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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_codecs/audio_encoder.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/audio_codecs/audio_encoder.cc')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/audio_encoder.cc | 114 |
1 files changed, 114 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc new file mode 100644 index 0000000000..31bb8739f7 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc @@ -0,0 +1,114 @@ +/* + * 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 "api/audio_codecs/audio_encoder.h" + +#include "rtc_base/checks.h" +#include "rtc_base/trace_event.h" + +namespace webrtc { + +ANAStats::ANAStats() = default; +ANAStats::~ANAStats() = default; +ANAStats::ANAStats(const ANAStats&) = default; + +AudioEncoder::EncodedInfo::EncodedInfo() = default; +AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default; +AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default; +AudioEncoder::EncodedInfo::~EncodedInfo() = default; +AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=( + const EncodedInfo&) = default; +AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) = + default; + +int AudioEncoder::RtpTimestampRateHz() const { + return SampleRateHz(); +} + +AudioEncoder::EncodedInfo AudioEncoder::Encode( + uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) { + TRACE_EVENT0("webrtc", "AudioEncoder::Encode"); + RTC_CHECK_EQ(audio.size(), + static_cast<size_t>(NumChannels() * SampleRateHz() / 100)); + + const size_t old_size = encoded->size(); + EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded); + RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes); + return info; +} + +bool AudioEncoder::SetFec(bool enable) { + return !enable; +} + +bool AudioEncoder::SetDtx(bool enable) { + return !enable; +} + +bool AudioEncoder::GetDtx() const { + return false; +} + +bool AudioEncoder::SetApplication(Application application) { + return false; +} + +void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {} + +void AudioEncoder::SetTargetBitrate(int target_bps) {} + +rtc::ArrayView<std::unique_ptr<AudioEncoder>> +AudioEncoder::ReclaimContainedEncoders() { + return nullptr; +} + +bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string, + RtcEventLog* event_log) { + return false; +} + +void AudioEncoder::DisableAudioNetworkAdaptor() {} + +void AudioEncoder::OnReceivedUplinkPacketLossFraction( + float uplink_packet_loss_fraction) {} + +void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction( + float uplink_recoverable_packet_loss_fraction) { + RTC_DCHECK_NOTREACHED(); +} + +void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) { + OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt); +} + +void AudioEncoder::OnReceivedUplinkBandwidth( + int target_audio_bitrate_bps, + absl::optional<int64_t> bwe_period_ms) {} + +void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) { + OnReceivedUplinkBandwidth(update.target_bitrate.bps(), + update.bwe_period.ms()); +} + +void AudioEncoder::OnReceivedRtt(int rtt_ms) {} + +void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {} + +void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms, + int max_frame_length_ms) {} + +ANAStats AudioEncoder::GetANAStats() const { + return ANAStats(); +} + +constexpr int AudioEncoder::kMaxNumberOfChannels; +} // namespace webrtc |