy; + mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); } } |