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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/g711/audio_encoder_pcm.h | |
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/g711/audio_encoder_pcm.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h | 128 |
1 files changed, 128 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h new file mode 100644 index 0000000000..d50be4b457 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h @@ -0,0 +1,128 @@ +/* + * 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_G711_AUDIO_ENCODER_PCM_H_ +#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ + +#include <utility> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/units/time_delta.h" + +namespace webrtc { + +class AudioEncoderPcm : public AudioEncoder { + public: + struct Config { + public: + bool IsOk() const; + + int frame_size_ms; + size_t num_channels; + int payload_type; + + protected: + explicit Config(int pt) + : frame_size_ms(20), num_channels(1), payload_type(pt) {} + }; + + ~AudioEncoderPcm() override; + + int SampleRateHz() const override; + size_t NumChannels() 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: + AudioEncoderPcm(const Config& config, int sample_rate_hz); + + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) override; + + virtual size_t EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) = 0; + + virtual size_t BytesPerSample() const = 0; + + // Used to set EncodedInfoLeaf::encoder_type in + // AudioEncoderPcm::EncodeImpl + virtual AudioEncoder::CodecType GetCodecType() const = 0; + + private: + const int sample_rate_hz_; + const size_t num_channels_; + const int payload_type_; + const size_t num_10ms_frames_per_packet_; + const size_t full_frame_samples_; + std::vector<int16_t> speech_buffer_; + uint32_t first_timestamp_in_buffer_; +}; + +class AudioEncoderPcmA final : public AudioEncoderPcm { + public: + struct Config : public AudioEncoderPcm::Config { + Config() : AudioEncoderPcm::Config(8) {} + }; + + explicit AudioEncoderPcmA(const Config& config) + : AudioEncoderPcm(config, kSampleRateHz) {} + + AudioEncoderPcmA(const AudioEncoderPcmA&) = delete; + AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete; + + protected: + size_t EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) override; + + size_t BytesPerSample() const override; + + AudioEncoder::CodecType GetCodecType() const override; + + private: + static const int kSampleRateHz = 8000; +}; + +class AudioEncoderPcmU final : public AudioEncoderPcm { + public: + struct Config : public AudioEncoderPcm::Config { + Config() : AudioEncoderPcm::Config(0) {} + }; + + explicit AudioEncoderPcmU(const Config& config) + : AudioEncoderPcm(config, kSampleRateHz) {} + + AudioEncoderPcmU(const AudioEncoderPcmU&) = delete; + AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete; + + protected: + size_t EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) override; + + size_t BytesPerSample() const override; + + AudioEncoder::CodecType GetCodecType() const override; + + private: + static const int kSampleRateHz = 8000; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ |