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Increase the default maximum jitter buffer size to 200 packets.

Browse Source 
Bug: webrtc:10415
Change-Id: Iec5a5a263c11d92a23290c1c2de053fe9e5d5839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128082
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27142}
This commit is contained in:
Jakob Ivarsson 2019-03-15 10:37:31 +01:00 committed by Commit Bot
parent dbce09003d
commit 647d5e6d91
6 changed files with 7 additions and 7 deletions
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2
api/peer_connection_interface.h
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@ -351,7 +351,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
static const int kUndefined = -1;
// Default maximum number of packets in the audio jitter buffer.
static const int kAudioJitterBufferMaxPackets = 50;
static const int kAudioJitterBufferMaxPackets = 200;
// ICE connection receiving timeout for aggressive configuration.
static const int kAggressiveIceConnectionReceivingTimeout = 1000;

2
call/audio_receive_stream.h
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@ -114,7 +114,7 @@ class AudioReceiveStream {
MediaTransportInterface* media_transport = nullptr;
// NetEq settings.
size_t jitter_buffer_max_packets = 50;
size_t jitter_buffer_max_packets = 200;
bool jitter_buffer_fast_accelerate = false;
int jitter_buffer_min_delay_ms = 0;
bool jitter_buffer_enable_rtx_handling = false;

2
media/engine/webrtc_voice_engine.cc
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@ -272,7 +272,7 @@ void WebRtcVoiceEngine::Init() {
options.noise_suppression = true;
options.highpass_filter = true;
options.stereo_swapping = false;
options.audio_jitter_buffer_max_packets = 50;
options.audio_jitter_buffer_max_packets = 200;
options.audio_jitter_buffer_fast_accelerate = false;
options.audio_jitter_buffer_min_delay_ms = 0;
options.audio_jitter_buffer_enable_rtx_handling = false;

2
media/engine/webrtc_voice_engine.h
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@ -133,7 +133,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
absl::optional<bool> delay_agnostic_aec_;
absl::optional<bool> experimental_ns_;
// Jitter buffer settings for new streams.
size_t audio_jitter_buffer_max_packets_ = 50;
size_t audio_jitter_buffer_max_packets_ = 200;
bool audio_jitter_buffer_fast_accelerate_ = false;
int audio_jitter_buffer_min_delay_ms_ = 0;
bool audio_jitter_buffer_enable_rtx_handling_ = false;

4
media/engine/webrtc_voice_engine_unittest.cc
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@ -2841,7 +2841,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
.Times(2)
.WillRepeatedly(Return(false));
EXPECT_EQ(50u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
// Nothing set in AudioOptions, so everything should be as default.
@ -2850,7 +2850,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
EXPECT_TRUE(IsEchoCancellationEnabled());
EXPECT_TRUE(IsHighPassFilterEnabled());
EXPECT_TRUE(IsTypingDetectionEnabled());
EXPECT_EQ(50u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
// Turn typing detection off.

2
modules/audio_coding/neteq/include/neteq.h
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@ -123,7 +123,7 @@ class NetEq {
int sample_rate_hz = 16000; // Initial value. Will change with input data.
bool enable_post_decode_vad = false;
size_t max_packets_in_buffer = 50;
size_t max_packets_in_buffer = 200;
int max_delay_ms = 0;
int min_delay_ms = 0;
bool enable_fast_accelerate = false;