Merge "MediaCodec: change onError cb to return CodecException"
diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index c85368f..a795c80 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -53,6 +53,9 @@
// Convert a MIME type to a AudioSystem::audio_format
status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime);
+// Convert a aac profile to a AudioSystem::audio_format
+void mapAACProfileToAudioFormat(audio_format_t& format, uint64_t eAacProfile);
+
// Send information from MetaData to the HAL via AudioSink
status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink, const sp<MetaData>& meta);
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 99c8e9f..be9af5e 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -65,16 +65,15 @@
$(TOP)/frameworks/av/include/media/stagefright/timedtext \
$(TOP)/frameworks/native/include/media/hardware \
$(TOP)/frameworks/native/include/media/openmax \
- $(TOP)/frameworks/native/services/connectivitymanager \
$(TOP)/external/flac/include \
$(TOP)/external/tremolo \
$(TOP)/external/openssl/include \
$(TOP)/external/libvpx/libwebm \
+ $(TOP)/system/netd/include \
LOCAL_SHARED_LIBRARIES := \
libbinder \
libcamera_client \
- libconnectivitymanager \
libcutils \
libdl \
libdrmframework \
@@ -84,6 +83,7 @@
libicuuc \
liblog \
libmedia \
+ libnetd_client \
libopus \
libsonivox \
libssl \
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index fdac8fc..e24824b 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -23,6 +23,7 @@
#include <binder/IPCThreadState.h>
#include <media/AudioTrack.h>
+#include <media/openmax/OMX_Audio.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/AudioPlayer.h>
@@ -141,6 +142,12 @@
} else {
ALOGV("Mime type \"%s\" mapped to audio_format 0x%x", mime, audioFormat);
}
+
+ int32_t aacaot = -1;
+ if ((audioFormat == AUDIO_FORMAT_AAC) && format->findInt32(kKeyAACAOT, &aacaot)) {
+ // Redefine AAC format corrosponding to aac profile
+ mapAACProfileToAudioFormat(audioFormat,(OMX_AUDIO_AACPROFILETYPE) aacaot);
+ }
}
int avgBitRate = -1;
@@ -759,6 +766,11 @@
if (mSeeking) {
return mSeekTimeUs;
}
+ if (mReachedEOS) {
+ int64_t durationUs;
+ mSource->getFormat()->findInt64(kKeyDuration, &durationUs);
+ return durationUs;
+ }
mPositionTimeRealUs = getOutputPlayPositionUs_l();
ALOGV("getMediaTimeUs getOutputPlayPositionUs_l() mPositionTimeRealUs %" PRId64,
mPositionTimeRealUs);
diff --git a/media/libstagefright/HTTPBase.cpp b/media/libstagefright/HTTPBase.cpp
index ca68c3d..32291c8 100644
--- a/media/libstagefright/HTTPBase.cpp
+++ b/media/libstagefright/HTTPBase.cpp
@@ -26,7 +26,7 @@
#include <cutils/properties.h>
#include <cutils/qtaguid.h>
-#include <ConnectivityManager.h>
+#include <NetdClient.h>
namespace android {
@@ -122,7 +122,7 @@
// static
void HTTPBase::RegisterSocketUserMark(int sockfd, uid_t uid) {
- ConnectivityManager::markSocketAsUser(sockfd, uid);
+ setNetworkForUser(uid, sockfd);
}
// static
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 8c5deb3..750bff0 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -23,6 +23,7 @@
#include <arpa/inet.h>
#include <cutils/properties.h>
+#include <media/openmax/OMX_Audio.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -609,6 +610,39 @@
return BAD_VALUE;
}
+struct aac_format_conv_t {
+ OMX_AUDIO_AACPROFILETYPE eAacProfileType;
+ audio_format_t format;
+};
+
+static const struct aac_format_conv_t profileLookup[] = {
+ { OMX_AUDIO_AACObjectMain, AUDIO_FORMAT_AAC_MAIN},
+ { OMX_AUDIO_AACObjectLC, AUDIO_FORMAT_AAC_LC},
+ { OMX_AUDIO_AACObjectSSR, AUDIO_FORMAT_AAC_SSR},
+ { OMX_AUDIO_AACObjectLTP, AUDIO_FORMAT_AAC_LTP},
+ { OMX_AUDIO_AACObjectHE, AUDIO_FORMAT_AAC_HE_V1},
+ { OMX_AUDIO_AACObjectScalable, AUDIO_FORMAT_AAC_SCALABLE},
+ { OMX_AUDIO_AACObjectERLC, AUDIO_FORMAT_AAC_ERLC},
+ { OMX_AUDIO_AACObjectLD, AUDIO_FORMAT_AAC_LD},
+ { OMX_AUDIO_AACObjectHE_PS, AUDIO_FORMAT_AAC_HE_V2},
+ { OMX_AUDIO_AACObjectELD, AUDIO_FORMAT_AAC_ELD},
+ { OMX_AUDIO_AACObjectNull, AUDIO_FORMAT_AAC},
+};
+
+void mapAACProfileToAudioFormat( audio_format_t& format, uint64_t eAacProfile)
+{
+const struct aac_format_conv_t* p = &profileLookup[0];
+ while (p->eAacProfileType != OMX_AUDIO_AACObjectNull) {
+ if (eAacProfile == p->eAacProfileType) {
+ format = p->format;
+ return;
+ }
+ ++p;
+ }
+ format = AUDIO_FORMAT_AAC;
+ return;
+}
+
bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo,
bool isStreaming, audio_stream_type_t streamType)
{
@@ -634,15 +668,11 @@
return false;
}
- // check whether it is ELD/LD content -> no offloading
- // FIXME: this should depend on audio DSP capabilities. mapMimeToAudioFormat() should use the
- // metadata to refine the AAC format and the audio HAL should only list supported profiles.
+ // Redefine aac format according to its profile
+ // Offloading depends on audio DSP capabilities.
int32_t aacaot = -1;
if (meta->findInt32(kKeyAACAOT, &aacaot)) {
- if (aacaot == 23 || aacaot == 39 ) {
- ALOGV("track of type '%s' is ELD/LD content", mime);
- return false;
- }
+ mapAACProfileToAudioFormat(info.format,(OMX_AUDIO_AACPROFILETYPE) aacaot);
}
int32_t srate = -1;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2124f85..5fed0c1 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -106,13 +106,19 @@
// ----------------------------------------------------------------------------
const char *formatToString(audio_format_t format) {
- switch(format) {
- case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
- case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
- case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
- case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
- case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
- case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
+ switch (format & AUDIO_FORMAT_MAIN_MASK) {
+ case AUDIO_FORMAT_PCM:
+ switch (format) {
+ case AUDIO_FORMAT_PCM_16_BIT: return "pcm16";
+ case AUDIO_FORMAT_PCM_8_BIT: return "pcm8";
+ case AUDIO_FORMAT_PCM_32_BIT: return "pcm32";
+ case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24";
+ case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat";
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24";
+ default:
+ break;
+ }
+ break;
case AUDIO_FORMAT_MP3: return "mp3";
case AUDIO_FORMAT_AMR_NB: return "amr-nb";
case AUDIO_FORMAT_AMR_WB: return "amr-wb";
@@ -120,6 +126,9 @@
case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
case AUDIO_FORMAT_VORBIS: return "vorbis";
+ case AUDIO_FORMAT_OPUS: return "opus";
+ case AUDIO_FORMAT_AC3: return "ac-3";
+ case AUDIO_FORMAT_E_AC3: return "e-ac-3";
default:
break;
}
diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/AudioPolicyEffects.cpp
index 185e1cc..cc0e965 100755
--- a/services/audiopolicy/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/AudioPolicyEffects.cpp
@@ -15,7 +15,7 @@
*/
#define LOG_TAG "AudioPolicyEffects"
-#define LOG_NDEBUG 0
+//#define LOG_NDEBUG 0
#include <stdlib.h>
#include <stdio.h>
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index 73c9313..1b4796b 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -122,6 +122,16 @@
STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
STRING_TO_ENUM(AUDIO_FORMAT_MP3),
STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
@@ -2701,10 +2711,15 @@
ALOGW("checkOutputsForDevice() direct output missing param");
mpClientInterface->closeOutput(output);
output = 0;
- } else if (profile->mSamplingRates[0] == 0) {
+ } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 ||
+ profile->mChannelMasks[0] == 0) {
mpClientInterface->closeOutput(output);
- desc->mSamplingRate = profile->mSamplingRates[1];
+ desc->mSamplingRate = profile->pickSamplingRate();
+ desc->mFormat = profile->pickFormat();
+ desc->mChannelMask = profile->pickChannelMask();
offloadInfo.sample_rate = desc->mSamplingRate;
+ offloadInfo.format = desc->mFormat;
+ offloadInfo.channel_mask = desc->mChannelMask;
output = mpClientInterface->openOutput(
profile->mModule->mHandle,
&desc->mDevice,
@@ -4490,9 +4505,9 @@
}
if (profile != NULL) {
mAudioPort = profile;
- mSamplingRate = profile->mSamplingRates[0];
- mFormat = profile->mFormats[0];
- mChannelMask = profile->mChannelMasks[0];
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
if (profile->mGains.size() > 0) {
profile->mGains[0]->getDefaultConfig(&mGain);
}
@@ -4671,16 +4686,12 @@
{
if (profile != NULL) {
mAudioPort = profile;
- mSamplingRate = profile->mSamplingRates[0];
- mFormat = profile->mFormats[0];
- mChannelMask = profile->mChannelMasks[0];
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
if (profile->mGains.size() > 0) {
profile->mGains[0]->getDefaultConfig(&mGain);
}
- } else {
- mSamplingRate = 0;
- mFormat = AUDIO_FORMAT_DEFAULT;
- mChannelMask = 0;
}
}
@@ -4996,7 +5007,7 @@
AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type,
audio_port_role_t role, const sp<HwModule>& module) :
- mName(name), mType(type), mRole(role), mModule(module)
+ mName(name), mType(type), mRole(role), mModule(module), mFlags((audio_output_flags_t)0)
{
mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
@@ -5224,6 +5235,127 @@
return BAD_VALUE;
}
+
+uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const
+{
+ // special case for uninitialized dynamic profile
+ if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
+ return 0;
+ }
+
+ uint32_t samplingRate = 0;
+ uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
+
+ // For mixed output and inputs, use max mixer sampling rates. Do not
+ // limit sampling rate otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) ||
+ ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)))) {
+ maxRate = UINT_MAX;
+ }
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return samplingRate;
+}
+
+audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
+{
+ // special case for uninitialized dynamic profile
+ if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
+ return AUDIO_CHANNEL_NONE;
+ }
+
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+ uint32_t channelCount = 0;
+ uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+ // For mixed output and inputs, use max mixer channel count. Do not
+ // limit channel count otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) ||
+ ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)))) {
+ maxCount = UINT_MAX;
+ }
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+ channelMask = mChannelMasks[i];
+ }
+ }
+ return channelMask;
+}
+
+const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = {
+ AUDIO_FORMAT_DEFAULT,
+ AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_FORMAT_PCM_24_BIT_PACKED,
+};
+
+int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1,
+ audio_format_t format2)
+{
+ // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+ // compressed format and better than any PCM format. This is by design of pickFormat()
+ if (!audio_is_linear_pcm(format1)) {
+ if (!audio_is_linear_pcm(format2)) {
+ return 0;
+ }
+ return 1;
+ }
+ if (!audio_is_linear_pcm(format2)) {
+ return -1;
+ }
+
+ int index1 = -1, index2 = -1;
+ for (size_t i = 0;
+ (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+ i ++) {
+ if (sPcmFormatCompareTable[i] == format1) {
+ index1 = i;
+ }
+ if (sPcmFormatCompareTable[i] == format2) {
+ index2 = i;
+ }
+ }
+ // format1 not found => index1 < 0 => format2 > format1
+ // format2 not found => index2 < 0 => format2 < format1
+ return index1 - index2;
+}
+
+audio_format_t AudioPolicyManager::AudioPort::pickFormat() const
+{
+ // special case for uninitialized dynamic profile
+ if (mFormats.size() == 1 && mFormats[0] == 0) {
+ return AUDIO_FORMAT_DEFAULT;
+ }
+
+ audio_format_t format = AUDIO_FORMAT_DEFAULT;
+ audio_format_t bestFormat = BEST_MIXER_FORMAT;
+ // For mixed output and inputs, use best mixer output format. Do not
+ // limit format otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) ||
+ ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) == 0)))) {
+ bestFormat = AUDIO_FORMAT_INVALID;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if ((compareFormats(mFormats[i], format) > 0) &&
+ (compareFormats(mFormats[i], bestFormat) <= 0)) {
+ format = mFormats[i];
+ }
+ }
+ return format;
+}
+
status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig,
int index) const
{
@@ -5248,7 +5380,11 @@
snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
result.append(buffer);
for (size_t i = 0; i < mSamplingRates.size(); i++) {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ if (i == 0 && mSamplingRates[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ }
result.append(buffer);
result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
}
@@ -5259,7 +5395,13 @@
snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
result.append(buffer);
for (size_t i = 0; i < mChannelMasks.size(); i++) {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
+
+ if (i == 0 && mChannelMasks[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ }
result.append(buffer);
result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
}
@@ -5270,9 +5412,14 @@
snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
result.append(buffer);
for (size_t i = 0; i < mFormats.size(); i++) {
- snprintf(buffer, SIZE, "%-48s", enumToString(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- mFormats[i]));
+ const char *formatStr = enumToString(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ mFormats[i]);
+ if (i == 0 && strcmp(formatStr, "") == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%-48s", formatStr);
+ }
result.append(buffer);
result.append(i == (mFormats.size() - 1) ? "" : ", ");
}
@@ -5495,7 +5642,7 @@
AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
const sp<HwModule>& module)
- : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module), mFlags((audio_output_flags_t)0)
+ : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
{
}
@@ -5586,8 +5733,7 @@
audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
AUDIO_PORT_ROLE_SOURCE,
NULL),
- mDeviceType(type), mAddress(""),
- mChannelMask(AUDIO_CHANNEL_NONE), mId(0)
+ mDeviceType(type), mAddress(""), mId(0)
{
mAudioPort = this;
if (mGains.size() > 0) {
@@ -5807,10 +5953,6 @@
snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
result.append(buffer);
}
- if (mChannelMask != AUDIO_CHANNEL_NONE) {
- snprintf(buffer, SIZE, "%*s- channel mask: %08x\n", spaces, "", mChannelMask);
- result.append(buffer);
- }
write(fd, result.string(), result.size());
AudioPort::dump(fd, spaces);
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
index c23d994..4caecca 100644
--- a/services/audiopolicy/AudioPolicyManager.h
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -52,6 +52,12 @@
// Can be overridden by the audio.offload.min.duration.secs property
#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+#define MAX_MIXER_SAMPLING_RATE 48000
+#define MAX_MIXER_CHANNEL_COUNT 2
+// See AudioPort::compareFormats()
+#define WORST_MIXER_FORMAT AUDIO_FORMAT_PCM_16_BIT
+#define BEST_MIXER_FORMAT AUDIO_FORMAT_PCM_24_BIT_PACKED
+
// ----------------------------------------------------------------------------
// AudioPolicyManager implements audio policy manager behavior common to all platforms.
// ----------------------------------------------------------------------------
@@ -238,6 +244,13 @@
status_t checkFormat(audio_format_t format) const;
status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+ uint32_t pickSamplingRate() const;
+ audio_channel_mask_t pickChannelMask() const;
+ audio_format_t pickFormat() const;
+
+ static const audio_format_t sPcmFormatCompareTable[];
+ static int compareFormats(audio_format_t format1, audio_format_t format2);
+
void dump(int fd, int spaces) const;
String8 mName;
@@ -252,6 +265,8 @@
Vector <audio_format_t> mFormats; // supported audio formats
Vector < sp<AudioGain> > mGains; // gain controllers
sp<HwModule> mModule; // audio HW module exposing this I/O stream
+ audio_output_flags_t mFlags; // attribute flags (e.g primary output,
+ // direct output...). For outputs only.
};
class AudioPortConfig: public virtual RefBase
@@ -302,7 +317,6 @@
audio_devices_t mDeviceType;
String8 mAddress;
- audio_channel_mask_t mChannelMask;
audio_port_handle_t mId;
};
@@ -352,11 +366,10 @@
DeviceVector mSupportedDevices; // supported devices
// (devices this output can be routed to)
- audio_output_flags_t mFlags; // attribute flags (e.g primary output,
- // direct output...). For outputs only.
};
- class HwModule : public RefBase{
+ class HwModule : public RefBase
+ {
public:
HwModule(const char *name);
~HwModule();
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp
index 9435797..ae9cc35 100755
--- a/services/audiopolicy/AudioPolicyService.cpp
+++ b/services/audiopolicy/AudioPolicyService.cpp
@@ -841,8 +841,8 @@
}
// insert command at the right place according to its time stamp
- ALOGV("inserting command: %d at index %d, num commands %d",
- command->mCommand, (int)i+1, mAudioCommands.size());
+ ALOGV("inserting command: %d at index %zd, num commands %zu",
+ command->mCommand, i+1, mAudioCommands.size());
mAudioCommands.insertAt(command, i + 1);
}