diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc index 52c30f0987..3ce35e7a4b 100644 --- a/webrtc/api/call/audio_send_stream.cc +++ b/webrtc/api/call/audio_send_stream.cc @@ -41,8 +41,8 @@ std::string AudioSendStream::Config::ToString() const { ss << "{rtp: " << rtp.ToString(); ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr"); ss << ", voe_channel_id: " << voe_channel_id; - ss << ", min_bitrate_kbps: " << min_bitrate_kbps; - ss << ", max_bitrate_kbps: " << max_bitrate_kbps; + ss << ", min_bitrate_bps: " << min_bitrate_bps; + ss << ", max_bitrate_bps: " << max_bitrate_bps; ss << ", send_codec_spec: " << send_codec_spec.ToString(); ss << '}'; return ss.str(); diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h index 78ab8ec52e..658c9de371 100644 --- a/webrtc/api/call/audio_send_stream.h +++ b/webrtc/api/call/audio_send_stream.h @@ -90,8 +90,8 @@ class AudioSendStream { // Bitrate limits used for variable audio bitrate streams. Set both to -1 to // disable audio bitrate adaptation. // Note: This is still an experimental feature and not ready for real usage. - int min_bitrate_kbps = -1; - int max_bitrate_kbps = -1; + int min_bitrate_bps = -1; + int max_bitrate_bps = -1; // Defines whether to turn on audio network adaptor, and defines its config // string. diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc index ad6366bc80..7e8e426ed9 100644 --- a/webrtc/audio/audio_send_stream.cc +++ b/webrtc/audio/audio_send_stream.cc @@ -101,12 +101,12 @@ AudioSendStream::~AudioSendStream() { void AudioSendStream::Start() { RTC_DCHECK(thread_checker_.CalledOnValidThread()); - if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) { - RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps); + if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) { + RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps); rtc::Event thread_sync_event(false /* manual_reset */, false); worker_queue_->PostTask([this, &thread_sync_event] { - bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000, - config_.max_bitrate_kbps * 1000, 0, true); + bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps, + config_.max_bitrate_bps, 0, true); thread_sync_event.Set(); }); thread_sync_event.Wait(rtc::Event::kForever); @@ -249,10 +249,10 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps, uint8_t fraction_loss, int64_t rtt) { RTC_DCHECK_GE(bitrate_bps, - static_cast(config_.min_bitrate_kbps * 1000)); + static_cast(config_.min_bitrate_bps)); // The bitrate allocator might allocate an higher than max configured bitrate // if there is room, to allow for, as example, extra FEC. Ignore that for now. - const uint32_t max_bitrate_bps = config_.max_bitrate_kbps * 1000; + const uint32_t max_bitrate_bps = config_.max_bitrate_bps; if (bitrate_bps > max_bitrate_bps) bitrate_bps = max_bitrate_bps; diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc index f310d00ba9..bd7ffb7751 100644 --- a/webrtc/audio/audio_send_stream_unittest.cc +++ b/webrtc/audio/audio_send_stream_unittest.cc @@ -215,8 +215,8 @@ TEST(AudioSendStreamTest, ConfigToString) { config.rtp.ssrc = kSsrc; config.rtp.c_name = kCName; config.voe_channel_id = kChannelId; - config.min_bitrate_kbps = 12; - config.max_bitrate_kbps = 34; + config.min_bitrate_bps = 12000; + config.max_bitrate_bps = 34000; config.send_codec_spec.nack_enabled = true; config.send_codec_spec.transport_cc_enabled = false; config.send_codec_spec.enable_codec_fec = true; @@ -233,7 +233,7 @@ TEST(AudioSendStreamTest, ConfigToString) { "{rtp: {ssrc: 1234, extensions: [{uri: " "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " - "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, " + "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: " "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: " diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc index df64f41051..3cdcc991d2 100644 --- a/webrtc/call/rampup_tests.cc +++ b/webrtc/call/rampup_tests.cc @@ -212,8 +212,8 @@ void RampUpTester::ModifyAudioConfigs( send_config->rtp.ssrc = audio_ssrcs_[0]; send_config->rtp.extensions.clear(); - send_config->min_bitrate_kbps = 6; - send_config->max_bitrate_kbps = 60; + send_config->min_bitrate_bps = 6000; + send_config->max_bitrate_bps = 60000; bool transport_cc = false; if (extension_type_ == RtpExtension::kAbsSendTimeUri) { diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc index 99369a2547..6f24ef8b27 100644 --- a/webrtc/media/engine/webrtcvoiceengine.cc +++ b/webrtc/media/engine/webrtcvoiceengine.cc @@ -83,16 +83,16 @@ constexpr int kNackRtpHistoryMs = 5000; // 64-128 kb/s for FB stereo music. // The current implementation applies the following values to mono signals, // and multiplies them by 2 for stereo. -const int kOpusBitrateNb = 12000; -const int kOpusBitrateWb = 20000; -const int kOpusBitrateFb = 32000; +const int kOpusBitrateNbBps = 12000; +const int kOpusBitrateWbBps = 20000; +const int kOpusBitrateFbBps = 32000; // Opus bitrate should be in the range between 6000 and 510000. -const int kOpusMinBitrate = 6000; -const int kOpusMaxBitrate = 510000; +const int kOpusMinBitrateBps = 6000; +const int kOpusMaxBitrateBps = 510000; // iSAC bitrate should be <= 56000. -const int kIsacMaxBitrate = 56000; +const int kIsacMaxBitrateBps = 56000; // Default audio dscp value. // See http://tools.ietf.org/html/rfc2474 for details. @@ -222,18 +222,19 @@ int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) { } if (bitrate <= 0) { if (max_playback_rate <= 8000) { - bitrate = kOpusBitrateNb; + bitrate = kOpusBitrateNbBps; } else if (max_playback_rate <= 16000) { - bitrate = kOpusBitrateWb; + bitrate = kOpusBitrateWbBps; } else { - bitrate = kOpusBitrateFb; + bitrate = kOpusBitrateFbBps; } if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) { bitrate *= 2; } - } else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) { - bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate; + } else if (bitrate < kOpusMinBitrateBps || bitrate > kOpusMaxBitrateBps) { + bitrate = (bitrate < kOpusMinBitrateBps) ? kOpusMinBitrateBps + : kOpusMaxBitrateBps; std::string rate_source = use_param ? "Codec parameter \"maxaveragebitrate\"" : "Supplied Opus bitrate"; @@ -478,9 +479,9 @@ class WebRtcVoiceCodecs final { }; const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[11] = { - {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrate}, - {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrate}, - {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrate}, + {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps}, + {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps}, + {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrateBps}, // G722 should be advertised as 8000 Hz because of the RFC "bug". {kG722CodecName, 8000, 1, 9, false, {10, 20, 30, 40, 50, 60}}, {kIlbcCodecName, 8000, 1, 102, false, {20, 30, 40, 60}}, @@ -489,8 +490,7 @@ const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[11] = { {kCnCodecName, 32000, 1, 106, false, {}}, {kCnCodecName, 16000, 1, 105, false, {}}, {kCnCodecName, 8000, 1, 13, false, {}}, - {kDtmfCodecName, 8000, 1, 126, false, {}} -}; + {kDtmfCodecName, 8000, 1, 126, false, {}}}; rtc::Optional ComputeSendBitrate(int max_send_bitrate_bps, int rtp_max_bitrate_bps, @@ -1392,8 +1392,8 @@ 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_kbps = kOpusMinBitrate; - config_.max_bitrate_kbps = kOpusBitrateFb; + config_.min_bitrate_bps = kOpusMinBitrateBps; + config_.max_bitrate_bps = kOpusBitrateFbBps; } stream_ = call_->CreateAudioSendStream(config_); RTC_CHECK(stream_); diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc index c773e43150..c80647da4b 100644 --- a/webrtc/video/video_quality_test.cc +++ b/webrtc/video/video_quality_test.cc @@ -46,8 +46,8 @@ constexpr int kPayloadTypeVP8 = 123; constexpr int kPayloadTypeVP9 = 124; constexpr size_t kMaxComparisons = 10; constexpr char kSyncGroup[] = "av_sync"; -constexpr int kOpusMinBitrate = 6000; -constexpr int kOpusBitrateFb = 32000; +constexpr int kOpusMinBitrateBps = 6000; +constexpr int kOpusBitrateFbBps = 32000; struct VoiceEngineState { VoiceEngineState() @@ -1264,8 +1264,8 @@ void VideoQualityTest::SetupAudio(int send_channel_id, audio_send_config_.rtp.extensions.push_back( webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri, test::kTransportSequenceNumberExtensionId)); - audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000; - audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000; + audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps; + audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps; } audio_send_config_.send_codec_spec.codec_inst = CodecInst{120, "OPUS", 48000, 960, 2, 64000};