Adds support for OpenSL ES based audio capture on Android

webrtc/modules/audio_device/android/opensles_common.cc

Index: webrtc/modules/audio_device/android/opensles_common.cc
diff --git a/webrtc/modules/audio_device/android/opensles_common.cc b/webrtc/modules/audio_device/android/opensles_common.cc
index 9e3cbf7b9a9da6f5f20a53761d585d696701ee4a..d0768efaa6e89040f0d9e30a78e572f20a058560 100644
--- a/webrtc/modules/audio_device/android/opensles_common.cc
+++ b/webrtc/modules/audio_device/android/opensles_common.cc
@@ -10,13 +10,10 @@
 
 #include "webrtc/modules/audio_device/android/opensles_common.h"
 
-#include <assert.h>
 #include <SLES/OpenSLES.h>
 
 #include "webrtc/base/arraysize.h"
-#include "webrtc/modules/audio_device/android/audio_common.h"
-
-using webrtc::kNumChannels;
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 
@@ -49,24 +46,47 @@ const char* GetSLErrorString(size_t code) {
   return sl_error_strings[code];
 }
 
-SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) {
-  SLDataFormat_PCM configuration;
-  configuration.formatType = SL_DATAFORMAT_PCM;
-  configuration.numChannels = kNumChannels;
-  // According to the opensles documentation in the ndk:
-  // samplesPerSec is actually in units of milliHz, despite the misleading name.
-  // It further recommends using constants. However, this would lead to a lot
-  // of boilerplate code so it is not done here.
-  configuration.samplesPerSec = sample_rate * 1000;
-  configuration.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
-  configuration.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
-  configuration.channelMask = SL_SPEAKER_FRONT_CENTER;
-  if (2 == configuration.numChannels) {
-    configuration.channelMask =
-        SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
+                                        int sample_rate,
+                                        size_t bits_per_sample) {
+  RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
+  SLDataFormat_PCM format;
+  format.formatType = SL_DATAFORMAT_PCM;
+  format.numChannels = static_cast<SLuint32>(channels);
+  // Note that, the unit of sample rate is actually in milliHertz and not Hertz.
+  switch (sample_rate) {
+    case 8000:
+      format.samplesPerSec = SL_SAMPLINGRATE_8;
+      break;
+    case 16000:
+      format.samplesPerSec = SL_SAMPLINGRATE_16;
+      break;
+    case 22050:
+      format.samplesPerSec = SL_SAMPLINGRATE_22_05;
+      break;
+    case 32000:
+      format.samplesPerSec = SL_SAMPLINGRATE_32;
+      break;
+    case 44100:
+      format.samplesPerSec = SL_SAMPLINGRATE_44_1;
+      break;
+    case 48000:
+      format.samplesPerSec = SL_SAMPLINGRATE_48;
+      break;
+    default:

[tommi, 2016/09/15 09:34:13]

I think I've read somewhere that 96kHz is supported (possibly 88.2kHz then as
well?).

Do we have any stats from the field that give us confidence that we will never
see higher than 48kHz in should-be-supported situations?

[henrika_webrtc, 2016/09/16 13:30:47]

Great feedback. You are correct. I have never seen such a device but I am adding
it just in case.

+      RTC_CHECK(false) << "Unsupported sample rate: " << sample_rate;

[tommi, 2016/09/15 09:34:12]

nit: also include a break; here for consistency.

[henrika_webrtc, 2016/09/16 13:30:47]

Done.

   }
-  configuration.endianness = SL_BYTEORDER_LITTLEENDIAN;
-  return configuration;
+  format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.endianness = SL_BYTEORDER_LITTLEENDIAN;
+  if (format.numChannels == 1)

[tommi, 2016/09/15 09:34:13]

{}
(see lines below)

[henrika_webrtc, 2016/09/16 13:30:47]

Done.

+    format.channelMask = SL_SPEAKER_FRONT_CENTER;
+  else if (format.numChannels == 2)
+    format.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+  else
+    RTC_CHECK(false) << "Unsupported number of channels: "
+                     << format.numChannels;
+  return format;
 }
 
-}  // namespace webrtc_opensl
+}  // namespace webrtc