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}
This commit is contained in:
kwiberg
@ -8,172 +8,13 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file is for backwards compatibility only! Use
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// webrtc/api/audio_codecs/audio_decoder.h instead!
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// TODO(kwiberg): Remove it.
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
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#include <memory>
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#include <vector>
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#include "webrtc/api/audio_codecs/audio_decoder.h"
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#include "webrtc/base/array_view.h"
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// This is the interface class for decoders in NetEQ. Each codec type will have
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// and implementation of this class.
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class AudioDecoder {
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public:
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enum SpeechType {
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kSpeech = 1,
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kComfortNoise = 2
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};
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// Used by PacketDuration below. Save the value -1 for errors.
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enum { kNotImplemented = -2 };
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AudioDecoder() = default;
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virtual ~AudioDecoder() = default;
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class EncodedAudioFrame {
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public:
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struct DecodeResult {
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size_t num_decoded_samples;
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SpeechType speech_type;
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};
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virtual ~EncodedAudioFrame() = default;
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// Returns the duration in samples-per-channel of this audio frame.
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// If no duration can be ascertained, returns zero.
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virtual size_t Duration() const = 0;
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// Decodes this frame of audio and writes the result in |decoded|.
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// |decoded| must be large enough to store as many samples as indicated by a
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// call to Duration() . On success, returns an rtc::Optional containing the
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// total number of samples across all channels, as well as whether the
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// decoder produced comfort noise or speech. On failure, returns an empty
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// rtc::Optional. Decode may be called at most once per frame object.
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virtual rtc::Optional<DecodeResult> Decode(
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rtc::ArrayView<int16_t> decoded) const = 0;
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};
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struct ParseResult {
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ParseResult();
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ParseResult(uint32_t timestamp,
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int priority,
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std::unique_ptr<EncodedAudioFrame> frame);
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ParseResult(ParseResult&& b);
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~ParseResult();
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ParseResult& operator=(ParseResult&& b);
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// The timestamp of the frame is in samples per channel.
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uint32_t timestamp;
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// The relative priority of the frame compared to other frames of the same
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// payload and the same timeframe. A higher value means a lower priority.
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// The highest priority is zero - negative values are not allowed.
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int priority;
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std::unique_ptr<EncodedAudioFrame> frame;
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};
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// Let the decoder parse this payload and prepare zero or more decodable
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// frames. Each frame must be between 10 ms and 120 ms long. The caller must
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// ensure that the AudioDecoder object outlives any frame objects returned by
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// this call. The decoder is free to swap or move the data from the |payload|
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// buffer. |timestamp| is the input timestamp, in samples, corresponding to
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// the start of the payload.
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virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
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uint32_t timestamp);
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// Decodes |encode_len| bytes from |encoded| and writes the result in
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// |decoded|. The maximum bytes allowed to be written into |decoded| is
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// |max_decoded_bytes|. Returns the total number of samples across all
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// channels. If the decoder produced comfort noise, |speech_type|
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// is set to kComfortNoise, otherwise it is kSpeech. The desired output
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// sample rate is provided in |sample_rate_hz|, which must be valid for the
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// codec at hand.
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int Decode(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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size_t max_decoded_bytes,
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int16_t* decoded,
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SpeechType* speech_type);
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// Same as Decode(), but interfaces to the decoders redundant decode function.
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// The default implementation simply calls the regular Decode() method.
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int DecodeRedundant(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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size_t max_decoded_bytes,
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int16_t* decoded,
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SpeechType* speech_type);
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// Indicates if the decoder implements the DecodePlc method.
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virtual bool HasDecodePlc() const;
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// Calls the packet-loss concealment of the decoder to update the state after
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// one or several lost packets. The caller has to make sure that the
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// memory allocated in |decoded| should accommodate |num_frames| frames.
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virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
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// Resets the decoder state (empty buffers etc.).
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virtual void Reset() = 0;
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// Notifies the decoder of an incoming packet to NetEQ.
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virtual int IncomingPacket(const uint8_t* payload,
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size_t payload_len,
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uint16_t rtp_sequence_number,
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uint32_t rtp_timestamp,
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uint32_t arrival_timestamp);
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// Returns the last error code from the decoder.
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virtual int ErrorCode();
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// Returns the duration in samples-per-channel of the payload in |encoded|
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// which is |encoded_len| bytes long. Returns kNotImplemented if no duration
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// estimate is available, or -1 in case of an error.
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virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
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// Returns the duration in samples-per-channel of the redandant payload in
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// |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
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// duration estimate is available, or -1 in case of an error.
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virtual int PacketDurationRedundant(const uint8_t* encoded,
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size_t encoded_len) const;
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// Detects whether a packet has forward error correction. The packet is
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// comprised of the samples in |encoded| which is |encoded_len| bytes long.
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// Returns true if the packet has FEC and false otherwise.
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virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
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// Returns the actual sample rate of the decoder's output. This value may not
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// change during the lifetime of the decoder.
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virtual int SampleRateHz() const = 0;
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// The number of channels in the decoder's output. This value may not change
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// during the lifetime of the decoder.
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virtual size_t Channels() const = 0;
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protected:
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static SpeechType ConvertSpeechType(int16_t type);
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virtual int DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) = 0;
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virtual int DecodeRedundantInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type);
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
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