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Set max bitrate for audio send stream based on RtpParameters.

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BUG=webrtc:7392

Review-Url: https://codereview.webrtc.org/2775483004
Cr-Commit-Position: refs/heads/master@{#17399}
This commit is contained in:
minyue 2017-03-27 13:04:25 -07:00 committed by Commit bot
parent 488b85e502
commit cecec1060f
2 changed files with 78 additions and 14 deletions
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35
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1346,10 +1346,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
return false;
}
const rtc::Optional<int> old_rtp_max_bitrate =
rtp_parameters_.encodings[0].max_bitrate_bps;
rtp_parameters_ = parameters;
// parameters.encodings[0].encodings[0].max_bitrate_bps could have changed.
if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
if (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) {
// Recreate AudioSendStream with new bit rate.
config_.send_codec_spec.codec_inst.rate = *send_rate;
RecreateAudioSendStream();
@ -1381,7 +1383,15 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
RTC_DCHECK(!stream_);
if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
config_.min_bitrate_bps = kOpusMinBitrateBps;
config_.max_bitrate_bps = kOpusBitrateFbBps;
// This means that when RtpParameters is reset, we may change the
// encoder's bit rate immediately (through call_->CreateAudioSendStream),
// meanwhile change the cap to the output of BWE.
config_.max_bitrate_bps =
rtp_parameters_.encodings[0].max_bitrate_bps
? *rtp_parameters_.encodings[0].max_bitrate_bps
: kOpusBitrateFbBps;
// TODO(mflodman): Keep testing this and set proper values.
// Note: This is an early experiment currently only supported by Opus.
if (send_side_bwe_with_overhead_) {
@ -1390,8 +1400,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
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.
@ -1402,7 +1410,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
#else
max_packet_size_ms = 60;
#endif
min_packet_size_ms = 20;
}
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
@ -1411,11 +1418,19 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
int min_overhead_bps =
kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;
int max_overhead_bps =
kOverheadPerPacket * 8 * 1000 / min_packet_size_ms;
// We assume that |config_.max_bitrate_bps| before the next line is
// a hard limit on the payload bitrate, so we add min_overhead_bps to
// it to ensure that, when overhead is deducted, the payload rate
// never goes beyond the limit.
// Note: this also means that if a higher overhead is forced, we
// cannot reach the limit.
// TODO(minyue): Reconsider this when the signaling to BWE is done
// through a dedicated API.
config_.max_bitrate_bps += min_overhead_bps;
config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps;
config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps;
// In contrast to max_bitrate_bps, we let min_bitrate_bps always be
// reachable.
config_.min_bitrate_bps += min_overhead_bps;
}
}
}

57
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -1172,6 +1172,29 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrcX));
}
// Test that max_bitrate_bps in send stream config gets updated correctly when
// SetRtpSendParameters is called.
TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
webrtc::test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/");
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters send_parameters;
send_parameters.codecs.push_back(kOpusCodec);
SetSendParameters(send_parameters);
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
// Expect empty on parameters.encodings[0].max_bitrate_bps;
EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps);
constexpr int kMaxBitrateBps = 6000;
rtp_parameters.encodings[0].max_bitrate_bps =
rtc::Optional<int>(kMaxBitrateBps);
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
EXPECT_EQ(max_bitrate, kMaxBitrateBps);
}
// Test that GetRtpReceiveParameters returns the currently configured codecs.
TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) {
EXPECT_TRUE(SetupRecvStream());
@ -2585,13 +2608,12 @@ TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest, MinAndMaxBitrate) {
constexpr int kOpusMaxPtimeMs = WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
constexpr int kMinOverheadBps =
kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs;
constexpr int kMaxOverheadBps = kOverheadPerPacket * 8 * 1000 / 10;
constexpr int kOpusMinBitrateBps = 6000;
EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadBps,
GetSendStreamConfig(kSsrcX).min_bitrate_bps);
constexpr int kOpusBitrateFbBps = 32000;
EXPECT_EQ(kOpusBitrateFbBps + kMaxOverheadBps,
EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadBps,
GetSendStreamConfig(kSsrcX).max_bitrate_bps);
parameters.options.audio_network_adaptor = rtc::Optional<bool>(true);
@ -2601,15 +2623,42 @@ TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest, MinAndMaxBitrate) {
constexpr int kMinOverheadWithAnaBps =
kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs;
constexpr int kMaxOverheadWithAnaBps = kOverheadPerPacket * 8 * 1000 / 20;
EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadWithAnaBps,
GetSendStreamConfig(kSsrcX).min_bitrate_bps);
EXPECT_EQ(kOpusBitrateFbBps + kMaxOverheadWithAnaBps,
EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadWithAnaBps,
GetSendStreamConfig(kSsrcX).max_bitrate_bps);
}
// This test is similar to
// WebRtcVoiceEngineTestFake.SetRtpSendParameterUpdatesMaxBitrate but with an
// additional field trial.
TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest,
SetRtpSendParameterUpdatesMaxBitrate) {
EXPECT_TRUE(SetupSendStream());
cricket::AudioSendParameters send_parameters;
send_parameters.codecs.push_back(kOpusCodec);
SetSendParameters(send_parameters);
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
// Expect empty on parameters.encodings[0].max_bitrate_bps;
EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps);
constexpr int kMaxBitrateBps = 6000;
rtp_parameters.encodings[0].max_bitrate_bps =
rtc::Optional<int>(kMaxBitrateBps);
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
constexpr int kMinOverhead = 3333;
#else
constexpr int kMinOverhead = 6666;
#endif
EXPECT_EQ(max_bitrate, kMaxBitrateBps + kMinOverhead);
}
// Test that we can set the outgoing SSRC properly.
// SSRC is set in SetupSendStream() by calling AddSendStream.
TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) {