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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc | 128 |
1 files changed, 128 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc new file mode 100644 index 0000000000..cff9685548 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc @@ -0,0 +1,128 @@ +/* + * Copyright (c) 2015 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_decoder_opus.h" + +#include <memory> +#include <utility> + +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels, + int sample_rate_hz) + : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} { + RTC_DCHECK(num_channels == 1 || num_channels == 2); + RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000); + const int error = + WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_); + RTC_DCHECK(error == 0); + WebRtcOpus_DecoderInit(dec_state_); +} + +AudioDecoderOpusImpl::~AudioDecoderOpusImpl() { + WebRtcOpus_DecoderFree(dec_state_); +} + +std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload( + rtc::Buffer&& payload, + uint32_t timestamp) { + std::vector<ParseResult> results; + + if (PacketHasFec(payload.data(), payload.size())) { + const int duration = + PacketDurationRedundant(payload.data(), payload.size()); + RTC_DCHECK_GE(duration, 0); + rtc::Buffer payload_copy(payload.data(), payload.size()); + std::unique_ptr<EncodedAudioFrame> fec_frame( + new OpusFrame(this, std::move(payload_copy), false)); + results.emplace_back(timestamp - duration, 1, std::move(fec_frame)); + } + std::unique_ptr<EncodedAudioFrame> frame( + new OpusFrame(this, std::move(payload), true)); + results.emplace_back(timestamp, 0, std::move(frame)); + return results; +} + +int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_); + int16_t temp_type = 1; // Default is speech. + int ret = + WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type); + if (ret > 0) + ret *= static_cast<int>(channels_); // Return total number of samples. + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + if (!PacketHasFec(encoded, encoded_len)) { + // This packet is a RED packet. + return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, + speech_type); + } + + RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_); + int16_t temp_type = 1; // Default is speech. + int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded, + &temp_type); + if (ret > 0) + ret *= static_cast<int>(channels_); // Return total number of samples. + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +void AudioDecoderOpusImpl::Reset() { + WebRtcOpus_DecoderInit(dec_state_); +} + +int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded, + size_t encoded_len) const { + return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len); +} + +int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const { + if (!PacketHasFec(encoded, encoded_len)) { + // This packet is a RED packet. + return PacketDuration(encoded, encoded_len); + } + + return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_); +} + +bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded, + size_t encoded_len) const { + int fec; + fec = WebRtcOpus_PacketHasFec(encoded, encoded_len); + return (fec == 1); +} + +int AudioDecoderOpusImpl::SampleRateHz() const { + return sample_rate_hz_; +} + +size_t AudioDecoderOpusImpl::Channels() const { + return channels_; +} + +} // namespace webrtc |