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Add more audio control and safe defaults

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This commit is contained in:
Jim Gustafson 2023-08-23 10:42:30 -07:00 committed by GitHub
parent 968d756463
commit 7da0a87124
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15 changed files with 113 additions and 38 deletions
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1
.gitignore vendored
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@ -71,4 +71,5 @@
/x86-generic_out/
/xcodebuild
/.vscode
/.idea
!webrtc/*

38
api/audio_codecs/audio_encoder.h
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@ -119,8 +119,17 @@ class AudioEncoder {
// Very OPUS-specific
struct Config {
// AKA ptime or frame size
// One of 10, 20, 40, 60, 120
uint32_t packet_size_ms = 20;
// One of 10, 20, 40, 60, 80, 100, 120
int32_t initial_packet_size_ms = 60;
int32_t min_packet_size_ms = 60;
int32_t max_packet_size_ms = 60;
// 500 to 192000
// Start at initial_bitrate_bps, and let the BWE and bitrate allocator
// move up to max_bitrate_bps or down to min_bitrate_bps.
int32_t initial_bitrate_bps = 32000;
int32_t min_bitrate_bps = 32000;
int32_t max_bitrate_bps = 32000;
// 1101 = OPUS_BANDWIDTH_NARROWBAND
// 1102 = OPUS_BADWIDTH_MEDIUMBAND
@ -128,20 +137,21 @@ class AudioEncoder {
// 1104 = OPUS_BANDWIDTH_SUPERWIDEBAND
// 1105 = OPUS_BANDWIDTH_FULLBAND
int32_t bandwidth = -1000; // OPUS_AUTO
// 500 to 192000
// Start at start_bitrate_bps, and let the BWE and bitrate allocator
// move up to max_bitrate_bps or down to min_bitrate_bps.
int32_t start_bitrate_bps = 40000;
int32_t min_bitrate_bps = 16000;
int32_t max_bitrate_bps = 40000;
// 0 (least complex) to 9 (most complex)
int32_t complexity = 9;
// 0 = CBR; 1 = VBR
int32_t enable_vbr = 0;
// 0 = disable; 1 = enable
int32_t enable_dtx = 0;
// 0 = disable; 1 = enable
int32_t enable_fec = 1;
// Adaptation method to use, 0 to disable
int32_t adaptation = 0;
// CBR is used by default
bool enable_cbr = true;
// DTX is enabled by default
bool enable_dtx = true;
// In-band FEC is enabled by default
bool enable_fec = true;
};
virtual ~AudioEncoder() = default;

2
audio/audio_receive_stream.cc
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@ -71,6 +71,8 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
config.rtcp_send_transport, event_log, config.rtp.local_ssrc,
config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
// RingRTC change to configure the RTCP report interval.
config.rtcp_report_interval_ms,
config.enable_non_sender_rtt, config.decoder_factory,
config.codec_pair_id, std::move(config.frame_decryptor),
config.crypto_options, std::move(config.frame_transformer));

7
audio/audio_send_stream.cc
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@ -922,9 +922,10 @@ void AudioSendStream::ConfigureEncoder(const webrtc::AudioEncoder::Config& confi
// This makes it so that if BWE changes cause us to change the bitrate,
// it doesn't actually change.
config_.min_bitrate_bps = config.min_bitrate_bps;
config_.max_bitrate_bps = config.max_bitrate_bps;
frame_length_range_ = {{TimeDelta::Millis(config.packet_size_ms), TimeDelta::Millis(config.packet_size_ms)}};
config_.min_bitrate_bps = config.initial_bitrate_bps;
config_.max_bitrate_bps = config.initial_bitrate_bps;
frame_length_range_ = {{TimeDelta::Millis(config.initial_packet_size_ms),
TimeDelta::Millis(config.initial_packet_size_ms)}};
channel_send_->CallEncoder([&](AudioEncoder* encoder) {
if (!encoder->Configure(config)) {
RTC_LOG(LS_INFO) << "Failed to configure audio send stream";

10
audio/channel_receive.cc
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@ -97,6 +97,8 @@ class ChannelReceive : public ChannelReceiveInterface,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
// RingRTC change to configure the RTCP report interval.
int rtcp_report_interval_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
@ -535,6 +537,8 @@ ChannelReceive::ChannelReceive(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
// RingRTC change to configure the RTCP report interval.
int rtcp_report_interval_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
@ -586,6 +590,8 @@ ChannelReceive::ChannelReceive(
configuration.local_media_ssrc = local_ssrc;
configuration.rtcp_packet_type_counter_observer = this;
configuration.non_sender_rtt_measurement = enable_non_sender_rtt;
// RingRTC change to configure the RTCP report interval.
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
if (frame_transformer)
InitFrameTransformerDelegate(std::move(frame_transformer));
@ -1108,6 +1114,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
// RingRTC change to configure the RTCP report interval.
int rtcp_report_interval_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
@ -1118,6 +1126,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
clock, neteq_factory, audio_device_module, rtcp_send_transport,
rtc_event_log, local_ssrc, remote_ssrc, jitter_buffer_max_packets,
jitter_buffer_fast_playout, jitter_buffer_min_delay_ms,
// RingRTC change to configure the RTCP report interval.
rtcp_report_interval_ms,
enable_non_sender_rtt, decoder_factory, codec_pair_id,
std::move(frame_decryptor), crypto_options, std::move(frame_transformer));
}

2
audio/channel_receive.h
View file

@ -181,6 +181,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
// RingRTC change to configure the RTCP report interval.
int rtcp_report_interval_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,

3
call/audio_receive_stream.h
View file

@ -130,6 +130,9 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
bool jitter_buffer_fast_accelerate = false;
int jitter_buffer_min_delay_ms = 0;
// RingRTC change to configure the RTCP report interval.
int rtcp_report_interval_ms = 5000;
// Identifier for an A/V synchronization group. Empty string to disable.
// TODO(pbos): Synchronize streams in a sync group, not just one video
// stream to one audio stream. Tracked by issue webrtc:4762.

8
media/engine/webrtc_voice_engine.cc
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@ -279,6 +279,8 @@ webrtc::AudioReceiveStreamInterface::Config BuildReceiveStreamConfig(
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_accelerate,
int jitter_buffer_min_delay_ms,
// RingRTC change to configure the RTCP report interval.
int rtcp_report_interval_ms,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
@ -297,6 +299,8 @@ webrtc::AudioReceiveStreamInterface::Config BuildReceiveStreamConfig(
config.jitter_buffer_max_packets = jitter_buffer_max_packets;
config.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate;
config.jitter_buffer_min_delay_ms = jitter_buffer_min_delay_ms;
// RingRTC change to configure the RTCP report interval.
config.rtcp_report_interval_ms = rtcp_report_interval_ms;
config.frame_decryptor = std::move(frame_decryptor);
config.crypto_options = crypto_options;
config.frame_transformer = std::move(frame_transformer);
@ -2307,7 +2311,9 @@ bool WebRtcVoiceReceiveChannel::AddRecvStream(const StreamParams& sp) {
sp.stream_ids(), recv_rtp_extensions_, this, engine()->decoder_factory_,
decoder_map_, codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
engine()->audio_jitter_buffer_fast_accelerate_,
engine()->audio_jitter_buffer_min_delay_ms_, unsignaled_frame_decryptor_,
engine()->audio_jitter_buffer_min_delay_ms_,
// RingRTC change to configure the RTCP report interval.
audio_config_.rtcp_report_interval_ms, unsignaled_frame_decryptor_,
crypto_options_, unsignaled_frame_transformer_);
recv_streams_.insert(std::make_pair(

32
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -994,15 +994,20 @@ bool AudioEncoderOpusImpl::Configure(const webrtc::AudioEncoder::Config& config)
// This sets next_frame_length_ms_ until the next time audio is sampled,
// and then it sets config_.frame_size_ms as well.
// It needs to be delayed to avoid a CHECK in Encode.
SetFrameLength(config.packet_size_ms);
SetFrameLength(config.initial_packet_size_ms);
// I don't think any of the below are necessary, but the above is, so we might as well set these.
config_.bitrate_bps = config.start_bitrate_bps;
config_.fec_enabled = config.enable_fec == 1;
config_.cbr_enabled = config.enable_vbr == 0;
config_.bitrate_bps = config.initial_bitrate_bps;
config_.fec_enabled = config.enable_fec;
config_.cbr_enabled = config.enable_cbr;
config_.complexity = config.complexity;
config_.low_rate_complexity = config_.low_rate_complexity;
config_.dtx_enabled = config.enable_dtx == 1;
config_.dtx_enabled = config.enable_dtx;
if (config.adaptation > 0) {
RTC_LOG(LS_WARNING) << "ringrtc_adapt!,audio,0," << config.initial_bitrate_bps
<< "," << config.initial_packet_size_ms;
}
if (WebRtcOpus_SetBandwidth(inst_, config.bandwidth) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to bandwidth=" << config.bandwidth;
@ -1010,11 +1015,11 @@ bool AudioEncoderOpusImpl::Configure(const webrtc::AudioEncoder::Config& config)
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to bandwidth=" << config.bandwidth;
if (WebRtcOpus_SetBitRate(inst_, config.start_bitrate_bps) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to bitrate_bps=" << config.start_bitrate_bps;
if (WebRtcOpus_SetBitRate(inst_, config.initial_bitrate_bps) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to bitrate_bps=" << config.initial_bitrate_bps;
return false;
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to bitrate_bps=" << config.start_bitrate_bps;
RTC_LOG(LS_INFO) << "Successfully configured OPUS to bitrate_bps=" << config.initial_bitrate_bps;
if (WebRtcOpus_SetComplexity(inst_, config.complexity) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to complexity=" << config.complexity;
@ -1022,7 +1027,7 @@ bool AudioEncoderOpusImpl::Configure(const webrtc::AudioEncoder::Config& config)
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to complexity=" << config.complexity;
if (config.enable_fec == 1) {
if (config.enable_fec) {
if (WebRtcOpus_EnableFec(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_fec=" << config.enable_fec;
return false;
@ -1035,7 +1040,7 @@ bool AudioEncoderOpusImpl::Configure(const webrtc::AudioEncoder::Config& config)
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to enable_fec=" << config.enable_fec;
if (config.enable_dtx == 1) {
if (config.enable_dtx) {
if (WebRtcOpus_EnableDtx(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_dtx=" << config.enable_dtx;
return false;
@ -1048,14 +1053,14 @@ bool AudioEncoderOpusImpl::Configure(const webrtc::AudioEncoder::Config& config)
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to enable_dtx=" << config.enable_dtx;
if (config.enable_vbr == 0) {
if (config.enable_cbr) {
if (WebRtcOpus_EnableCbr(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_vbr=" << config.enable_vbr;
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_cbr=" << config.enable_cbr;
return false;
}
} else {
if (WebRtcOpus_DisableCbr(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_vbr=" << config.enable_vbr;
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_cbr=" << config.enable_cbr;
return false;
}
}
@ -1063,5 +1068,4 @@ bool AudioEncoderOpusImpl::Configure(const webrtc::AudioEncoder::Config& config)
return true;
}
} // namespace webrtc

5
modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
View file

@ -269,4 +269,9 @@ AudioEncoderCopyRed::ReclaimContainedEncoders() {
return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
}
// RingRTC change to configure opus (the only codec we use RED with)
bool AudioEncoderCopyRed::Configure(const webrtc::AudioEncoder::Config& config) {
return speech_encoder_->Configure(config);
}
} // namespace webrtc

3
modules/audio_coding/codecs/red/audio_encoder_copy_red.h
View file

@ -84,6 +84,9 @@ class AudioEncoderCopyRed final : public AudioEncoder {
rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
override;
// RingRTC change to configure opus (the only codec we use RED with)
bool Configure(const webrtc::AudioEncoder::Config& config) override;
protected:
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,

1
ringrtc/rffi/api/peer_connection_factory.h
View file

@ -109,6 +109,7 @@ RUSTEXPORT webrtc::PeerConnectionInterface* Rust_createPeerConnection(
webrtc::rffi::PeerConnectionObserverRffi* observer_borrowed,
RffiPeerConnectionKind kind,
int audio_jitter_buffer_max_packets,
int audio_rtcp_report_interval_ms,
RffiIceServer ice_server,
webrtc::AudioTrackInterface* outgoing_audio_track_borrowed_rc,
webrtc::VideoTrackInterface* outgoing_video_track_borrowed_rc);

1
ringrtc/rffi/api/peer_connection_intf.h
View file

@ -100,6 +100,7 @@ RUSTEXPORT webrtc::SessionDescriptionInterface*
Rust_sessionDescriptionFromV4(bool offer,
const RffiConnectionParametersV4* v4_borrowed,
bool enable_tcc_audio,
bool enable_red_audio,
bool enable_vp9);
RUSTEXPORT void

36
ringrtc/rffi/src/peer_connection.cc
View file

@ -39,6 +39,7 @@ int TX_TIME_OFFSET_EXT_ID = 13;
// 101 used by connection.rs
int DATA_PT = 101;
int OPUS_PT = 102;
int OPUS_RED_PT = 105;
int VP8_PT = 108;
int VP8_RTX_PT = 118;
int VP9_PT = 109;
@ -246,12 +247,14 @@ RUSTEXPORT webrtc::SessionDescriptionInterface*
Rust_sessionDescriptionFromV4(bool offer,
const RffiConnectionParametersV4* v4_borrowed,
bool enable_tcc_audio,
bool enable_red_audio,
bool enable_vp9) {
// Major changes from the default WebRTC behavior:
// 1. We remove all codecs except Opus, VP8, and VP9
// 2. We remove all header extensions except for transport-cc, video orientation,
// and abs send time.
// 3. Opus CBR and DTX is enabled.
// 4. RED is enabled for audio.
// For some reason, WebRTC insists that the video SSRCs for one side don't
// overlap with SSRCs from the other side. To avoid potential problems, we'll give the
@ -289,13 +292,22 @@ Rust_sessionDescriptionFromV4(bool offer,
auto video = std::make_unique<cricket::VideoContentDescription>();
set_rtp_params(video.get());
// Turn on the RED "meta codec" for Opus redundancy.
auto opus_red = cricket::CreateAudioCodec(OPUS_RED_PT, cricket::kRedCodecName, 48000, 2);
opus_red.SetParam("", std::to_string(OPUS_PT) + "/" + std::to_string(OPUS_PT));
if (enable_red_audio) {
// Add RED before Opus to use it by default when sending.
audio->AddCodec(opus_red);
}
auto opus = cricket::CreateAudioCodec(OPUS_PT, cricket::kOpusCodecName, 48000, 2);
// These are the current defaults for WebRTC
// We set them explicitly to avoid having the defaults change on us.
opus.SetParam("stereo", "0"); // "1" would cause non-VOIP mode to be used
opus.SetParam("ptime", "20");
opus.SetParam("minptime", "10");
opus.SetParam("maxptime", "120");
opus.SetParam("ptime", "60");
opus.SetParam("minptime", "60");
opus.SetParam("maxptime", "60");
opus.SetParam("useinbandfec", "1");
// This is not a default. We enable this to help reduce bandwidth because we
// are using CBR.
@ -306,6 +318,11 @@ Rust_sessionDescriptionFromV4(bool offer,
opus.AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
audio->AddCodec(opus);
if (!enable_red_audio) {
// Add RED after Opus so that RED packets can at least be decoded properly if received.
audio->AddCodec(opus_red);
}
auto add_video_feedback_params = [] (cricket::VideoCodec* video_codec) {
video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir));
@ -477,9 +494,9 @@ CreateSessionDescriptionForGroupCall(bool local,
// These are the current defaults for WebRTC
// We set them explicitly to avoid having the defaults change on us.
opus.SetParam("stereo", "0"); // "1" would cause non-VOIP mode to be used
opus.SetParam("ptime", "20");
opus.SetParam("minptime", "10");
opus.SetParam("maxptime", "120");
opus.SetParam("ptime", "60");
opus.SetParam("minptime", "60");
opus.SetParam("maxptime", "60");
opus.SetParam("useinbandfec", "1");
// This is not a default. We enable this to help reduce bandwidth because we
// are using CBR.
@ -490,6 +507,13 @@ CreateSessionDescriptionForGroupCall(bool local,
opus.AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
audio->AddCodec(opus);
// Turn on the RED "meta codec" for Opus redundancy.
auto opus_red = cricket::CreateAudioCodec(OPUS_RED_PT, cricket::kRedCodecName, 48000, 2);
opus_red.SetParam("", std::to_string(OPUS_PT) + "/" + std::to_string(OPUS_PT));
// Add RED after Opus so that RED packets can at least be decoded properly if received.
audio->AddCodec(opus_red);
auto add_video_feedback_params = [] (cricket::VideoCodec* video_codec) {
video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir));

2
ringrtc/rffi/src/peer_connection_factory.cc
View file

@ -365,6 +365,7 @@ RUSTEXPORT PeerConnectionInterface* Rust_createPeerConnection(
PeerConnectionObserverRffi* observer_borrowed,
RffiPeerConnectionKind kind,
int audio_jitter_buffer_max_packets,
int audio_rtcp_report_interval_ms,
RffiIceServer ice_server,
webrtc::AudioTrackInterface* outgoing_audio_track_borrowed_rc,
webrtc::VideoTrackInterface* outgoing_video_track_borrowed_rc) {
@ -380,6 +381,7 @@ RUSTEXPORT PeerConnectionInterface* Rust_createPeerConnection(
config.tcp_candidate_policy = PeerConnectionInterface::kTcpCandidatePolicyEnabled;
}
config.audio_jitter_buffer_max_packets = audio_jitter_buffer_max_packets;
config.set_audio_rtcp_report_interval_ms(audio_rtcp_report_interval_ms);
config.sdp_semantics = SdpSemantics::kPlanB_DEPRECATED;
if (ice_server.urls_size > 0) {
webrtc::PeerConnectionInterface::IceServer rtc_ice_server;