Move AudioDecoder and related stuff to the api/ directory
BUG=webrtc:5805, webrtc:6725
Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}
diff --git a/webrtc/api/audio_codecs/audio_decoder.h b/webrtc/api/audio_codecs/audio_decoder.h
new file mode 100644
index 0000000..dab7d3b
--- /dev/null
+++ b/webrtc/api/audio_codecs/audio_decoder.h
@@ -0,0 +1,177 @@
+/*
+ * Copyright (c) 2012 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 WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_
+#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioDecoder {
+ public:
+ enum SpeechType {
+ kSpeech = 1,
+ kComfortNoise = 2,
+ };
+
+ // Used by PacketDuration below. Save the value -1 for errors.
+ enum { kNotImplemented = -2 };
+
+ AudioDecoder() = default;
+ virtual ~AudioDecoder() = default;
+
+ class EncodedAudioFrame {
+ public:
+ struct DecodeResult {
+ size_t num_decoded_samples;
+ SpeechType speech_type;
+ };
+
+ virtual ~EncodedAudioFrame() = default;
+
+ // Returns the duration in samples-per-channel of this audio frame.
+ // If no duration can be ascertained, returns zero.
+ virtual size_t Duration() const = 0;
+
+ // Decodes this frame of audio and writes the result in |decoded|.
+ // |decoded| must be large enough to store as many samples as indicated by a
+ // call to Duration() . On success, returns an rtc::Optional containing the
+ // total number of samples across all channels, as well as whether the
+ // decoder produced comfort noise or speech. On failure, returns an empty
+ // rtc::Optional. Decode may be called at most once per frame object.
+ virtual rtc::Optional<DecodeResult> Decode(
+ rtc::ArrayView<int16_t> decoded) const = 0;
+ };
+
+ struct ParseResult {
+ ParseResult();
+ ParseResult(uint32_t timestamp,
+ int priority,
+ std::unique_ptr<EncodedAudioFrame> frame);
+ ParseResult(ParseResult&& b);
+ ~ParseResult();
+
+ ParseResult& operator=(ParseResult&& b);
+
+ // The timestamp of the frame is in samples per channel.
+ uint32_t timestamp;
+ // The relative priority of the frame compared to other frames of the same
+ // payload and the same timeframe. A higher value means a lower priority.
+ // The highest priority is zero - negative values are not allowed.
+ int priority;
+ std::unique_ptr<EncodedAudioFrame> frame;
+ };
+
+ // Let the decoder parse this payload and prepare zero or more decodable
+ // frames. Each frame must be between 10 ms and 120 ms long. The caller must
+ // ensure that the AudioDecoder object outlives any frame objects returned by
+ // this call. The decoder is free to swap or move the data from the |payload|
+ // buffer. |timestamp| is the input timestamp, in samples, corresponding to
+ // the start of the payload.
+ virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+ uint32_t timestamp);
+
+ // Decodes |encode_len| bytes from |encoded| and writes the result in
+ // |decoded|. The maximum bytes allowed to be written into |decoded| is
+ // |max_decoded_bytes|. Returns the total number of samples across all
+ // channels. If the decoder produced comfort noise, |speech_type|
+ // is set to kComfortNoise, otherwise it is kSpeech. The desired output
+ // sample rate is provided in |sample_rate_hz|, which must be valid for the
+ // codec at hand.
+ int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ size_t max_decoded_bytes,
+ int16_t* decoded,
+ SpeechType* speech_type);
+
+ // Same as Decode(), but interfaces to the decoders redundant decode function.
+ // The default implementation simply calls the regular Decode() method.
+ int DecodeRedundant(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ size_t max_decoded_bytes,
+ int16_t* decoded,
+ SpeechType* speech_type);
+
+ // Indicates if the decoder implements the DecodePlc method.
+ virtual bool HasDecodePlc() const;
+
+ // Calls the packet-loss concealment of the decoder to update the state after
+ // one or several lost packets. The caller has to make sure that the
+ // memory allocated in |decoded| should accommodate |num_frames| frames.
+ virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
+
+ // Resets the decoder state (empty buffers etc.).
+ virtual void Reset() = 0;
+
+ // Notifies the decoder of an incoming packet to NetEQ.
+ virtual int IncomingPacket(const uint8_t* payload,
+ size_t payload_len,
+ uint16_t rtp_sequence_number,
+ uint32_t rtp_timestamp,
+ uint32_t arrival_timestamp);
+
+ // Returns the last error code from the decoder.
+ virtual int ErrorCode();
+
+ // Returns the duration in samples-per-channel of the payload in |encoded|
+ // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
+ // estimate is available, or -1 in case of an error.
+ virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+
+ // Returns the duration in samples-per-channel of the redandant payload in
+ // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
+ // duration estimate is available, or -1 in case of an error.
+ virtual int PacketDurationRedundant(const uint8_t* encoded,
+ size_t encoded_len) const;
+
+ // Detects whether a packet has forward error correction. The packet is
+ // comprised of the samples in |encoded| which is |encoded_len| bytes long.
+ // Returns true if the packet has FEC and false otherwise.
+ virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
+
+ // Returns the actual sample rate of the decoder's output. This value may not
+ // change during the lifetime of the decoder.
+ virtual int SampleRateHz() const = 0;
+
+ // The number of channels in the decoder's output. This value may not change
+ // during the lifetime of the decoder.
+ virtual size_t Channels() const = 0;
+
+ protected:
+ static SpeechType ConvertSpeechType(int16_t type);
+
+ virtual int DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) = 0;
+
+ virtual int DecodeRedundantInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type);
+
+ private:
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_