Merge "audiopolicy: fix dynamic device & explicit routing management"
diff --git a/media/codec2/core/include/C2Config.h b/media/codec2/core/include/C2Config.h
index 0357115..fb6edb6 100644
--- a/media/codec2/core/include/C2Config.h
+++ b/media/codec2/core/include/C2Config.h
@@ -597,6 +597,9 @@
LEVEL_AVC_5, ///< AVC (H.264) Level 5
LEVEL_AVC_5_1, ///< AVC (H.264) Level 5.1
LEVEL_AVC_5_2, ///< AVC (H.264) Level 5.2
+ LEVEL_AVC_6, ///< AVC (H.264) Level 6
+ LEVEL_AVC_6_1, ///< AVC (H.264) Level 6.1
+ LEVEL_AVC_6_2, ///< AVC (H.264) Level 6.2
// HEVC (H.265) tiers and levels
LEVEL_HEVC_MAIN_1 = _C2_PL_HEVC_BASE, ///< HEVC (H.265) Main Tier Level 1
diff --git a/media/codec2/sfplugin/utils/Codec2Mapper.cpp b/media/codec2/sfplugin/utils/Codec2Mapper.cpp
index c369e16..0a6a717 100644
--- a/media/codec2/sfplugin/utils/Codec2Mapper.cpp
+++ b/media/codec2/sfplugin/utils/Codec2Mapper.cpp
@@ -65,7 +65,9 @@
{ C2Config::LEVEL_AVC_5, AVCLevel5 },
{ C2Config::LEVEL_AVC_5_1, AVCLevel51 },
{ C2Config::LEVEL_AVC_5_2, AVCLevel52 },
-
+ { C2Config::LEVEL_AVC_6, AVCLevel6 },
+ { C2Config::LEVEL_AVC_6_1, AVCLevel61 },
+ { C2Config::LEVEL_AVC_6_2, AVCLevel62 },
};
ALookup<C2Config::profile_t, int32_t> sAvcProfiles = {
diff --git a/media/codec2/vndk/C2Config.cpp b/media/codec2/vndk/C2Config.cpp
index 8a27088..34680a7 100644
--- a/media/codec2/vndk/C2Config.cpp
+++ b/media/codec2/vndk/C2Config.cpp
@@ -186,6 +186,9 @@
{ "avc-5", C2Config::LEVEL_AVC_5 },
{ "avc-5.1", C2Config::LEVEL_AVC_5_1 },
{ "avc-5.2", C2Config::LEVEL_AVC_5_2 },
+ { "avc-6", C2Config::LEVEL_AVC_6 },
+ { "avc-6.1", C2Config::LEVEL_AVC_6_1 },
+ { "avc-6.2", C2Config::LEVEL_AVC_6_2 },
{ "hevc-main-1", C2Config::LEVEL_HEVC_MAIN_1 },
{ "hevc-main-2", C2Config::LEVEL_HEVC_MAIN_2 },
{ "hevc-main-2.1", C2Config::LEVEL_HEVC_MAIN_2_1 },
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 75d425f..6578156 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -36,8 +36,11 @@
#include "LoopbackAnalyzer.h"
#include "../../utils/AAudioExampleUtils.h"
-// V0.4.00 = rectify and low-pass filter the echos, use auto-correlation on entire echo
-#define APP_VERSION "0.4.00"
+// V0.4.00 = rectify and low-pass filter the echos, auto-correlate entire echo
+// V0.4.01 = add -h hang option
+// fix -n option to set output buffer for -tm
+// plot first glitch
+#define APP_VERSION "0.4.01"
// Tag for machine readable results as property = value pairs
#define RESULT_TAG "RESULT: "
@@ -396,7 +399,7 @@
int32_t requestedInputCapacity = AAUDIO_UNSPECIFIED;
aaudio_performance_mode_t inputPerformanceLevel = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
- int32_t outputFramesPerBurst = 0;
+ int32_t outputFramesPerBurst = 0;
aaudio_format_t actualOutputFormat = AAUDIO_FORMAT_INVALID;
int32_t actualSampleRate = 0;
@@ -476,6 +479,8 @@
int32_t timeMillis = 0;
int32_t recordingDuration = std::min(60 * 5, requestedDuration);
+ int32_t requestedOutputBursts = argParser.getNumberOfBursts();
+
switch(testMode) {
case TEST_SINE_MAGNITUDE:
loopbackData.loopbackProcessor = &loopbackData.sineAnalyzer;
@@ -541,20 +546,19 @@
}
inputStream = loopbackData.inputStream = recorder.getStream();
- argParser.compareWithStream(inputStream);
-
{
int32_t actualCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
(void) AAudioStream_setBufferSizeInFrames(inputStream, actualCapacity);
- if (testMode == TEST_SINE_MAGNITUDE) {
+ if (testMode == TEST_SINE_MAGNITUDE
+ && requestedOutputBursts == AAUDIO_UNSPECIFIED) {
result = AAudioStream_setBufferSizeInFrames(outputStream, actualCapacity);
if (result < 0) {
fprintf(stderr, "ERROR - AAudioStream_setBufferSizeInFrames(output) returned %d\n",
result);
goto finish;
} else {
- printf("Output buffer size set to match input capacity = %d frames.\n", result);
+ printf("Output buffer size set to match input capacity = %d frames!\n", result);
}
}
@@ -565,6 +569,8 @@
}
}
+ argParser.compareWithStream(inputStream);
+
// ------- Setup loopbackData -----------------------------
loopbackData.actualInputFormat = AAudioStream_getFormat(inputStream);
diff --git a/media/libaaudio/examples/utils/AAudioSimplePlayer.h b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
index 1645986..4373fa9 100644
--- a/media/libaaudio/examples/utils/AAudioSimplePlayer.h
+++ b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
@@ -32,8 +32,6 @@
// Arbitrary period for glitches
#define FORCED_UNDERRUN_PERIOD_FRAMES (2 * 48000)
-// How long to sleep in a callback to cause an intentional glitch. For testing.
-#define FORCED_UNDERRUN_SLEEP_MICROS (10 * 1000)
#define MAX_TIMESTAMPS 16
@@ -275,7 +273,7 @@
int scheduler = 0;
bool schedulerChecked = false;
- bool forceUnderruns = false;
+ int32_t hangTimeMSec = 0;
AAudioSimplePlayer simplePlayer;
int32_t callbackCount = 0;
@@ -327,10 +325,12 @@
sineData->setupSineSweeps();
}
- if (sineData->forceUnderruns) {
+ if (sineData->hangTimeMSec > 0) {
if (sineData->framesTotal > sineData->nextFrameToGlitch) {
- usleep(FORCED_UNDERRUN_SLEEP_MICROS);
- printf("Simulate glitch at %lld\n", (long long) sineData->framesTotal);
+ usleep(sineData->hangTimeMSec * 1000);
+ printf("Hang callback at %lld frames for %d msec\n",
+ (long long) sineData->framesTotal,
+ sineData->hangTimeMSec);
sineData->nextFrameToGlitch += FORCED_UNDERRUN_PERIOD_FRAMES;
}
}
diff --git a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
index 7a48153..2b05f10 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
@@ -26,11 +26,14 @@
#include <string.h>
#include <time.h>
#include <aaudio/AAudio.h>
+
#include "AAudioExampleUtils.h"
#include "AAudioSimplePlayer.h"
#include "AAudioArgsParser.h"
-#define APP_VERSION "0.1.5"
+#define APP_VERSION "0.1.6"
+
+constexpr int32_t kDefaultHangTimeMSec = 10;
/**
* Open stream, play some sine waves, then close the stream.
@@ -41,7 +44,7 @@
static aaudio_result_t testOpenPlayClose(AAudioArgsParser &argParser,
int32_t loopCount,
int32_t prefixToneMsec,
- bool forceUnderruns)
+ int32_t hangTimeMSec)
{
SineThreadedData_t myData;
AAudioSimplePlayer &player = myData.simplePlayer;
@@ -53,10 +56,12 @@
printf("----------------------- run complete test --------------------------\n");
myData.schedulerChecked = false;
myData.callbackCount = 0;
- myData.forceUnderruns = forceUnderruns; // test AAudioStream_getXRunCount()
+ myData.hangTimeMSec = hangTimeMSec; // test AAudioStream_getXRunCount()
result = player.open(argParser,
- SimplePlayerDataCallbackProc, SimplePlayerErrorCallbackProc, &myData);
+ SimplePlayerDataCallbackProc,
+ SimplePlayerErrorCallbackProc,
+ &myData);
if (result != AAUDIO_OK) {
fprintf(stderr, "ERROR - player.open() returned %s\n",
AAudio_convertResultToText(result));
@@ -115,12 +120,17 @@
int64_t millis =
(getNanoseconds(CLOCK_MONOTONIC) - startedAtNanos) / NANOS_PER_MILLISECOND;
result = myData.waker.get();
+ const int32_t framesWritten = (int32_t) AAudioStream_getFramesWritten(player.getStream());
+ const int32_t framesRead = (int32_t) AAudioStream_getFramesRead(player.getStream());
+ const int32_t xruns = AAudioStream_getXRunCount(player.getStream());
printf(" waker result = %d, at %6d millis"
- ", second = %3d, framesWritten = %8d, underruns = %d\n",
+ ", second = %3d, frames written %8d - read %8d = %8d, underruns = %d\n",
result, (int) millis,
second,
- (int) AAudioStream_getFramesWritten(player.getStream()),
- (int) AAudioStream_getXRunCount(player.getStream()));
+ framesWritten,
+ framesRead,
+ framesWritten - framesRead,
+ xruns);
if (result != AAUDIO_OK) {
disconnected = (result == AAUDIO_ERROR_DISCONNECTED);
bailOut = true;
@@ -210,7 +220,9 @@
AAudioArgsParser::usage();
printf(" -l{count} loopCount start/stop, every other one is silent\n");
printf(" -t{msec} play a high pitched tone at the beginning\n");
- printf(" -z force periodic underruns by sleeping in callback\n");
+ printf(" -h{msec} force periodic underruns by hanging in callback\n");
+ printf(" If no value specified then %d used.\n",
+ kDefaultHangTimeMSec);
}
int main(int argc, const char **argv)
@@ -219,13 +231,14 @@
aaudio_result_t result;
int32_t loopCount = 1;
int32_t prefixToneMsec = 0;
- bool forceUnderruns = false;
+ int32_t hangTimeMSec = 0;
// Make printf print immediately so that debug info is not stuck
// in a buffer if we hang or crash.
setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
- printf("%s - Play a sine sweep using an AAudio callback V%s\n", argv[0], APP_VERSION);
+ printf("%s - Play a sine sweep using an AAudio callback V%s\n",
+ argv[0], APP_VERSION);
for (int i = 1; i < argc; i++) {
const char *arg = argv[i];
@@ -240,8 +253,10 @@
case 't':
prefixToneMsec = atoi(&arg[2]);
break;
- case 'z':
- forceUnderruns = true; // Zzzzzzz
+ case 'h':
+ hangTimeMSec = (arg[2]) // value specified?
+ ? atoi(&arg[2])
+ : kDefaultHangTimeMSec;
break;
default:
usage();
@@ -257,7 +272,8 @@
}
// Keep looping until we can complete the test without disconnecting.
- while((result = testOpenPlayClose(argParser, loopCount, prefixToneMsec, forceUnderruns))
+ while((result = testOpenPlayClose(argParser, loopCount,
+ prefixToneMsec, hangTimeMSec))
== AAUDIO_ERROR_DISCONNECTED);
return (result) ? EXIT_FAILURE : EXIT_SUCCESS;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 6d2329f..52cb5fa 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -4298,24 +4298,27 @@
int maxDimension = max(width, height);
static const int limits[][5] = {
- /* MBps MB dim bitrate level */
- { 1485, 99, 28, 64, OMX_VIDEO_AVCLevel1 },
- { 1485, 99, 28, 128, OMX_VIDEO_AVCLevel1b },
- { 3000, 396, 56, 192, OMX_VIDEO_AVCLevel11 },
- { 6000, 396, 56, 384, OMX_VIDEO_AVCLevel12 },
- { 11880, 396, 56, 768, OMX_VIDEO_AVCLevel13 },
- { 11880, 396, 56, 2000, OMX_VIDEO_AVCLevel2 },
- { 19800, 792, 79, 4000, OMX_VIDEO_AVCLevel21 },
- { 20250, 1620, 113, 4000, OMX_VIDEO_AVCLevel22 },
- { 40500, 1620, 113, 10000, OMX_VIDEO_AVCLevel3 },
- { 108000, 3600, 169, 14000, OMX_VIDEO_AVCLevel31 },
- { 216000, 5120, 202, 20000, OMX_VIDEO_AVCLevel32 },
- { 245760, 8192, 256, 20000, OMX_VIDEO_AVCLevel4 },
- { 245760, 8192, 256, 50000, OMX_VIDEO_AVCLevel41 },
- { 522240, 8704, 263, 50000, OMX_VIDEO_AVCLevel42 },
- { 589824, 22080, 420, 135000, OMX_VIDEO_AVCLevel5 },
- { 983040, 36864, 543, 240000, OMX_VIDEO_AVCLevel51 },
- { 2073600, 36864, 543, 240000, OMX_VIDEO_AVCLevel52 },
+ /* MBps MB dim bitrate level */
+ { 1485, 99, 28, 64, OMX_VIDEO_AVCLevel1 },
+ { 1485, 99, 28, 128, OMX_VIDEO_AVCLevel1b },
+ { 3000, 396, 56, 192, OMX_VIDEO_AVCLevel11 },
+ { 6000, 396, 56, 384, OMX_VIDEO_AVCLevel12 },
+ { 11880, 396, 56, 768, OMX_VIDEO_AVCLevel13 },
+ { 11880, 396, 56, 2000, OMX_VIDEO_AVCLevel2 },
+ { 19800, 792, 79, 4000, OMX_VIDEO_AVCLevel21 },
+ { 20250, 1620, 113, 4000, OMX_VIDEO_AVCLevel22 },
+ { 40500, 1620, 113, 10000, OMX_VIDEO_AVCLevel3 },
+ { 108000, 3600, 169, 14000, OMX_VIDEO_AVCLevel31 },
+ { 216000, 5120, 202, 20000, OMX_VIDEO_AVCLevel32 },
+ { 245760, 8192, 256, 20000, OMX_VIDEO_AVCLevel4 },
+ { 245760, 8192, 256, 50000, OMX_VIDEO_AVCLevel41 },
+ { 522240, 8704, 263, 50000, OMX_VIDEO_AVCLevel42 },
+ { 589824, 22080, 420, 135000, OMX_VIDEO_AVCLevel5 },
+ { 983040, 36864, 543, 240000, OMX_VIDEO_AVCLevel51 },
+ { 2073600, 36864, 543, 240000, OMX_VIDEO_AVCLevel52 },
+ { 4177920, 139264, 1055, 240000, OMX_VIDEO_AVCLevel6 },
+ { 8355840, 139264, 1055, 480000, OMX_VIDEO_AVCLevel61 },
+ { 16711680, 139264, 1055, 800000, OMX_VIDEO_AVCLevel62 },
};
for (size_t i = 0; i < ARRAY_SIZE(limits); i++) {
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 2e7da01..82f7026 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -192,6 +192,9 @@
{ 50, OMX_VIDEO_AVCLevel5 },
{ 51, OMX_VIDEO_AVCLevel51 },
{ 52, OMX_VIDEO_AVCLevel52 },
+ { 60, OMX_VIDEO_AVCLevel6 },
+ { 61, OMX_VIDEO_AVCLevel61 },
+ { 62, OMX_VIDEO_AVCLevel62 },
};
const static ALookup<uint8_t, OMX_VIDEO_AVCPROFILETYPE> profiles {
{ 66, OMX_VIDEO_AVCProfileBaseline },
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 2854a98..5c22727 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -953,8 +953,8 @@
// FIXME: in case of RENDER policy, the output capabilities should be checked
if ((usePrimaryOutputFromPolicyMixes || !secondaryDescs->empty())
- && !audio_has_proportional_frames(config->format)) {
- ALOGW("%s: audio loopback only supports proportional frames", __func__);
+ && !audio_is_linear_pcm(config->format)) {
+ ALOGD("%s: rejecting request as dynamic audio policy only support pcm", __func__);
return BAD_VALUE;
}
if (usePrimaryOutputFromPolicyMixes) {