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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h | 71 |
1 files changed, 71 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h new file mode 100644 index 0000000000..a932aa8b7d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h @@ -0,0 +1,71 @@ +/* + * 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. + */ + +#ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ +#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ + +#include <memory> +#include <utility> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/g722/audio_encoder_g722_config.h" +#include "api/units/time_delta.h" +#include "modules/audio_coding/codecs/g722/g722_interface.h" +#include "rtc_base/buffer.h" + +namespace webrtc { + +class AudioEncoderG722Impl final : public AudioEncoder { + public: + AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type); + ~AudioEncoderG722Impl() override; + + AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete; + AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete; + + int SampleRateHz() const override; + size_t NumChannels() const override; + int RtpTimestampRateHz() const override; + size_t Num10MsFramesInNextPacket() const override; + size_t Max10MsFramesInAPacket() const override; + int GetTargetBitrate() const override; + void Reset() override; + absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange() + const override; + + protected: + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) override; + + private: + // The encoder state for one channel. + struct EncoderState { + G722EncInst* encoder; + std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding. + rtc::Buffer encoded_buffer; // Already encoded. + EncoderState(); + ~EncoderState(); + }; + + size_t SamplesPerChannel() const; + + const size_t num_channels_; + const int payload_type_; + const size_t num_10ms_frames_per_packet_; + size_t num_10ms_frames_buffered_; + uint32_t first_timestamp_in_buffer_; + const std::unique_ptr<EncoderState[]> encoders_; + rtc::Buffer interleave_buffer_; +}; + +} // namespace webrtc +#endif // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ |