am 073dec19: am 22990fe1: Merge "Back to the old way of making sure that no more buffers are submitted" into klp-dev
* commit '073dec19db486290a02d9952e392df9cda5700ad':
Back to the old way of making sure that no more buffers are submitted
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
index 234aef2..f400732 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
@@ -316,6 +316,7 @@
if (-1 < fileDesc) {
if (FwdLockFile_attach(fileDesc) < 0) {
+ close(fileDesc);
return mimeString;
}
const char* pMimeType = FwdLockFile_GetContentType(fileDesc);
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 052064d..fb47448 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -60,7 +60,7 @@
size_t frameCount; // number of sample frames corresponding to size;
// on input it is the number of frames available,
// on output is the number of frames actually drained
- // (currently ignored, but will make the primary field in future)
+ // (currently ignored but will make the primary field in future)
size_t size; // input/output in bytes == frameCount * frameSize
// FIXME this is redundant with respect to frameCount,
@@ -446,7 +446,8 @@
// notification callback
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback
- bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+ bool mRefreshRemaining; // processAudioBuffer() should refresh
+ // mRemainingFrames and mRetryOnPartialBuffer
// These are private to processAudioBuffer(), and are not protected by a lock
uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 225ef76..b96b8a1 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -155,7 +155,8 @@
class OutputDescriptor {
public:
OutputDescriptor()
- : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {}
+ : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)
+ {}
uint32_t samplingRate;
audio_format_t format;
diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
index c29c7e5..99e9c3e 100644
--- a/include/media/AudioTimestamp.h
+++ b/include/media/AudioTimestamp.h
@@ -19,6 +19,8 @@
#include <time.h>
+namespace android {
+
class AudioTimestamp {
public:
AudioTimestamp() : mPosition(0) {
@@ -30,4 +32,6 @@
struct timespec mTime; // corresponding CLOCK_MONOTONIC when frame is expected to present
};
+} // namespace
+
#endif // ANDROID_AUDIO_TIMESTAMP_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index f379ee5..bec77ce 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -666,7 +666,6 @@
size_t mReqFrameCount; // frame count to request the next time a new
// IAudioTrack is needed
-
// constant after constructor or set()
audio_format_t mFormat; // as requested by client, not forced to 16-bit
audio_stream_type_t mStreamType;
@@ -705,7 +704,8 @@
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback,
// at initial source sample rate
- bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+ bool mRefreshRemaining; // processAudioBuffer() should refresh
+ // mRemainingFrames and mRetryOnPartialBuffer
// These are private to processAudioBuffer(), and are not protected by a lock
uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index e796ab3..2fcfc62 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -247,6 +247,8 @@
int32_t numChannels, int32_t sampleRate, int32_t bitRate,
int32_t aacProfile, bool isADTS);
+ status_t setupAC3Codec(bool encoder, int32_t numChannels, int32_t sampleRate);
+
status_t selectAudioPortFormat(
OMX_U32 portIndex, OMX_AUDIO_CODINGTYPE desiredFormat);
diff --git a/include/media/stagefright/DataSource.h b/include/media/stagefright/DataSource.h
index 742bc0e..157b1aa 100644
--- a/include/media/stagefright/DataSource.h
+++ b/include/media/stagefright/DataSource.h
@@ -80,7 +80,6 @@
const sp<DataSource> &source, String8 *mimeType,
float *confidence, sp<AMessage> *meta);
- static void RegisterSniffer(SnifferFunc func);
static void RegisterDefaultSniffers();
// for DRM
@@ -101,6 +100,9 @@
private:
static Mutex gSnifferMutex;
static List<SnifferFunc> gSniffers;
+ static bool gSniffersRegistered;
+
+ static void RegisterSniffer_l(SnifferFunc func);
DataSource(const DataSource &);
DataSource &operator=(const DataSource &);
diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h
index 85693d4..cf5beda 100644
--- a/include/media/stagefright/MediaDefs.h
+++ b/include/media/stagefright/MediaDefs.h
@@ -44,6 +44,7 @@
extern const char *MEDIA_MIMETYPE_AUDIO_FLAC;
extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS;
extern const char *MEDIA_MIMETYPE_AUDIO_MSGSM;
+extern const char *MEDIA_MIMETYPE_AUDIO_AC3;
extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;
extern const char *MEDIA_MIMETYPE_CONTAINER_WAV;
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index daaf20f..5121c17 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -248,6 +248,8 @@
int32_t numChannels, int32_t sampleRate, int32_t bitRate,
int32_t aacProfile, bool isADTS);
+ status_t setAC3Format(int32_t numChannels, int32_t sampleRate);
+
void setG711Format(int32_t numChannels);
status_t setVideoPortFormatType(
diff --git a/include/media/stagefright/SkipCutBuffer.h b/include/media/stagefright/SkipCutBuffer.h
index 2653b53..098aa69 100644
--- a/include/media/stagefright/SkipCutBuffer.h
+++ b/include/media/stagefright/SkipCutBuffer.h
@@ -47,6 +47,7 @@
private:
void write(const char *src, size_t num);
size_t read(char *dst, size_t num);
+ int32_t mSkip;
int32_t mFrontPadding;
int32_t mBackPadding;
int32_t mWriteHead;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 7fd9379..85862a8 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -48,7 +48,7 @@
#define CBLK_STREAM_END_DONE 0x400 // set by server on render completion, cleared by client
//EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation
-#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 //assuming upto a maximum of 20 seconds of offloaded
+#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 // assuming up to a maximum of 20 seconds of offloaded
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index b8a89a0..8319dcd 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -250,9 +250,6 @@
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
- if (channelMask == 0) {
- channelMask = AUDIO_CHANNEL_OUT_STEREO;
- }
// validate parameters
if (!audio_is_valid_format(format)) {
@@ -260,6 +257,11 @@
return BAD_VALUE;
}
+ if (!audio_is_output_channel(channelMask)) {
+ ALOGE("Invalid channel mask %#x", channelMask);
+ return BAD_VALUE;
+ }
+
// AudioFlinger does not currently support 8-bit data in shared memory
if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
ALOGE("8-bit data in shared memory is not supported");
@@ -282,10 +284,6 @@
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
- if (!audio_is_output_channel(channelMask)) {
- ALOGE("Invalid channel mask %#x", channelMask);
- return BAD_VALUE;
- }
mChannelMask = channelMask;
uint32_t channelCount = popcount(channelMask);
mChannelCount = channelCount;
@@ -445,8 +443,7 @@
void AudioTrack::stop()
{
AutoMutex lock(mLock);
- // FIXME pause then stop should not be a nop
- if (mState != STATE_ACTIVE) {
+ if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
return;
}
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index acfaea0..9df10f0 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -139,7 +139,7 @@
lStatus = reply.readInt32();
track = interface_cast<IAudioTrack>(reply.readStrongBinder());
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
return track;
@@ -198,7 +198,7 @@
}
}
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
return record;
@@ -415,15 +415,25 @@
audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
ALOGV("openOutput() returned output, %d", output);
devices = (audio_devices_t)reply.readInt32();
- if (pDevices != NULL) *pDevices = devices;
+ if (pDevices != NULL) {
+ *pDevices = devices;
+ }
samplingRate = reply.readInt32();
- if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = samplingRate;
+ }
format = (audio_format_t) reply.readInt32();
- if (pFormat != NULL) *pFormat = format;
+ if (pFormat != NULL) {
+ *pFormat = format;
+ }
channelMask = (audio_channel_mask_t)reply.readInt32();
- if (pChannelMask != NULL) *pChannelMask = channelMask;
+ if (pChannelMask != NULL) {
+ *pChannelMask = channelMask;
+ }
latency = reply.readInt32();
- if (pLatencyMs != NULL) *pLatencyMs = latency;
+ if (pLatencyMs != NULL) {
+ *pLatencyMs = latency;
+ }
return output;
}
@@ -487,13 +497,21 @@
remote()->transact(OPEN_INPUT, data, &reply);
audio_io_handle_t input = (audio_io_handle_t) reply.readInt32();
devices = (audio_devices_t)reply.readInt32();
- if (pDevices != NULL) *pDevices = devices;
+ if (pDevices != NULL) {
+ *pDevices = devices;
+ }
samplingRate = reply.readInt32();
- if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = samplingRate;
+ }
format = (audio_format_t) reply.readInt32();
- if (pFormat != NULL) *pFormat = format;
+ if (pFormat != NULL) {
+ *pFormat = format;
+ }
channelMask = (audio_channel_mask_t)reply.readInt32();
- if (pChannelMask != NULL) *pChannelMask = channelMask;
+ if (pChannelMask != NULL) {
+ *pChannelMask = channelMask;
+ }
return input;
}
@@ -535,11 +553,11 @@
status_t status = reply.readInt32();
if (status == NO_ERROR) {
uint32_t tmp = reply.readInt32();
- if (halFrames) {
+ if (halFrames != NULL) {
*halFrames = tmp;
}
tmp = reply.readInt32();
- if (dspFrames) {
+ if (dspFrames != NULL) {
*dspFrames = tmp;
}
}
@@ -657,7 +675,7 @@
if (pDesc == NULL) {
return effect;
- if (status) {
+ if (status != NULL) {
*status = BAD_VALUE;
}
}
@@ -675,7 +693,7 @@
} else {
lStatus = reply.readInt32();
int tmp = reply.readInt32();
- if (id) {
+ if (id != NULL) {
*id = tmp;
}
tmp = reply.readInt32();
@@ -685,7 +703,7 @@
effect = interface_cast<IEffect>(reply.readStrongBinder());
reply.read(pDesc, sizeof(effect_descriptor_t));
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index 22e9fad..b420c95 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -600,16 +600,15 @@
// wrong audio audio buffer size (mAudioBufferSize)
unsigned long toggle = mToggle ^ 1;
void *userData = (void *)((unsigned long)this | toggle);
- uint32_t channels = (numChannels == 2) ?
- AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO;
+ audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(numChannels);
// do not create a new audio track if current track is compatible with sample parameters
#ifdef USE_SHARED_MEM_BUFFER
newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
- channels, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
+ channelMask, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
#else
newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
- channels, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
+ channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
bufferFrames);
#endif
oldTrack = mAudioTrack;
diff --git a/media/libmediaplayerservice/HDCP.cpp b/media/libmediaplayerservice/HDCP.cpp
index c2ac1a3..afe3936 100644
--- a/media/libmediaplayerservice/HDCP.cpp
+++ b/media/libmediaplayerservice/HDCP.cpp
@@ -107,11 +107,7 @@
return NO_INIT;
}
- // TO-DO:
- // Only support HDCP_CAPS_ENCRYPT (byte-array to byte-array) for now.
- // use mHDCPModule->getCaps() when the HDCP libraries get updated.
- //return mHDCPModule->getCaps();
- return HDCPModule::HDCP_CAPS_ENCRYPT;
+ return mHDCPModule->getCaps();
}
status_t HDCP::encrypt(
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 528fdb9..6d2191d 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -35,7 +35,9 @@
#include <media/hardware/HardwareAPI.h>
+#include <OMX_AudioExt.h>
#include <OMX_Component.h>
+#include <OMX_IndexExt.h>
#include "include/avc_utils.h"
@@ -965,6 +967,10 @@
"audio_decoder.flac", "audio_encoder.flac" },
{ MEDIA_MIMETYPE_AUDIO_MSGSM,
"audio_decoder.gsm", "audio_encoder.gsm" },
+ { MEDIA_MIMETYPE_VIDEO_MPEG2,
+ "video_decoder.mpeg2", "video_encoder.mpeg2" },
+ { MEDIA_MIMETYPE_AUDIO_AC3,
+ "audio_decoder.ac3", "audio_encoder.ac3" },
};
static const size_t kNumMimeToRole =
@@ -1256,6 +1262,15 @@
} else {
err = setupRawAudioFormat(kPortIndexInput, sampleRate, numChannels);
}
+ } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AC3)) {
+ int32_t numChannels;
+ int32_t sampleRate;
+ if (!msg->findInt32("channel-count", &numChannels)
+ || !msg->findInt32("sample-rate", &sampleRate)) {
+ err = INVALID_OPERATION;
+ } else {
+ err = setupAC3Codec(encoder, numChannels, sampleRate);
+ }
}
if (err != OK) {
@@ -1452,6 +1467,44 @@
mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
}
+status_t ACodec::setupAC3Codec(
+ bool encoder, int32_t numChannels, int32_t sampleRate) {
+ status_t err = setupRawAudioFormat(
+ encoder ? kPortIndexInput : kPortIndexOutput, sampleRate, numChannels);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (encoder) {
+ ALOGW("AC3 encoding is not supported.");
+ return INVALID_OPERATION;
+ }
+
+ OMX_AUDIO_PARAM_ANDROID_AC3TYPE def;
+ InitOMXParams(&def);
+ def.nPortIndex = kPortIndexInput;
+
+ err = mOMX->getParameter(
+ mNode,
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+ &def,
+ sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ def.nChannels = numChannels;
+ def.nSampleRate = sampleRate;
+
+ return mOMX->setParameter(
+ mNode,
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+ &def,
+ sizeof(def));
+}
+
static OMX_AUDIO_AMRBANDMODETYPE pickModeFromBitRate(
bool isAMRWB, int32_t bps) {
if (isAMRWB) {
@@ -2546,7 +2599,7 @@
{
OMX_AUDIO_PORTDEFINITIONTYPE *audioDef = &def.format.audio;
- switch (audioDef->eEncoding) {
+ switch ((int)audioDef->eEncoding) {
case OMX_AUDIO_CodingPCM:
{
OMX_AUDIO_PARAM_PCMMODETYPE params;
@@ -2652,6 +2705,24 @@
break;
}
+ case OMX_AUDIO_CodingAndroidAC3:
+ {
+ OMX_AUDIO_PARAM_ANDROID_AC3TYPE params;
+ InitOMXParams(¶ms);
+ params.nPortIndex = kPortIndexOutput;
+
+ CHECK_EQ((status_t)OK, mOMX->getParameter(
+ mNode,
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+ ¶ms,
+ sizeof(params)));
+
+ notify->setString("mime", MEDIA_MIMETYPE_AUDIO_AC3);
+ notify->setInt32("channel-count", params.nChannels);
+ notify->setInt32("sample-rate", params.nSampleRate);
+ break;
+ }
+
default:
TRESPASS();
}
diff --git a/media/libstagefright/DataSource.cpp b/media/libstagefright/DataSource.cpp
index fc6fd9c..97987e2 100644
--- a/media/libstagefright/DataSource.cpp
+++ b/media/libstagefright/DataSource.cpp
@@ -107,6 +107,7 @@
Mutex DataSource::gSnifferMutex;
List<DataSource::SnifferFunc> DataSource::gSniffers;
+bool DataSource::gSniffersRegistered = false;
bool DataSource::sniff(
String8 *mimeType, float *confidence, sp<AMessage> *meta) {
@@ -114,7 +115,13 @@
*confidence = 0.0f;
meta->clear();
- Mutex::Autolock autoLock(gSnifferMutex);
+ {
+ Mutex::Autolock autoLock(gSnifferMutex);
+ if (!gSniffersRegistered) {
+ return false;
+ }
+ }
+
for (List<SnifferFunc>::iterator it = gSniffers.begin();
it != gSniffers.end(); ++it) {
String8 newMimeType;
@@ -133,9 +140,7 @@
}
// static
-void DataSource::RegisterSniffer(SnifferFunc func) {
- Mutex::Autolock autoLock(gSnifferMutex);
-
+void DataSource::RegisterSniffer_l(SnifferFunc func) {
for (List<SnifferFunc>::iterator it = gSniffers.begin();
it != gSniffers.end(); ++it) {
if (*it == func) {
@@ -148,23 +153,29 @@
// static
void DataSource::RegisterDefaultSniffers() {
- RegisterSniffer(SniffMPEG4);
- RegisterSniffer(SniffMatroska);
- RegisterSniffer(SniffOgg);
- RegisterSniffer(SniffWAV);
- RegisterSniffer(SniffFLAC);
- RegisterSniffer(SniffAMR);
- RegisterSniffer(SniffMPEG2TS);
- RegisterSniffer(SniffMP3);
- RegisterSniffer(SniffAAC);
- RegisterSniffer(SniffMPEG2PS);
- RegisterSniffer(SniffWVM);
+ Mutex::Autolock autoLock(gSnifferMutex);
+ if (gSniffersRegistered) {
+ return;
+ }
+
+ RegisterSniffer_l(SniffMPEG4);
+ RegisterSniffer_l(SniffMatroska);
+ RegisterSniffer_l(SniffOgg);
+ RegisterSniffer_l(SniffWAV);
+ RegisterSniffer_l(SniffFLAC);
+ RegisterSniffer_l(SniffAMR);
+ RegisterSniffer_l(SniffMPEG2TS);
+ RegisterSniffer_l(SniffMP3);
+ RegisterSniffer_l(SniffAAC);
+ RegisterSniffer_l(SniffMPEG2PS);
+ RegisterSniffer_l(SniffWVM);
char value[PROPERTY_VALUE_MAX];
if (property_get("drm.service.enabled", value, NULL)
&& (!strcmp(value, "1") || !strcasecmp(value, "true"))) {
- RegisterSniffer(SniffDRM);
+ RegisterSniffer_l(SniffDRM);
}
+ gSniffersRegistered = true;
}
// static
diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp
index b5d4e44..340cba7 100644
--- a/media/libstagefright/MediaDefs.cpp
+++ b/media/libstagefright/MediaDefs.cpp
@@ -42,6 +42,7 @@
const char *MEDIA_MIMETYPE_AUDIO_FLAC = "audio/flac";
const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS = "audio/aac-adts";
const char *MEDIA_MIMETYPE_AUDIO_MSGSM = "audio/gsm";
+const char *MEDIA_MIMETYPE_AUDIO_AC3 = "audio/ac3";
const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4";
const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav";
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 7f56af8..063ab49 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -40,7 +40,9 @@
#include <utils/Vector.h>
#include <OMX_Audio.h>
+#include <OMX_AudioExt.h>
#include <OMX_Component.h>
+#include <OMX_IndexExt.h>
#include "include/avc_utils.h"
@@ -533,6 +535,17 @@
sampleRate,
numChannels);
}
+ } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_AC3, mMIME)) {
+ int32_t numChannels;
+ int32_t sampleRate;
+ CHECK(meta->findInt32(kKeyChannelCount, &numChannels));
+ CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
+
+ status_t err = setAC3Format(numChannels, sampleRate);
+ if (err != OK) {
+ CODEC_LOGE("setAC3Format() failed (err = %d)", err);
+ return err;
+ }
} else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_ALAW, mMIME)
|| !strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_MLAW, mMIME)) {
// These are PCM-like formats with a fixed sample rate but
@@ -1400,6 +1413,10 @@
"audio_decoder.flac", "audio_encoder.flac" },
{ MEDIA_MIMETYPE_AUDIO_MSGSM,
"audio_decoder.gsm", "audio_encoder.gsm" },
+ { MEDIA_MIMETYPE_VIDEO_MPEG2,
+ "video_decoder.mpeg2", "video_encoder.mpeg2" },
+ { MEDIA_MIMETYPE_AUDIO_AC3,
+ "audio_decoder.ac3", "audio_encoder.ac3" },
};
static const size_t kNumMimeToRole =
@@ -3495,6 +3512,31 @@
return OK;
}
+status_t OMXCodec::setAC3Format(int32_t numChannels, int32_t sampleRate) {
+ OMX_AUDIO_PARAM_ANDROID_AC3TYPE def;
+ InitOMXParams(&def);
+ def.nPortIndex = kPortIndexInput;
+
+ status_t err = mOMX->getParameter(
+ mNode,
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+ &def,
+ sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ def.nChannels = numChannels;
+ def.nSampleRate = sampleRate;
+
+ return mOMX->setParameter(
+ mNode,
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+ &def,
+ sizeof(def));
+}
+
void OMXCodec::setG711Format(int32_t numChannels) {
CHECK(!mIsEncoder);
setRawAudioFormat(kPortIndexInput, 8000, numChannels);
@@ -4428,6 +4470,17 @@
mOutputFormat->setInt32(kKeyChannelCount, numChannels);
mOutputFormat->setInt32(kKeySampleRate, sampleRate);
mOutputFormat->setInt32(kKeyBitRate, bitRate);
+ } else if (audio_def->eEncoding ==
+ (OMX_AUDIO_CODINGTYPE)OMX_AUDIO_CodingAndroidAC3) {
+ mOutputFormat->setCString(
+ kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+ int32_t numChannels, sampleRate, bitRate;
+ inputFormat->findInt32(kKeyChannelCount, &numChannels);
+ inputFormat->findInt32(kKeySampleRate, &sampleRate);
+ inputFormat->findInt32(kKeyBitRate, &bitRate);
+ mOutputFormat->setInt32(kKeyChannelCount, numChannels);
+ mOutputFormat->setInt32(kKeySampleRate, sampleRate);
+ mOutputFormat->setInt32(kKeyBitRate, bitRate);
} else {
CHECK(!"Should not be here. Unknown audio encoding.");
}
diff --git a/media/libstagefright/SkipCutBuffer.cpp b/media/libstagefright/SkipCutBuffer.cpp
index 773854f..e2e6d79 100644
--- a/media/libstagefright/SkipCutBuffer.cpp
+++ b/media/libstagefright/SkipCutBuffer.cpp
@@ -25,7 +25,7 @@
namespace android {
SkipCutBuffer::SkipCutBuffer(int32_t skip, int32_t cut) {
- mFrontPadding = skip;
+ mFrontPadding = mSkip = skip;
mBackPadding = cut;
mWriteHead = 0;
mReadHead = 0;
@@ -94,6 +94,7 @@
void SkipCutBuffer::clear() {
mWriteHead = mReadHead = 0;
+ mFrontPadding = mSkip;
}
void SkipCutBuffer::write(const char *src, size_t num) {
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 1b20cbb..f842e27 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -58,6 +58,8 @@
mIsADTS(false),
mInputBufferCount(0),
mSignalledError(false),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mAnchorTimeUs(0),
mNumSamplesOutput(0),
mOutputPortSettingsChange(NONE) {
@@ -350,115 +352,83 @@
return;
}
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFlags = 0;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ if (inHeader) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- if (mDecoderHasData) {
- // flush out the decoder's delayed data by calling DecodeFrame
- // one more time, with the AACDEC_FLUSH flag set
- INT_PCM *outBuffer =
- reinterpret_cast<INT_PCM *>(
- outHeader->pBuffer + outHeader->nOffset);
+ if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumSamplesOutput = 0;
+ }
- AAC_DECODER_ERROR decoderErr =
- aacDecoder_DecodeFrame(mAACDecoder,
- outBuffer,
- outHeader->nAllocLen,
- AACDEC_FLUSH);
- mDecoderHasData = false;
+ if (mIsADTS) {
+ size_t adtsHeaderSize = 0;
+ // skip 30 bits, aac_frame_length follows.
+ // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
- if (decoderErr != AAC_DEC_OK) {
+ const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
+
+ bool signalError = false;
+ if (inHeader->nFilledLen < 7) {
+ ALOGE("Audio data too short to contain even the ADTS header. "
+ "Got %ld bytes.", inHeader->nFilledLen);
+ hexdump(adtsHeader, inHeader->nFilledLen);
+ signalError = true;
+ } else {
+ bool protectionAbsent = (adtsHeader[1] & 1);
+
+ unsigned aac_frame_length =
+ ((adtsHeader[3] & 3) << 11)
+ | (adtsHeader[4] << 3)
+ | (adtsHeader[5] >> 5);
+
+ if (inHeader->nFilledLen < aac_frame_length) {
+ ALOGE("Not enough audio data for the complete frame. "
+ "Got %ld bytes, frame size according to the ADTS "
+ "header is %u bytes.",
+ inHeader->nFilledLen, aac_frame_length);
+ hexdump(adtsHeader, inHeader->nFilledLen);
+ signalError = true;
+ } else {
+ adtsHeaderSize = (protectionAbsent ? 7 : 9);
+
+ inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
+ inBufferLength[0] = aac_frame_length - adtsHeaderSize;
+
+ inHeader->nOffset += adtsHeaderSize;
+ inHeader->nFilledLen -= adtsHeaderSize;
+ }
+ }
+
+ if (signalError) {
mSignalledError = true;
- notify(OMX_EventError, OMX_ErrorUndefined, decoderErr,
+ notify(OMX_EventError,
+ OMX_ErrorStreamCorrupt,
+ ERROR_MALFORMED,
NULL);
return;
}
-
- outHeader->nFilledLen =
- mStreamInfo->frameSize
- * sizeof(int16_t)
- * mStreamInfo->numChannels;
} else {
- // we never submitted any data to the decoder, so there's nothing to flush out
- outHeader->nFilledLen = 0;
- }
-
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
-
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
- }
-
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumSamplesOutput = 0;
- }
-
- size_t adtsHeaderSize = 0;
- if (mIsADTS) {
- // skip 30 bits, aac_frame_length follows.
- // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
-
- const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
-
- bool signalError = false;
- if (inHeader->nFilledLen < 7) {
- ALOGE("Audio data too short to contain even the ADTS header. "
- "Got %ld bytes.", inHeader->nFilledLen);
- hexdump(adtsHeader, inHeader->nFilledLen);
- signalError = true;
- } else {
- bool protectionAbsent = (adtsHeader[1] & 1);
-
- unsigned aac_frame_length =
- ((adtsHeader[3] & 3) << 11)
- | (adtsHeader[4] << 3)
- | (adtsHeader[5] >> 5);
-
- if (inHeader->nFilledLen < aac_frame_length) {
- ALOGE("Not enough audio data for the complete frame. "
- "Got %ld bytes, frame size according to the ADTS "
- "header is %u bytes.",
- inHeader->nFilledLen, aac_frame_length);
- hexdump(adtsHeader, inHeader->nFilledLen);
- signalError = true;
- } else {
- adtsHeaderSize = (protectionAbsent ? 7 : 9);
-
- inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
- inBufferLength[0] = aac_frame_length - adtsHeaderSize;
-
- inHeader->nOffset += adtsHeaderSize;
- inHeader->nFilledLen -= adtsHeaderSize;
- }
- }
-
- if (signalError) {
- mSignalledError = true;
-
- notify(OMX_EventError,
- OMX_ErrorStreamCorrupt,
- ERROR_MALFORMED,
- NULL);
-
- return;
+ inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
+ inBufferLength[0] = inHeader->nFilledLen;
}
} else {
- inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
- inBufferLength[0] = inHeader->nFilledLen;
+ inBufferLength[0] = 0;
}
// Fill and decode
@@ -471,50 +441,66 @@
int prevNumChannels = mStreamInfo->numChannels;
AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS;
- while (bytesValid[0] > 0 && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
+ while ((bytesValid[0] > 0 || mSawInputEos) && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
+ mDecoderHasData |= (bytesValid[0] > 0);
aacDecoder_Fill(mAACDecoder,
inBuffer,
inBufferLength,
bytesValid);
- mDecoderHasData = true;
decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
outBuffer,
outHeader->nAllocLen,
0 /* flags */);
-
if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
- ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+ if (mSawInputEos && bytesValid[0] <= 0) {
+ if (mDecoderHasData) {
+ // flush out the decoder's delayed data by calling DecodeFrame
+ // one more time, with the AACDEC_FLUSH flag set
+ decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
+ outBuffer,
+ outHeader->nAllocLen,
+ AACDEC_FLUSH);
+ mDecoderHasData = false;
+ }
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ break;
+ } else {
+ ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+ }
}
}
size_t numOutBytes =
mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
- if (decoderErr == AAC_DEC_OK) {
- UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
- inHeader->nFilledLen -= inBufferUsedLength;
- inHeader->nOffset += inBufferUsedLength;
- } else {
- ALOGW("AAC decoder returned error %d, substituting silence",
- decoderErr);
+ if (inHeader) {
+ if (decoderErr == AAC_DEC_OK) {
+ UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
+ inHeader->nFilledLen -= inBufferUsedLength;
+ inHeader->nOffset += inBufferUsedLength;
+ } else {
+ ALOGW("AAC decoder returned error %d, substituting silence",
+ decoderErr);
- memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
+ memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
- // Discard input buffer.
- inHeader->nFilledLen = 0;
+ // Discard input buffer.
+ inHeader->nFilledLen = 0;
- aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
+ aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
- // fall through
- }
+ // fall through
+ }
- if (inHeader->nFilledLen == 0) {
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
+ if (inHeader->nFilledLen == 0) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
}
/*
@@ -555,7 +541,6 @@
// we've previously decoded valid data, in the latter case
// (decode failed) we'll output a silent frame.
outHeader->nFilledLen = numOutBytes;
- outHeader->nFlags = 0;
outHeader->nTimeStamp =
mAnchorTimeUs
@@ -582,6 +567,12 @@
// depend on fragments from the last one decoded.
// drain all existing data
drainDecoder();
+ // force decoder loop to drop the first decoded buffer by resetting these state variables,
+ // but only if initialization has already happened.
+ if (mInputBufferCount != 0) {
+ mInputBufferCount = 1;
+ mStreamInfo->sampleRate = 0;
+ }
}
}
@@ -606,6 +597,8 @@
mStreamInfo->sampleRate = 0;
mSignalledError = false;
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
index 2d960ab..a7ea1e2 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -55,6 +55,8 @@
bool mDecoderHasData;
size_t mInputBufferCount;
bool mSignalledError;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
int64_t mAnchorTimeUs;
int64_t mNumSamplesOutput;
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index 7c382fb..877e3cb 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -49,6 +49,8 @@
mNumChannels(2),
mSamplingRate(44100),
mSignalledError(false),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mOutputPortSettingsChange(NONE) {
initPorts();
initDecoder();
@@ -194,48 +196,36 @@
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFlags = 0;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
-
- if (!mIsFirst) {
- // pad the end of the stream with 529 samples, since that many samples
- // were trimmed off the beginning when decoding started
- outHeader->nFilledLen =
- kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
-
- memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
- } else {
- // Since we never discarded frames from the start, we won't have
- // to add any padding at the end either.
- outHeader->nFilledLen = 0;
+ if (inHeader) {
+ if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumFramesOutput = 0;
}
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
+ mConfig->pInputBuffer =
+ inHeader->pBuffer + inHeader->nOffset;
+
+ mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
+ } else {
+ mConfig->pInputBuffer = NULL;
+ mConfig->inputBufferCurrentLength = 0;
}
-
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumFramesOutput = 0;
- }
-
- mConfig->pInputBuffer =
- inHeader->pBuffer + inHeader->nOffset;
-
- mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
mConfig->inputBufferMaxLength = 0;
mConfig->inputBufferUsedLength = 0;
@@ -262,13 +252,28 @@
mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t);
}
- // This is recoverable, just ignore the current frame and
- // play silence instead.
- memset(outHeader->pBuffer,
- 0,
- mConfig->outputFrameSize * sizeof(int16_t));
+ if (decoderErr == NO_ENOUGH_MAIN_DATA_ERROR && mSawInputEos) {
+ if (!mIsFirst) {
+ // pad the end of the stream with 529 samples, since that many samples
+ // were trimmed off the beginning when decoding started
+ outHeader->nOffset = 0;
+ outHeader->nFilledLen = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
- mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+ memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
+ }
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ } else {
+ // This is recoverable, just ignore the current frame and
+ // play silence instead.
+ memset(outHeader->pBuffer,
+ 0,
+ mConfig->outputFrameSize * sizeof(int16_t));
+
+ if (inHeader) {
+ mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+ }
+ }
} else if (mConfig->samplingRate != mSamplingRate
|| mConfig->num_channels != mNumChannels) {
mSamplingRate = mConfig->samplingRate;
@@ -289,7 +294,7 @@
outHeader->nFilledLen =
mConfig->outputFrameSize * sizeof(int16_t) - outHeader->nOffset;
- } else {
+ } else if (!mSignalledOutputEos) {
outHeader->nOffset = 0;
outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t);
}
@@ -298,23 +303,24 @@
mAnchorTimeUs
+ (mNumFramesOutput * 1000000ll) / mConfig->samplingRate;
- outHeader->nFlags = 0;
+ if (inHeader) {
+ CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
- CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
+ inHeader->nOffset += mConfig->inputBufferUsedLength;
+ inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
- inHeader->nOffset += mConfig->inputBufferUsedLength;
- inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
+
+ if (inHeader->nFilledLen == 0) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
+ }
mNumFramesOutput += mConfig->outputFrameSize / mNumChannels;
- if (inHeader->nFilledLen == 0) {
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
- }
-
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
outInfo = NULL;
@@ -362,6 +368,8 @@
pvmp3_InitDecoder(mConfig, mDecoderBuf);
mIsFirst = true;
mSignalledError = false;
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.h b/media/libstagefright/codecs/mp3dec/SoftMP3.h
index 4af91ea..f9e7b53 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.h
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.h
@@ -61,6 +61,8 @@
bool mIsFirst;
bool mSignalledError;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
enum {
NONE,
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
index 51bb958..515e4d3 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
@@ -54,6 +54,8 @@
mAnchorTimeUs(0),
mNumFramesOutput(0),
mNumFramesLeftOnPage(-1),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mOutputPortSettingsChange(NONE) {
initPorts();
CHECK_EQ(initDecoder(), (status_t)OK);
@@ -290,48 +292,47 @@
return;
}
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ int32_t numPageSamples = 0;
- outHeader->nFilledLen = 0;
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ if (inHeader) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
+ if (inHeader->nFilledLen || !mSawInputEos) {
+ CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
+ memcpy(&numPageSamples,
+ inHeader->pBuffer
+ + inHeader->nOffset + inHeader->nFilledLen - 4,
+ sizeof(numPageSamples));
+
+ if (inHeader->nOffset == 0) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumFramesOutput = 0;
+ }
+
+ inHeader->nFilledLen -= sizeof(numPageSamples);;
+ }
}
- int32_t numPageSamples;
- CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
- memcpy(&numPageSamples,
- inHeader->pBuffer
- + inHeader->nOffset + inHeader->nFilledLen - 4,
- sizeof(numPageSamples));
-
if (numPageSamples >= 0) {
mNumFramesLeftOnPage = numPageSamples;
}
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumFramesOutput = 0;
- }
-
- inHeader->nFilledLen -= sizeof(numPageSamples);;
-
ogg_buffer buf;
- buf.data = inHeader->pBuffer + inHeader->nOffset;
- buf.size = inHeader->nFilledLen;
+ buf.data = inHeader ? inHeader->pBuffer + inHeader->nOffset : NULL;
+ buf.size = inHeader ? inHeader->nFilledLen : 0;
buf.refcount = 1;
buf.ptr.owner = NULL;
@@ -351,6 +352,7 @@
int numFrames = 0;
+ outHeader->nFlags = 0;
int err = vorbis_dsp_synthesis(mState, &pack, 1);
if (err != 0) {
ALOGW("vorbis_dsp_synthesis returned %d", err);
@@ -370,13 +372,16 @@
ALOGV("discarding %d frames at end of page",
numFrames - mNumFramesLeftOnPage);
numFrames = mNumFramesLeftOnPage;
+ if (mSawInputEos) {
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ }
}
mNumFramesLeftOnPage -= numFrames;
}
outHeader->nFilledLen = numFrames * sizeof(int16_t) * mVi->channels;
outHeader->nOffset = 0;
- outHeader->nFlags = 0;
outHeader->nTimeStamp =
mAnchorTimeUs
@@ -384,11 +389,13 @@
mNumFramesOutput += numFrames;
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
+ if (inHeader) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
@@ -425,6 +432,8 @@
mVi = NULL;
}
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
index cb628a0..1d00816 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
@@ -59,6 +59,8 @@
int64_t mAnchorTimeUs;
int64_t mNumFramesOutput;
int32_t mNumFramesLeftOnPage;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
enum {
NONE,
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 9850a46..f87b9da 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -506,6 +506,11 @@
ElementaryStreamQueue::PCM_AUDIO);
break;
+ case STREAMTYPE_AC3:
+ mQueue = new ElementaryStreamQueue(
+ ElementaryStreamQueue::AC3);
+ break;
+
default:
break;
}
@@ -614,6 +619,7 @@
case STREAMTYPE_MPEG2_AUDIO:
case STREAMTYPE_MPEG2_AUDIO_ADTS:
case STREAMTYPE_PCM_AUDIO:
+ case STREAMTYPE_AC3:
return true;
default:
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index a10edc9..d4e30b4 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -88,6 +88,10 @@
STREAMTYPE_MPEG2_AUDIO_ADTS = 0x0f,
STREAMTYPE_MPEG4_VIDEO = 0x10,
STREAMTYPE_H264 = 0x1b,
+
+ // From ATSC A/53 Part 3:2009, 6.7.1
+ STREAMTYPE_AC3 = 0x81,
+
STREAMTYPE_PCM_AUDIO = 0x83,
};
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index 8f9c9c8..ea79885 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -56,6 +56,122 @@
}
}
+// Parse AC3 header assuming the current ptr is start position of syncframe,
+// update metadata only applicable, and return the payload size
+static unsigned parseAC3SyncFrame(
+ const uint8_t *ptr, size_t size, sp<MetaData> *metaData) {
+ static const unsigned channelCountTable[] = {2, 1, 2, 3, 4, 4, 5, 6};
+ static const unsigned samplingRateTable[] = {48000, 44100, 32000};
+ static const unsigned rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256,
+ 320, 384, 448, 512, 576, 640};
+
+ static const unsigned frameSizeTable[19][3] = {
+ { 64, 69, 96 },
+ { 80, 87, 120 },
+ { 96, 104, 144 },
+ { 112, 121, 168 },
+ { 128, 139, 192 },
+ { 160, 174, 240 },
+ { 192, 208, 288 },
+ { 224, 243, 336 },
+ { 256, 278, 384 },
+ { 320, 348, 480 },
+ { 384, 417, 576 },
+ { 448, 487, 672 },
+ { 512, 557, 768 },
+ { 640, 696, 960 },
+ { 768, 835, 1152 },
+ { 896, 975, 1344 },
+ { 1024, 1114, 1536 },
+ { 1152, 1253, 1728 },
+ { 1280, 1393, 1920 },
+ };
+
+ ABitReader bits(ptr, size);
+ unsigned syncStartPos = 0; // in bytes
+ if (bits.numBitsLeft() < 16) {
+ return 0;
+ }
+ if (bits.getBits(16) != 0x0B77) {
+ return 0;
+ }
+
+ if (bits.numBitsLeft() < 16 + 2 + 6 + 5 + 3 + 3) {
+ ALOGV("Not enough bits left for further parsing");
+ return 0;
+ }
+ bits.skipBits(16); // crc1
+
+ unsigned fscod = bits.getBits(2);
+ if (fscod == 3) {
+ ALOGW("Incorrect fscod in AC3 header");
+ return 0;
+ }
+
+ unsigned frmsizecod = bits.getBits(6);
+ if (frmsizecod > 37) {
+ ALOGW("Incorrect frmsizecod in AC3 header");
+ return 0;
+ }
+
+ unsigned bsid = bits.getBits(5);
+ if (bsid > 8) {
+ ALOGW("Incorrect bsid in AC3 header. Possibly E-AC-3?");
+ return 0;
+ }
+
+ unsigned bsmod = bits.getBits(3);
+ unsigned acmod = bits.getBits(3);
+ unsigned cmixlev = 0;
+ unsigned surmixlev = 0;
+ unsigned dsurmod = 0;
+
+ if ((acmod & 1) > 0 && acmod != 1) {
+ if (bits.numBitsLeft() < 2) {
+ return 0;
+ }
+ cmixlev = bits.getBits(2);
+ }
+ if ((acmod & 4) > 0) {
+ if (bits.numBitsLeft() < 2) {
+ return 0;
+ }
+ surmixlev = bits.getBits(2);
+ }
+ if (acmod == 2) {
+ if (bits.numBitsLeft() < 2) {
+ return 0;
+ }
+ dsurmod = bits.getBits(2);
+ }
+
+ if (bits.numBitsLeft() < 1) {
+ return 0;
+ }
+ unsigned lfeon = bits.getBits(1);
+
+ unsigned samplingRate = samplingRateTable[fscod];
+ unsigned payloadSize = frameSizeTable[frmsizecod >> 1][fscod];
+ if (fscod == 1) {
+ payloadSize += frmsizecod & 1;
+ }
+ payloadSize <<= 1; // convert from 16-bit words to bytes
+
+ unsigned channelCount = channelCountTable[acmod] + lfeon;
+
+ if (metaData != NULL) {
+ (*metaData)->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+ (*metaData)->setInt32(kKeyChannelCount, channelCount);
+ (*metaData)->setInt32(kKeySampleRate, samplingRate);
+ }
+
+ return payloadSize;
+}
+
+static bool IsSeeminglyValidAC3Header(const uint8_t *ptr, size_t size) {
+ return parseAC3SyncFrame(ptr, size, NULL) > 0;
+}
+
static bool IsSeeminglyValidADTSHeader(const uint8_t *ptr, size_t size) {
if (size < 3) {
// Not enough data to verify header.
@@ -224,6 +340,33 @@
break;
}
+ case AC3:
+ {
+ uint8_t *ptr = (uint8_t *)data;
+
+ ssize_t startOffset = -1;
+ for (size_t i = 0; i < size; ++i) {
+ if (IsSeeminglyValidAC3Header(&ptr[i], size - i)) {
+ startOffset = i;
+ break;
+ }
+ }
+
+ if (startOffset < 0) {
+ return ERROR_MALFORMED;
+ }
+
+ if (startOffset > 0) {
+ ALOGI("found something resembling an AC3 syncword at "
+ "offset %d",
+ startOffset);
+ }
+
+ data = &ptr[startOffset];
+ size -= startOffset;
+ break;
+ }
+
case MPEG_AUDIO:
{
uint8_t *ptr = (uint8_t *)data;
@@ -328,6 +471,8 @@
return dequeueAccessUnitH264();
case AAC:
return dequeueAccessUnitAAC();
+ case AC3:
+ return dequeueAccessUnitAC3();
case MPEG_VIDEO:
return dequeueAccessUnitMPEGVideo();
case MPEG4_VIDEO:
@@ -340,6 +485,51 @@
}
}
+sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAC3() {
+ unsigned syncStartPos = 0; // in bytes
+ unsigned payloadSize = 0;
+ sp<MetaData> format = new MetaData;
+ while (true) {
+ if (syncStartPos + 2 >= mBuffer->size()) {
+ return NULL;
+ }
+
+ payloadSize = parseAC3SyncFrame(
+ mBuffer->data() + syncStartPos,
+ mBuffer->size() - syncStartPos,
+ &format);
+ if (payloadSize > 0) {
+ break;
+ }
+ ++syncStartPos;
+ }
+
+ if (mBuffer->size() < syncStartPos + payloadSize) {
+ ALOGV("Not enough buffer size for AC3");
+ return NULL;
+ }
+
+ if (mFormat == NULL) {
+ mFormat = format;
+ }
+
+ sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize);
+ memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize);
+
+ int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize);
+ CHECK_GE(timeUs, 0ll);
+ accessUnit->meta()->setInt64("timeUs", timeUs);
+
+ memmove(
+ mBuffer->data(),
+ mBuffer->data() + syncStartPos + payloadSize,
+ mBuffer->size() - syncStartPos - payloadSize);
+
+ mBuffer->setRange(0, mBuffer->size() - syncStartPos - payloadSize);
+
+ return accessUnit;
+}
+
sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitPCMAudio() {
if (mBuffer->size() < 4) {
return NULL;
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index 66a8087..a2cca77 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -32,6 +32,7 @@
enum Mode {
H264,
AAC,
+ AC3,
MPEG_AUDIO,
MPEG_VIDEO,
MPEG4_VIDEO,
@@ -67,6 +68,7 @@
sp<ABuffer> dequeueAccessUnitH264();
sp<ABuffer> dequeueAccessUnitAAC();
+ sp<ABuffer> dequeueAccessUnitAC3();
sp<ABuffer> dequeueAccessUnitMPEGAudio();
sp<ABuffer> dequeueAccessUnitMPEGVideo();
sp<ABuffer> dequeueAccessUnitMPEG4Video();
diff --git a/media/libstagefright/timedtext/test/Android.mk b/media/libstagefright/timedtext/test/Android.mk
index a5e7ba2..9a9fde2 100644
--- a/media/libstagefright/timedtext/test/Android.mk
+++ b/media/libstagefright/timedtext/test/Android.mk
@@ -2,7 +2,6 @@
# ================================================================
# Unit tests for libstagefright_timedtext
-# See also /development/testrunner/test_defs.xml
# ================================================================
# ================================================================
@@ -18,10 +17,13 @@
LOCAL_C_INCLUDES := \
$(TOP)/external/expat/lib \
- $(TOP)/frameworks/base/media/libstagefright/timedtext
+ $(TOP)/frameworks/av/media/libstagefright/timedtext
LOCAL_SHARED_LIBRARIES := \
+ libbinder \
libexpat \
- libstagefright
+ libstagefright \
+ libstagefright_foundation \
+ libutils
include $(BUILD_NATIVE_TEST)
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 @@
(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 @@
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 @@
}
}
}
+
}
+
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 @@
}
Exit:
- if (status != NULL) {
- *status = lStatus;
- }
+ *status = lStatus;
return trackHandle;
}
@@ -1293,6 +1298,7 @@
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 @@
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_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 @@
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 @@
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 @@
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_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 @@
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 @@
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 @@
// 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 @@
*pFormat = config.format;
}
if (pChannelMask != NULL) {
- *pChannelMask = reqChannels;
+ *pChannelMask = reqChannelMask;
}
// notify client processes of the new input creation
@@ -1954,7 +1958,7 @@
}
}
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 @@
}
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 @@
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 @@
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 @@
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:
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 @@
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 @@
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 @@
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 @@
// ----------------------------------------------------------------------------
-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 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 @@
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 @@
void AudioMixer::invalidateState(uint32_t mask)
{
- if (mask) {
+ if (mask != 0) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
@@ -276,13 +253,13 @@
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 @@
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 @@
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 @@
// 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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
{
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 @@
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 @@
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 @@
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 @@
}
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::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 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 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
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 @@
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 @@
// 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 ba8195e..6987dbd 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -269,8 +269,8 @@
: 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 @@
}
}
+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::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 @@
// 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 @@
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 @@
int sessionId,
effect_descriptor_t *desc,
int *enabled,
- status_t *status
- )
+ status_t *status)
{
sp<EffectModule> effect;
sp<EffectHandle> handle;
@@ -850,9 +867,7 @@
handle.clear();
}
- if (status != NULL) {
- *status = lStatus;
- }
+ *status = lStatus;
return handle;
}
@@ -1002,7 +1017,7 @@
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()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
- delete [] mAllocMixBuffer;
+ delete[] mMixBuffer;
}
void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
@@ -1150,16 +1165,6 @@
}
// 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 @@
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 @@
lStatus = NO_ERROR;
Exit:
- if (status) {
- *status = lStatus;
- }
+ *status = lStatus;
return track;
}
@@ -1642,7 +1649,8 @@
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 @@
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 @@
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 @@
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 @@
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 @@
// +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 @@
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 @@
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;
}
}
@@ -4152,15 +4162,15 @@
// 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;
}
@@ -4368,7 +4378,9 @@
) :
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()
@@ -4395,22 +4407,12 @@
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;
@@ -4420,27 +4422,38 @@
// used to verify we've read at least once before evaluating how many bytes were read
bool readOnce = false;
- // start recording
- while (!exitPending()) {
+ // used to request a deferred sleep, to be executed later while mutex is unlocked
+ bool doSleep = false;
- processConfigEvents();
+ // start recording
+ for (;;) {
+ TrackBase::track_state activeTrackState;
+ Vector< sp<EffectChain> > effectChains;
+
+ // 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
@@ -4449,176 +4462,238 @@
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()) {
- 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();
- }
- mStandby = false;
+ continue;
}
+ if (readOnce) {
+ mStartStopCond.broadcast();
+ // record start succeeds only if first read from audio input succeeds
+ if (mBytesRead < 0) {
+ mActiveTrack.clear();
+ continue;
+ }
+ 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);
- }
- }
- }
- if (framesOut && 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) {
- 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;
- }
-#ifdef TEE_SINK
- 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;
- }
- 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);
- }
- // now done with mRsmpOutBuffer
-
- }
- if (mFramestoDrop == 0) {
- mActiveTrack->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",
- mActiveTrack->sessionId(),
- (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
- clearSyncStartEvent();
- }
- }
- }
- mActiveTrack->clearOverflow();
- }
- // 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;
- }
- }
- // 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);
- }
+ 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;
+ }
+ 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);
+ }
+ }
+ }
+ 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 {
+ // 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;
+ }
+ 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);
+ }
+ mRsmpInRear += framesRead;
+ }
+
+ if (!madeProgress) {
+ ALOGV("Did not make progress");
+ usleep(((mFrameCount * 1000) / mSampleRate) * 1000);
+ }
+
+ // 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 {
+ ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ }
+ // now done with mRsmpOutBuffer
+
+ }
+ 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.
+ 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);
@@ -4636,7 +4711,7 @@
return false;
}
-void AudioFlinger::RecordThread::standby()
+void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
{
if (!mStandby) {
inputStandBy();
@@ -4649,7 +4724,7 @@
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,
@@ -4728,9 +4803,9 @@
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;
}
@@ -4752,9 +4827,7 @@
lStatus = NO_ERROR;
Exit:
- if (status) {
- *status = lStatus;
- }
+ *status = lStatus;
return track;
}
@@ -4785,6 +4858,7 @@
}
{
+ // This section is a rendezvous between binder thread executing start() and RecordThread
AutoMutex lock(mLock);
if (mActiveTrack != 0) {
if (recordTrack != mActiveTrack.get()) {
@@ -4795,21 +4869,29 @@
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");
@@ -4820,6 +4902,7 @@
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");
@@ -4870,11 +4953,13 @@
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()) {
@@ -4998,46 +5083,47 @@
// 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;
+ 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;
}
- if (framesReq > framesReady) {
- framesReq = framesReady;
- }
-
- if (mChannelCount == 1 && mReqChannelCount == 2) {
- channelCount = 1;
- } else {
- channelCount = 2;
- }
- buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
- buffer->frameCount = framesReq;
+ 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;
}
@@ -5052,7 +5138,7 @@
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;
@@ -5067,8 +5153,13 @@
}
}
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
@@ -5117,6 +5208,7 @@
}
mAudioSource = (audio_source_t)value;
}
+
if (status == NO_ERROR) {
status = mInput->stream->common.set_parameters(&mInput->stream->common,
keyValuePair.string());
@@ -5133,7 +5225,8 @@
<= (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) {
@@ -5169,7 +5262,7 @@
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:
@@ -5208,29 +5301,22 @@
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];
+ // 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)
- {
- 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);
+ 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 @@
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 @@
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 @@
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 @@
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 @@
void dump(int fd, const Vector<String16>& args);
// Thread virtuals
- virtual status_t readyToRun();
virtual bool threadLoop();
// RefBase
@@ -425,7 +428,7 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
}
+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 @@
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 @@
{
ALOGV("RecordTrack constructor");
if (mCblk != NULL) {
- mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
- mFrameSize);
- mServerProxy = mAudioRecordServerProxy;
+ mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
}
}