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Adds support for OpenSL ES based audio capture on Android.

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NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that
OpenSL ES is not accidentally activated in existing clients. There are still some
unresolved issues to sort out before it can be utilized.

Enables possibility to use OpenSL ES based audio in both directions for WebRTC.
All unit tests and demo clients have been tested with the new implementation but
the new support is behind a flag (see above).

More testing is needed before it can be used in the field and additional support for
hardware effects is still missing.

BUG=webrtc:5925
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2119633004 .

Cr-Commit-Position: refs/heads/master@{#14290}
This commit is contained in:
henrika
2016-09-19 15:44:09 +02:00
parent 27e177c8d8
commit 918b554789
21 changed files with 888 additions and 159 deletions
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27
webrtc/modules/audio_device/android/audio_manager_unittest.cc
View File

@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <SLES/OpenSLES_Android.h>
#include "testing/gtest/include/gtest/gtest.h"
@ -128,6 +127,11 @@ TEST_F(AudioManagerTest, IsLowLatencyPlayoutSupported) {
audio_manager()->IsLowLatencyPlayoutSupported() ? "Yes" : "No");
}
TEST_F(AudioManagerTest, IsLowLatencyRecordSupported) {
PRINT("%sLow latency input support: %s\n", kTag,
audio_manager()->IsLowLatencyRecordSupported() ? "Yes" : "No");
}
TEST_F(AudioManagerTest, IsProAudioSupported) {
PRINT("%sPro audio support: %s\n", kTag,
audio_manager()->IsProAudioSupported() ? "Yes" : "No");
@ -135,6 +139,7 @@ TEST_F(AudioManagerTest, IsProAudioSupported) {
TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
PRINT("PLAYOUT:\n");
PRINT("%saudio layer: %s\n", kTag,
low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
@ -144,7 +149,8 @@ TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
playout_parameters_.frames_per_buffer(),
playout_parameters_.GetBufferSizeInMilliseconds());
PRINT("RECORD: \n");
PRINT("%saudio layer: %s\n", kTag, "Java/JNI based AudioRecord");
PRINT("%saudio layer: %s\n", kTag,
low_latency_in ? "Low latency OpenSL" : "Java/JNI based AudioRecord");
PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
PRINT("%schannels: %" PRIuS "\n", kTag, record_parameters_.channels());
PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
@ -152,6 +158,21 @@ TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
record_parameters_.GetBufferSizeInMilliseconds());
}
// The audio device module only suppors the same sample rate in both directions.
// In addition, in full-duplex low-latency mode (OpenSL ES), both input and
// output must use the same native buffer size to allow for usage of the fast
// audio track in Android.
TEST_F(AudioManagerTest, VerifyAudioParameters) {
const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
EXPECT_EQ(playout_parameters_.sample_rate(),
record_parameters_.sample_rate());
if (low_latency_out && low_latency_in) {
EXPECT_EQ(playout_parameters_.frames_per_buffer(),
record_parameters_.frames_per_buffer());
}
}
// Add device-specific information to the test for logging purposes.
TEST_F(AudioManagerTest, ShowDeviceInfo) {
BuildInfo build_info;
@ -167,7 +188,7 @@ TEST_F(AudioManagerTest, ShowBuildInfo) {
PRINT("%sbuild release: %s\n", kTag, build_info.GetBuildRelease().c_str());
PRINT("%sbuild id: %s\n", kTag, build_info.GetAndroidBuildId().c_str());
PRINT("%sbuild type: %s\n", kTag, build_info.GetBuildType().c_str());
PRINT("%sSDK version: %s\n", kTag, build_info.GetSdkVersion().c_str());
PRINT("%sSDK version: %d\n", kTag, build_info.GetSdkVersion());
}
// Basic test of the AudioParameters class using default construction where