blob: 0a412b106436efce3d6a56090b1ed834b67fa8f8 [file] [log] [blame]
/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include <algorithm>
#include <iterator>
#include <utility>
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/exp_filter.h"
#include "rtc_base/protobuf_utils.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
#include "rtc_base/safe_minmax.h"
#include "rtc_base/string_to_number.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
// Codec parameters for Opus.
// draft-spittka-payload-rtp-opus-03
// Recommended bitrates:
// 8-12 kb/s for NB speech,
// 16-20 kb/s for WB speech,
// 28-40 kb/s for FB speech,
// 48-64 kb/s for FB mono music, and
// 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.
constexpr int kOpusBitrateNbBps = 12000;
constexpr int kOpusBitrateWbBps = 20000;
constexpr int kOpusBitrateFbBps = 32000;
constexpr int kSampleRateHz = 48000;
constexpr int kDefaultMaxPlaybackRate = 48000;
// These two lists must be sorted from low to high
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
constexpr int kANASupportedFrameLengths[] = {20, 60, 120};
constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
#else
constexpr int kANASupportedFrameLengths[] = {20, 60};
constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
#endif
// PacketLossFractionSmoother uses an exponential filter with a time constant
// of -1.0 / ln(0.9999) = 10000 ms.
constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f;
// Optimize the loss rate to configure Opus. Basically, optimized loss rate is
// the input loss rate rounded down to various levels, because a robustly good
// audio quality is achieved by lowering the packet loss down.
// Additionally, to prevent toggling, margins are used, i.e., when jumping to
// a loss rate from below, a higher threshold is used than jumping to the same
// level from above.
float OptimizePacketLossRate(float new_loss_rate, float old_loss_rate) {
RTC_DCHECK_GE(new_loss_rate, 0.0f);
RTC_DCHECK_LE(new_loss_rate, 1.0f);
RTC_DCHECK_GE(old_loss_rate, 0.0f);
RTC_DCHECK_LE(old_loss_rate, 1.0f);
constexpr float kPacketLossRate20 = 0.20f;
constexpr float kPacketLossRate10 = 0.10f;
constexpr float kPacketLossRate5 = 0.05f;
constexpr float kPacketLossRate1 = 0.01f;
constexpr float kLossRate20Margin = 0.02f;
constexpr float kLossRate10Margin = 0.01f;
constexpr float kLossRate5Margin = 0.01f;
if (new_loss_rate >=
kPacketLossRate20 +
kLossRate20Margin *
(kPacketLossRate20 - old_loss_rate > 0 ? 1 : -1)) {
return kPacketLossRate20;
} else if (new_loss_rate >=
kPacketLossRate10 +
kLossRate10Margin *
(kPacketLossRate10 - old_loss_rate > 0 ? 1 : -1)) {
return kPacketLossRate10;
} else if (new_loss_rate >=
kPacketLossRate5 +
kLossRate5Margin *
(kPacketLossRate5 - old_loss_rate > 0 ? 1 : -1)) {
return kPacketLossRate5;
} else if (new_loss_rate >= kPacketLossRate1) {
return kPacketLossRate1;
} else {
return 0.0f;
}
}
rtc::Optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
const std::string& param) {
auto it = format.parameters.find(param);
return (it == format.parameters.end())
? rtc::Optional<std::string>()
: rtc::Optional<std::string>(it->second);
}
template <typename T>
rtc::Optional<T> GetFormatParameter(const SdpAudioFormat& format,
const std::string& param) {
return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
}
int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
const int bitrate = [&] {
if (max_playback_rate <= 8000) {
return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
} else if (max_playback_rate <= 16000) {
return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
} else {
return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
}
}();
RTC_DCHECK_GE(bitrate, AudioEncoderOpusConfig::kMinBitrateBps);
RTC_DCHECK_LE(bitrate, AudioEncoderOpusConfig::kMaxBitrateBps);
return bitrate;
}
// Get the maxaveragebitrate parameter in string-form, so we can properly figure
// out how invalid it is and accurately log invalid values.
int CalculateBitrate(int max_playback_rate_hz,
size_t num_channels,
rtc::Optional<std::string> bitrate_param) {
const int default_bitrate =
CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
if (bitrate_param) {
const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
if (bitrate) {
const int chosen_bitrate =
std::max(AudioEncoderOpusConfig::kMinBitrateBps,
std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
if (bitrate != chosen_bitrate) {
LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
<< " clamped to " << chosen_bitrate;
}
return chosen_bitrate;
}
LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
<< "\" replaced by default bitrate " << default_bitrate;
}
return default_bitrate;
}
int GetChannelCount(const SdpAudioFormat& format) {
const auto param = GetFormatParameter(format, "stereo");
if (param == "1") {
return 2;
} else {
return 1;
}
}
int GetMaxPlaybackRate(const SdpAudioFormat& format) {
const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
if (param && *param >= 8000) {
return std::min(*param, kDefaultMaxPlaybackRate);
}
return kDefaultMaxPlaybackRate;
}
int GetFrameSizeMs(const SdpAudioFormat& format) {
const auto ptime = GetFormatParameter<int>(format, "ptime");
if (ptime) {
// Pick the next highest supported frame length from
// kOpusSupportedFrameLengths.
for (const int supported_frame_length : kOpusSupportedFrameLengths) {
if (supported_frame_length >= *ptime) {
return supported_frame_length;
}
}
// If none was found, return the largest supported frame length.
return *(std::end(kOpusSupportedFrameLengths) - 1);
}
return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
}
void FindSupportedFrameLengths(int min_frame_length_ms,
int max_frame_length_ms,
std::vector<int>* out) {
out->clear();
std::copy_if(std::begin(kANASupportedFrameLengths),
std::end(kANASupportedFrameLengths), std::back_inserter(*out),
[&](int frame_length_ms) {
return frame_length_ms >= min_frame_length_ms &&
frame_length_ms <= max_frame_length_ms;
});
RTC_DCHECK(std::is_sorted(out->begin(), out->end()));
}
int GetBitrateBps(const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
return *config.bitrate_bps;
}
} // namespace
void AudioEncoderOpusImpl::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
const SdpAudioFormat fmt = {
"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
AudioCodecInfo AudioEncoderOpusImpl::QueryAudioEncoder(
const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
AudioCodecInfo info(48000, config.num_channels, *config.bitrate_bps,
AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps);
info.allow_comfort_noise = false;
info.supports_network_adaption = true;
return info;
}
std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
const AudioEncoderOpusConfig& config,
int payload_type) {
RTC_DCHECK(config.IsOk());
return rtc::MakeUnique<AudioEncoderOpusImpl>(config, payload_type);
}
rtc::Optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
format.clockrate_hz == 48000 && format.num_channels == 2) {
const size_t num_channels = GetChannelCount(format);
const int bitrate =
CalculateBitrate(GetMaxPlaybackRate(format), num_channels,
GetFormatParameter(format, "maxaveragebitrate"));
AudioCodecInfo info(48000, num_channels, bitrate,
AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps);
info.allow_comfort_noise = false;
info.supports_network_adaption = true;
return rtc::Optional<AudioCodecInfo>(info);
}
return rtc::Optional<AudioCodecInfo>();
}
AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
const CodecInst& codec_inst) {
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
config.num_channels = codec_inst.channels;
config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
return config;
}
rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
format.clockrate_hz != 48000 || format.num_channels != 2) {
return rtc::Optional<AudioEncoderOpusConfig>();
}
AudioEncoderOpusConfig config;
config.num_channels = GetChannelCount(format);
config.frame_size_ms = GetFrameSizeMs(format);
config.max_playback_rate_hz = GetMaxPlaybackRate(format);
config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
config.bitrate_bps = rtc::Optional<int>(
CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
GetFormatParameter(format, "maxaveragebitrate")));
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
constexpr int kMinANAFrameLength = kANASupportedFrameLengths[0];
constexpr int kMaxANAFrameLength =
kANASupportedFrameLengths[arraysize(kANASupportedFrameLengths) - 1];
// For now, minptime and maxptime are only used with ANA. If ptime is outside
// of this range, it will get adjusted once ANA takes hold. Ideally, we'd know
// if ANA was to be used when setting up the config, and adjust accordingly.
const int min_frame_length_ms =
GetFormatParameter<int>(format, "minptime").value_or(kMinANAFrameLength);
const int max_frame_length_ms =
GetFormatParameter<int>(format, "maxptime").value_or(kMaxANAFrameLength);
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config.supported_frame_lengths_ms);
RTC_DCHECK(config.IsOk());
return rtc::Optional<AudioEncoderOpusConfig>(config);
}
rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
const int bitrate_bps = GetBitrateBps(config);
if (bitrate_bps >= config.complexity_threshold_bps -
config.complexity_threshold_window_bps &&
bitrate_bps <= config.complexity_threshold_bps +
config.complexity_threshold_window_bps) {
// Within the hysteresis window; make no change.
return rtc::Optional<int>();
} else {
return rtc::Optional<int>(bitrate_bps <= config.complexity_threshold_bps
? config.low_rate_complexity
: config.complexity);
}
}
class AudioEncoderOpusImpl::PacketLossFractionSmoother {
public:
explicit PacketLossFractionSmoother()
: last_sample_time_ms_(rtc::TimeMillis()),
smoother_(kAlphaForPacketLossFractionSmoother) {}
// Gets the smoothed packet loss fraction.
float GetAverage() const {
float value = smoother_.filtered();
return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value;
}
// Add new observation to the packet loss fraction smoother.
void AddSample(float packet_loss_fraction) {
int64_t now_ms = rtc::TimeMillis();
smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_),
packet_loss_fraction);
last_sample_time_ms_ = now_ms;
}
private:
int64_t last_sample_time_ms_;
// An exponential filter is used to smooth the packet loss fraction.
rtc::ExpFilter smoother_;
};
AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config,
int payload_type)
: AudioEncoderOpusImpl(
config,
payload_type,
[this](const ProtoString& config_string, RtcEventLog* event_log) {
return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
},
// We choose 5sec as initial time constant due to empirical data.
rtc::MakeUnique<SmoothingFilterImpl>(5000)) {}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(
const AudioEncoderOpusConfig& config,
int payload_type,
const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
std::unique_ptr<SmoothingFilter> bitrate_smoother)
: payload_type_(payload_type),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
packet_loss_rate_(0.0),
inst_(nullptr),
packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
audio_network_adaptor_creator_(audio_network_adaptor_creator),
bitrate_smoother_(std::move(bitrate_smoother)) {
RTC_DCHECK(0 <= payload_type && payload_type <= 127);
// Sanity check of the redundant payload type field that we want to get rid
// of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
RTC_CHECK(RecreateEncoderInstance(config));
}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(const CodecInst& codec_inst)
: AudioEncoderOpusImpl(CreateConfig(codec_inst), codec_inst.pltype) {}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(int payload_type,
const SdpAudioFormat& format)
: AudioEncoderOpusImpl(*SdpToConfig(format), payload_type) {}
AudioEncoderOpusImpl::~AudioEncoderOpusImpl() {
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
}
int AudioEncoderOpusImpl::SampleRateHz() const {
return kSampleRateHz;
}
size_t AudioEncoderOpusImpl::NumChannels() const {
return config_.num_channels;
}
size_t AudioEncoderOpusImpl::Num10MsFramesInNextPacket() const {
return Num10msFramesPerPacket();
}
size_t AudioEncoderOpusImpl::Max10MsFramesInAPacket() const {
return Num10msFramesPerPacket();
}
int AudioEncoderOpusImpl::GetTargetBitrate() const {
return GetBitrateBps(config_);
}
void AudioEncoderOpusImpl::Reset() {
RTC_CHECK(RecreateEncoderInstance(config_));
}
bool AudioEncoderOpusImpl::SetFec(bool enable) {
if (enable) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
}
config_.fec_enabled = enable;
return true;
}
bool AudioEncoderOpusImpl::SetDtx(bool enable) {
if (enable) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
}
config_.dtx_enabled = enable;
return true;
}
bool AudioEncoderOpusImpl::GetDtx() const {
return config_.dtx_enabled;
}
bool AudioEncoderOpusImpl::SetApplication(Application application) {
auto conf = config_;
switch (application) {
case Application::kSpeech:
conf.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
break;
case Application::kAudio:
conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
break;
}
return RecreateEncoderInstance(conf);
}
void AudioEncoderOpusImpl::SetMaxPlaybackRate(int frequency_hz) {
auto conf = config_;
conf.max_playback_rate_hz = frequency_hz;
RTC_CHECK(RecreateEncoderInstance(conf));
}
bool AudioEncoderOpusImpl::EnableAudioNetworkAdaptor(
const std::string& config_string,
RtcEventLog* event_log) {
audio_network_adaptor_ =
audio_network_adaptor_creator_(config_string, event_log);
return audio_network_adaptor_.get() != nullptr;
}
void AudioEncoderOpusImpl::DisableAudioNetworkAdaptor() {
audio_network_adaptor_.reset(nullptr);
}
void AudioEncoderOpusImpl::OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {
if (!audio_network_adaptor_) {
packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
return SetProjectedPacketLossRate(average_fraction_loss);
}
audio_network_adaptor_->SetUplinkPacketLossFraction(
uplink_packet_loss_fraction);
ApplyAudioNetworkAdaptor();
}
void AudioEncoderOpusImpl::OnReceivedUplinkRecoverablePacketLossFraction(
float uplink_recoverable_packet_loss_fraction) {
if (!audio_network_adaptor_)
return;
audio_network_adaptor_->SetUplinkRecoverablePacketLossFraction(
uplink_recoverable_packet_loss_fraction);
ApplyAudioNetworkAdaptor();
}
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> bwe_period_ms) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
// We give smoothed bitrate allocation to audio network adaptor as
// the uplink bandwidth.
// The BWE spikes should not affect the bitrate smoother more than 25%.
// To simplify the calculations we use a step response as input signal.
// The step response of an exponential filter is
// u(t) = 1 - e^(-t / time_constant).
// In order to limit the affect of a BWE spike within 25% of its value
// before
// the next BWE update, we would choose a time constant that fulfills
// 1 - e^(-bwe_period_ms / time_constant) < 0.25
// Then 4 * bwe_period_ms is a good choice.
if (bwe_period_ms)
bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
bitrate_smoother_->AddSample(target_audio_bitrate_bps);
ApplyAudioNetworkAdaptor();
} else if (send_side_bwe_with_overhead_) {
if (!overhead_bytes_per_packet_) {
LOG(LS_INFO)
<< "AudioEncoderOpusImpl: Overhead unknown, target audio bitrate "
<< target_audio_bitrate_bps << " bps is ignored.";
return;
}
const int overhead_bps = static_cast<int>(
*overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
SetTargetBitrate(
std::min(AudioEncoderOpusConfig::kMaxBitrateBps,
std::max(AudioEncoderOpusConfig::kMinBitrateBps,
target_audio_bitrate_bps - overhead_bps)));
} else {
SetTargetBitrate(target_audio_bitrate_bps);
}
}
void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
if (!audio_network_adaptor_)
return;
audio_network_adaptor_->SetRtt(rtt_ms);
ApplyAudioNetworkAdaptor();
}
void AudioEncoderOpusImpl::OnReceivedOverhead(
size_t overhead_bytes_per_packet) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
ApplyAudioNetworkAdaptor();
} else {
overhead_bytes_per_packet_ =
rtc::Optional<size_t>(overhead_bytes_per_packet);
}
}
void AudioEncoderOpusImpl::SetReceiverFrameLengthRange(
int min_frame_length_ms,
int max_frame_length_ms) {
// Ensure that |SetReceiverFrameLengthRange| is called before
// |EnableAudioNetworkAdaptor|, otherwise we need to recreate
// |audio_network_adaptor_|, which is not a needed use case.
RTC_DCHECK(!audio_network_adaptor_);
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config_.supported_frame_lengths_ms);
}
AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) {
MaybeUpdateUplinkBandwidth();
if (input_buffer_.empty())
first_timestamp_in_buffer_ = rtp_timestamp;
input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
if (input_buffer_.size() <
(Num10msFramesPerPacket() * SamplesPer10msFrame())) {
return EncodedInfo();
}
RTC_CHECK_EQ(input_buffer_.size(),
Num10msFramesPerPacket() * SamplesPer10msFrame());
const size_t max_encoded_bytes = SufficientOutputBufferSize();
EncodedInfo info;
info.encoded_bytes =
encoded->AppendData(
max_encoded_bytes, [&] (rtc::ArrayView<uint8_t> encoded) {
int status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
rtc::CheckedDivExact(input_buffer_.size(),
config_.num_channels),
rtc::saturated_cast<int16_t>(max_encoded_bytes),
encoded.data());
RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
return static_cast<size_t>(status);
});
input_buffer_.clear();
// Will use new packet size for next encoding.
config_.frame_size_ms = next_frame_length_ms_;
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.send_even_if_empty = true; // Allows Opus to send empty packets.
info.speech = (info.encoded_bytes > 0);
info.encoder_type = CodecType::kOpus;
return info;
}
size_t AudioEncoderOpusImpl::Num10msFramesPerPacket() const {
return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
}
size_t AudioEncoderOpusImpl::SamplesPer10msFrame() const {
return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
}
size_t AudioEncoderOpusImpl::SufficientOutputBufferSize() const {
// Calculate the number of bytes we expect the encoder to produce,
// then multiply by two to give a wide margin for error.
const size_t bytes_per_millisecond =
static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
const size_t approx_encoded_bytes =
Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
return 2 * approx_encoded_bytes;
}
// If the given config is OK, recreate the Opus encoder instance with those
// settings, save the config, and return true. Otherwise, do nothing and return
// false.
bool AudioEncoderOpusImpl::RecreateEncoderInstance(
const AudioEncoderOpusConfig& config) {
if (!config.IsOk())
return false;
config_ = config;
if (inst_)
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
input_buffer_.clear();
input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(
&inst_, config.num_channels,
config.application ==
AudioEncoderOpusConfig::ApplicationMode::kVoip
? 0
: 1));
RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config)));
if (config.fec_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
}
RTC_CHECK_EQ(
0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
// Use the default complexity if the start bitrate is within the hysteresis
// window.
complexity_ = GetNewComplexity(config).value_or(config.complexity);
RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
if (config.dtx_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
}
RTC_CHECK_EQ(0,
WebRtcOpus_SetPacketLossRate(
inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
if (config.cbr_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
}
num_channels_to_encode_ = NumChannels();
next_frame_length_ms_ = config_.frame_size_ms;
return true;
}
void AudioEncoderOpusImpl::SetFrameLength(int frame_length_ms) {
next_frame_length_ms_ = frame_length_ms;
}
void AudioEncoderOpusImpl::SetNumChannelsToEncode(
size_t num_channels_to_encode) {
RTC_DCHECK_GT(num_channels_to_encode, 0);
RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
if (num_channels_to_encode_ == num_channels_to_encode)
return;
RTC_CHECK_EQ(0, WebRtcOpus_SetForceChannels(inst_, num_channels_to_encode));
num_channels_to_encode_ = num_channels_to_encode;
}
void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
float opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_);
if (packet_loss_rate_ != opt_loss_rate) {
packet_loss_rate_ = opt_loss_rate;
RTC_CHECK_EQ(
0, WebRtcOpus_SetPacketLossRate(
inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
}
}
void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
config_.bitrate_bps = rtc::Optional<int>(rtc::SafeClamp<int>(
bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps));
RTC_DCHECK(config_.IsOk());
RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config_)));
const auto new_complexity = GetNewComplexity(config_);
if (new_complexity && complexity_ != *new_complexity) {
complexity_ = *new_complexity;
RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
}
}
void AudioEncoderOpusImpl::ApplyAudioNetworkAdaptor() {
auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
if (config.bitrate_bps)
SetTargetBitrate(*config.bitrate_bps);
if (config.frame_length_ms)
SetFrameLength(*config.frame_length_ms);
if (config.enable_fec)
SetFec(*config.enable_fec);
if (config.uplink_packet_loss_fraction)
SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
if (config.enable_dtx)
SetDtx(*config.enable_dtx);
if (config.num_channels)
SetNumChannelsToEncode(*config.num_channels);
}
std::unique_ptr<AudioNetworkAdaptor>
AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
const ProtoString& config_string,
RtcEventLog* event_log) const {
AudioNetworkAdaptorImpl::Config config;
config.event_log = event_log;
return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
config, ControllerManagerImpl::Create(
config_string, NumChannels(), supported_frame_lengths_ms(),
AudioEncoderOpusConfig::kMinBitrateBps,
num_channels_to_encode_, next_frame_length_ms_,
GetTargetBitrate(), config_.fec_enabled, GetDtx())));
}
void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
if (audio_network_adaptor_) {
int64_t now_ms = rtc::TimeMillis();
if (!bitrate_smoother_last_update_time_ ||
now_ms - *bitrate_smoother_last_update_time_ >=
config_.uplink_bandwidth_update_interval_ms) {
rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
}
}
}
ANAStats AudioEncoderOpusImpl::GetANAStats() const {
if (audio_network_adaptor_) {
return audio_network_adaptor_->GetStats();
}
return ANAStats();
}
} // namespace webrtc