diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc')
| -rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc | 366 |
1 files changed, 366 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc new file mode 100644 index 0000000000..38a11c123d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc @@ -0,0 +1,366 @@ +/* + * Copyright (c) 2019 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. + */ + +/* + * LEFT TO DO: + * - WRITE TESTS for the stuff in this file. + * - Check the creation, maybe make it safer by returning an empty optional or + * unique_ptr. --- It looks OK, but RecreateEncoderInstance can perhaps crash + * on a valid config. Can run it in the fuzzer for some time. Should prbl also + * fuzz the config. + */ + +#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h" + +#include <algorithm> +#include <memory> +#include <string> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" +#include "rtc_base/arraysize.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/string_to_number.h" + +namespace webrtc { + +namespace { + +// Recommended bitrates for one channel: +// 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 multiplies these values by the number of channels. +constexpr int kOpusBitrateNbBps = 12000; +constexpr int kOpusBitrateWbBps = 20000; +constexpr int kOpusBitrateFbBps = 32000; + +constexpr int kDefaultMaxPlaybackRate = 48000; +// These two lists must be sorted from low to high +#if WEBRTC_OPUS_SUPPORT_120MS_PTIME +constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120}; +#else +constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60}; +#endif + +int GetBitrateBps(const AudioEncoderMultiChannelOpusConfig& config) { + RTC_DCHECK(config.IsOk()); + return config.bitrate_bps; +} +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.has_value()) { + // 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; +} + +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, AudioEncoderMultiChannelOpusConfig::kMinBitrateBps); + 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, + absl::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) { + RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate + << " clamped to " << chosen_bitrate; + } + return chosen_bitrate; + } + RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param + << "\" replaced by default bitrate " << default_bitrate; + } + + return default_bitrate; +} + +} // namespace + +std::unique_ptr<AudioEncoder> +AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder( + const AudioEncoderMultiChannelOpusConfig& config, + int payload_type) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return std::make_unique<AudioEncoderMultiChannelOpusImpl>(config, + payload_type); +} + +AudioEncoderMultiChannelOpusImpl::AudioEncoderMultiChannelOpusImpl( + const AudioEncoderMultiChannelOpusConfig& config, + int payload_type) + : payload_type_(payload_type), inst_(nullptr) { + RTC_DCHECK(0 <= payload_type && payload_type <= 127); + + RTC_CHECK(RecreateEncoderInstance(config)); +} + +AudioEncoderMultiChannelOpusImpl::~AudioEncoderMultiChannelOpusImpl() { + RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_)); +} + +size_t AudioEncoderMultiChannelOpusImpl::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; +} + +void AudioEncoderMultiChannelOpusImpl::Reset() { + RTC_CHECK(RecreateEncoderInstance(config_)); +} + +absl::optional<std::pair<TimeDelta, TimeDelta>> +AudioEncoderMultiChannelOpusImpl::GetFrameLengthRange() const { + return {{TimeDelta::Millis(config_.frame_size_ms), + TimeDelta::Millis(config_.frame_size_ms)}}; +} + +// 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 AudioEncoderMultiChannelOpusImpl::RecreateEncoderInstance( + const AudioEncoderMultiChannelOpusConfig& 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_MultistreamEncoderCreate( + &inst_, config.num_channels, + config.application == + AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip + ? 0 + : 1, + config.num_streams, config.coupled_streams, + config.channel_mapping.data())); + const int bitrate = GetBitrateBps(config); + RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate)); + RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps."; + if (config.fec_enabled) { + RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_)); + RTC_LOG(LS_VERBOSE) << "Opus enable FEC"; + } else { + RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_)); + RTC_LOG(LS_VERBOSE) << "Opus disable FEC"; + } + RTC_CHECK_EQ( + 0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz)); + RTC_LOG(LS_VERBOSE) << "Set Opus playback rate to " + << config.max_playback_rate_hz << " hz."; + + // Use the DEFAULT complexity. + RTC_CHECK_EQ( + 0, WebRtcOpus_SetComplexity(inst_, AudioEncoderOpusConfig().complexity)); + RTC_LOG(LS_VERBOSE) << "Set Opus coding complexity to " + << AudioEncoderOpusConfig().complexity; + + if (config.dtx_enabled) { + RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_)); + RTC_LOG(LS_VERBOSE) << "Opus enable DTX"; + } else { + RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_)); + RTC_LOG(LS_VERBOSE) << "Opus disable DTX"; + } + + if (config.cbr_enabled) { + RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_)); + RTC_LOG(LS_VERBOSE) << "Opus enable CBR"; + } else { + RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_)); + RTC_LOG(LS_VERBOSE) << "Opus disable CBR"; + } + num_channels_to_encode_ = NumChannels(); + next_frame_length_ms_ = config_.frame_size_ms; + RTC_LOG(LS_VERBOSE) << "Set Opus frame length to " << config_.frame_size_ms + << " ms"; + return true; +} + +absl::optional<AudioEncoderMultiChannelOpusConfig> +AudioEncoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) { + if (!absl::EqualsIgnoreCase(format.name, "multiopus") || + format.clockrate_hz != 48000) { + return absl::nullopt; + } + + AudioEncoderMultiChannelOpusConfig config; + config.num_channels = format.num_channels; + 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 = + CalculateBitrate(config.max_playback_rate_hz, config.num_channels, + GetFormatParameter(format, "maxaveragebitrate")); + config.application = + config.num_channels == 1 + ? AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip + : AudioEncoderMultiChannelOpusConfig::ApplicationMode::kAudio; + + config.supported_frame_lengths_ms.clear(); + std::copy(std::begin(kOpusSupportedFrameLengths), + std::end(kOpusSupportedFrameLengths), + std::back_inserter(config.supported_frame_lengths_ms)); + + auto num_streams = GetFormatParameter<int>(format, "num_streams"); + if (!num_streams.has_value()) { + return absl::nullopt; + } + config.num_streams = *num_streams; + + auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams"); + if (!coupled_streams.has_value()) { + return absl::nullopt; + } + config.coupled_streams = *coupled_streams; + + auto channel_mapping = + GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping"); + if (!channel_mapping.has_value()) { + return absl::nullopt; + } + config.channel_mapping = *channel_mapping; + + if (!config.IsOk()) { + return absl::nullopt; + } + return config; +} + +AudioCodecInfo AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder( + const AudioEncoderMultiChannelOpusConfig& 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 = false; + return info; +} + +size_t AudioEncoderMultiChannelOpusImpl::Num10msFramesPerPacket() const { + return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10)); +} +size_t AudioEncoderMultiChannelOpusImpl::SamplesPer10msFrame() const { + return rtc::CheckedDivExact(48000, 100) * config_.num_channels; +} +int AudioEncoderMultiChannelOpusImpl::SampleRateHz() const { + return 48000; +} +size_t AudioEncoderMultiChannelOpusImpl::NumChannels() const { + return config_.num_channels; +} +size_t AudioEncoderMultiChannelOpusImpl::Num10MsFramesInNextPacket() const { + return Num10msFramesPerPacket(); +} +size_t AudioEncoderMultiChannelOpusImpl::Max10MsFramesInAPacket() const { + return Num10msFramesPerPacket(); +} +int AudioEncoderMultiChannelOpusImpl::GetTargetBitrate() const { + return GetBitrateBps(config_); +} + +AudioEncoder::EncodedInfo AudioEncoderMultiChannelOpusImpl::EncodeImpl( + uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) { + 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 = true; + info.encoder_type = CodecType::kOther; + + return info; +} + +} // namespace webrtc |