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Add overhead to audio bwe min, max.

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BUG=webrtc:6762

Review-Url: https://codereview.webrtc.org/2532433002
Cr-Commit-Position: refs/heads/master@{#16014}
This commit is contained in:
michaelt 2017-01-11 10:17:59 -08:00 committed by Commit bot
parent e8084706c3
commit 6672b26d02
2 changed files with 96 additions and 2 deletions
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54
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -405,6 +405,24 @@ class WebRtcVoiceCodecs final {
return 0;
}
static rtc::ArrayView<const int> GetPacketSizesMs(
const webrtc::CodecInst& codec) {
for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
if (IsCodec(codec, kCodecPrefs[i].name)) {
size_t num_packet_sizes = kMaxNumPacketSize;
for (int index = 0; index < kMaxNumPacketSize; index++) {
if (kCodecPrefs[i].packet_sizes_ms[index] == 0) {
num_packet_sizes = index;
break;
}
}
return rtc::ArrayView<const int>(kCodecPrefs[i].packet_sizes_ms,
num_packet_sizes);
}
}
return rtc::ArrayView<const int>();
}
// If the AudioCodec param kCodecParamPTime is set, then we will set it to
// codec pacsize if it's valid, or we will pick the next smallest value we
// support.
@ -1445,8 +1463,40 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
"Enabled") {
// TODO(mflodman): Keep testing this and set proper values.
// Note: This is an early experiment currently only supported by Opus.
config_.min_bitrate_bps = kOpusMinBitrateBps;
config_.max_bitrate_bps = kOpusBitrateFbBps;
if (webrtc::field_trial::FindFullName(
"WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs(
config_.send_codec_spec.codec_inst);
if (!packet_sizes_ms.empty()) {
int max_packet_size_ms =
*std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
int min_packet_size_ms =
*std::min_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
// Audio network adaptor will just use 20ms and 60ms frame lengths.
// The adaptor will only be active for the Opus encoder.
if (config_.audio_network_adaptor_config &&
IsCodec(config_.send_codec_spec.codec_inst, kOpusCodecName)) {
max_packet_size_ms = 60;
min_packet_size_ms = 20;
}
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
int min_overhead_bps =
kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;
int max_overhead_bps =
kOverheadPerPacket * 8 * 1000 / min_packet_size_ms;
config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps;
config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps;
}
} else {
config_.min_bitrate_bps = kOpusMinBitrateBps;
config_.max_bitrate_bps = kOpusBitrateFbBps;
}
}
stream_ = call_->CreateAudioSendStream(config_);
RTC_CHECK(stream_);

44
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -2548,6 +2548,50 @@ TEST_F(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
GetAudioNetworkAdaptorConfig(kSsrc1));
}
class WebRtcVoiceEngineWithSendSideBweWithOverheadTest
: public WebRtcVoiceEngineTestFake {
public:
WebRtcVoiceEngineWithSendSideBweWithOverheadTest()
: WebRtcVoiceEngineTestFake(
"WebRTC-Audio-SendSideBwe/Enabled/WebRTC-SendSideBwe-WithOverhead/"
"Enabled/") {}
};
TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest, MinAndMaxBitrate) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters parameters;
parameters.codecs.push_back(kOpusCodec);
SetSendParameters(parameters);
const int initial_num = call_.GetNumCreatedSendStreams();
EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
constexpr int kMinOverheadBps = kOverheadPerPacket * 8 * 1000 / 60;
constexpr int kMaxOverheadBps = kOverheadPerPacket * 8 * 1000 / 10;
constexpr int kOpusMinBitrateBps = 6000;
EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadBps,
GetSendStreamConfig(kSsrc1).min_bitrate_bps);
constexpr int kOpusBitrateFbBps = 32000;
EXPECT_EQ(kOpusBitrateFbBps + kMaxOverheadBps,
GetSendStreamConfig(kSsrc1).max_bitrate_bps);
parameters.options.audio_network_adaptor = rtc::Optional<bool>(true);
parameters.options.audio_network_adaptor_config =
rtc::Optional<std::string>("1234");
SetSendParameters(parameters);
constexpr int kMinOverheadWithAnaBps = kOverheadPerPacket * 8 * 1000 / 60;
constexpr int kMaxOverheadWithAnaBps = kOverheadPerPacket * 8 * 1000 / 20;
EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadWithAnaBps,
GetSendStreamConfig(kSsrc1).min_bitrate_bps);
EXPECT_EQ(kOpusBitrateFbBps + kMaxOverheadWithAnaBps,
GetSendStreamConfig(kSsrc1).max_bitrate_bps);
}
// Test that we can set the outgoing SSRC properly.
// SSRC is set in SetupSendStream() by calling AddSendStream.
TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) {