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Allows configuration of playout audio buffer

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Playout audio buffer length in Java audio device configuration with fieldtrial.

Bug: webrtc:10928
Change-Id: I79286f09591f4b2c6a6146f23d3dce92a29f6b21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150657
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#29005}
This commit is contained in:
Alex Narest
2019-08-29 13:17:11 +02:00
committed by Commit Bot
parent 008213a45b
commit 44dc241ae8
6 changed files with 38 additions and 15 deletions
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1
modules/audio_device/BUILD.gn
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@ -178,6 +178,7 @@ rtc_source_set("audio_device_impl") {
"../../rtc_base/system:file_wrapper",
"../../rtc_base/task_utils:repeating_task",
"../../system_wrappers",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"../utility",
"//third_party/abseil-cpp/absl/memory",

13
modules/audio_device/android/audio_track_jni.cc
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@ -18,6 +18,7 @@
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_thread.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@ -26,7 +27,7 @@ AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
NativeRegistration* native_reg,
std::unique_ptr<GlobalRef> audio_track)
: audio_track_(std::move(audio_track)),
init_playout_(native_reg->GetMethodId("initPlayout", "(II)Z")),
init_playout_(native_reg->GetMethodId("initPlayout", "(IID)Z")),
start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
@ -37,7 +38,15 @@ AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
bool AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
return audio_track_->CallBooleanMethod(init_playout_, sample_rate, channels);
double buffer_size_factor =
strtod(webrtc::field_trial::FindFullName(
"WebRTC-AudioDevicePlayoutBufferSizeFactor")
.c_str(),
nullptr);
if (buffer_size_factor == 0)
buffer_size_factor = 1.0;
return audio_track_->CallBooleanMethod(init_playout_, sample_rate, channels,
buffer_size_factor);
}
bool AudioTrackJni::JavaAudioTrack::StartPlayout() {

14
modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
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@ -215,9 +215,11 @@ public class WebRtcAudioTrack {
}
}
private boolean initPlayout(int sampleRate, int channels) {
private boolean initPlayout(int sampleRate, int channels, double bufferSizeFactor) {
threadChecker.checkIsOnValidThread();
Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels + ")");
Logging.d(TAG,
"initPlayout(sampleRate=" + sampleRate + ", channels=" + channels
+ ", bufferSizeFactor=" + bufferSizeFactor + ")");
final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
@ -230,11 +232,11 @@ public class WebRtcAudioTrack {
// Get the minimum buffer size required for the successful creation of an
// AudioTrack object to be created in the MODE_STREAM mode.
// Note that this size doesn't guarantee a smooth playback under load.
// TODO(henrika): should we extend the buffer size to avoid glitches?
final int channelConfig = channelCountToConfiguration(channels);
final int minBufferSizeInBytes =
AudioTrack.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
Logging.d(TAG, "AudioTrack.getMinBufferSize: " + minBufferSizeInBytes);
final int minBufferSizeInBytes = (int) (AudioTrack.getMinBufferSize(sampleRate, channelConfig,
AudioFormat.ENCODING_PCM_16BIT)
* bufferSizeFactor);
Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes);
// For the streaming mode, data must be written to the audio sink in
// chunks of size (given by byteBuffer.capacity()) less than or equal
// to the total buffer size |minBufferSizeInBytes|. But, we have seen

1
sdk/android/BUILD.gn
View File

@ -1108,6 +1108,7 @@ if (current_os == "linux" || is_android) {
"../../modules/audio_device:audio_device_buffer",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/types:optional",
]

14
sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
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@ -182,9 +182,11 @@ class WebRtcAudioTrack {
}
@CalledByNative
private boolean initPlayout(int sampleRate, int channels) {
private boolean initPlayout(int sampleRate, int channels, double bufferSizeFactor) {
threadChecker.checkIsOnValidThread();
Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels + ")");
Logging.d(TAG,
"initPlayout(sampleRate=" + sampleRate + ", channels=" + channels
+ ", bufferSizeFactor=" + bufferSizeFactor + ")");
final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
@ -197,11 +199,11 @@ class WebRtcAudioTrack {
// Get the minimum buffer size required for the successful creation of an
// AudioTrack object to be created in the MODE_STREAM mode.
// Note that this size doesn't guarantee a smooth playback under load.
// TODO(henrika): should we extend the buffer size to avoid glitches?
final int channelConfig = channelCountToConfiguration(channels);
final int minBufferSizeInBytes =
AudioTrack.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
Logging.d(TAG, "AudioTrack.getMinBufferSize: " + minBufferSizeInBytes);
final int minBufferSizeInBytes = (int) (AudioTrack.getMinBufferSize(sampleRate, channelConfig,
AudioFormat.ENCODING_PCM_16BIT)
* bufferSizeFactor);
Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes);
// For the streaming mode, data must be written to the audio sink in
// chunks of size (given by byteBuffer.capacity()) less than or equal
// to the total buffer size |minBufferSizeInBytes|. But, we have seen

10
sdk/android/src/jni/audio_device/audio_track_jni.cc
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@ -19,6 +19,7 @@
#include "rtc_base/platform_thread.h"
#include "sdk/android/generated_java_audio_device_module_native_jni/WebRtcAudioTrack_jni.h"
#include "sdk/android/src/jni/jni_helpers.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@ -81,9 +82,16 @@ int32_t AudioTrackJni::InitPlayout() {
return 0;
}
RTC_DCHECK(!playing_);
double buffer_size_factor =
strtod(webrtc::field_trial::FindFullName(
"WebRTC-AudioDevicePlayoutBufferSizeFactor")
.c_str(),
nullptr);
if (buffer_size_factor == 0)
buffer_size_factor = 1.0;
if (!Java_WebRtcAudioTrack_initPlayout(
env_, j_audio_track_, audio_parameters_.sample_rate(),
static_cast<int>(audio_parameters_.channels()))) {
static_cast<int>(audio_parameters_.channels()), buffer_size_factor)) {
RTC_LOG(LS_ERROR) << "InitPlayout failed";
return -1;
}