Index: webrtc/modules/audio_coding/codecs/audio_decoder.h |
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h |
index 8468da20f2b74669fabcd3607cbf8f47ef35c715..da0628254d18e83a7aa1898930e79ba25dee56d4 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h |
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h |
@@ -8,172 +8,13 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
+// This file is for backwards compatibility only! Use |
+// webrtc/api/audio_codecs/audio_decoder.h instead! |
+// TODO(kwiberg): Remove it. |
+ |
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ |
#define WEBRTC_MODULES_AUDIO_CODING_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 { |
- |
-// This is the interface class for decoders in NetEQ. Each codec type will have |
-// and implementation of this class. |
-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); |
-}; |
+#include "webrtc/api/audio_codecs/audio_decoder.h" |
-} // namespace webrtc |
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ |