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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
parentInitial commit. (diff)
downloadfirefox-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_decoder_pcm.h')
-rw-r--r--third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h81
1 files changed, 81 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
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+/*
+ * 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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioDecoderPcmU final : public AudioDecoder {
+ public:
+ explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
+ RTC_DCHECK_GE(num_channels, 1);
+ }
+
+ AudioDecoderPcmU(const AudioDecoderPcmU&) = delete;
+ AudioDecoderPcmU& operator=(const AudioDecoderPcmU&) = delete;
+
+ void Reset() override;
+ std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+ uint32_t timestamp) override;
+ int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+ int SampleRateHz() const override;
+ size_t Channels() const override;
+
+ protected:
+ int DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) override;
+
+ private:
+ const size_t num_channels_;
+};
+
+class AudioDecoderPcmA final : public AudioDecoder {
+ public:
+ explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
+ RTC_DCHECK_GE(num_channels, 1);
+ }
+
+ AudioDecoderPcmA(const AudioDecoderPcmA&) = delete;
+ AudioDecoderPcmA& operator=(const AudioDecoderPcmA&) = delete;
+
+ void Reset() override;
+ std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+ uint32_t timestamp) override;
+ int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+ int SampleRateHz() const override;
+ size_t Channels() const override;
+
+ protected:
+ int DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) override;
+
+ private:
+ const size_t num_channels_;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_