92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
81 lines
2.9 KiB
C++
81 lines
2.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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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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#ifndef VOICE_ENGINE_FILE_PLAYER_H_
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#define VOICE_ENGINE_FILE_PLAYER_H_
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#include <memory>
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#include "common_types.h"
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#include "modules/include/module_common_types.h"
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#include "typedefs.h"
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namespace webrtc {
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class FileCallback;
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class FilePlayer {
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public:
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// The largest decoded frame size in samples (60ms with 48kHz sample rate).
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enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 48 };
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enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
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// Note: will return NULL for unsupported formats.
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static std::unique_ptr<FilePlayer> CreateFilePlayer(
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const uint32_t instanceID,
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const FileFormats fileFormat);
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virtual ~FilePlayer() = default;
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// Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
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// will be set to the number of samples read (not the number of samples per
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// channel).
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virtual int Get10msAudioFromFile(int16_t* outBuffer,
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size_t* lengthInSamples,
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int frequencyInHz) = 0;
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// Register callback for receiving file playing notifications.
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virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
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// API for playing audio from fileName to channel.
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// Note: codecInst is used for pre-encoded files.
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virtual int32_t StartPlayingFile(const char* fileName,
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bool loop,
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uint32_t startPosition,
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float volumeScaling,
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uint32_t notification,
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uint32_t stopPosition,
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const CodecInst* codecInst) = 0;
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// Note: codecInst is used for pre-encoded files.
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virtual int32_t StartPlayingFile(InStream* sourceStream,
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uint32_t startPosition,
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float volumeScaling,
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uint32_t notification,
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uint32_t stopPosition,
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const CodecInst* codecInst) = 0;
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virtual int32_t StopPlayingFile() = 0;
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virtual bool IsPlayingFile() const = 0;
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virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
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// Set audioCodec to the currently used audio codec.
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virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
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virtual int32_t Frequency() const = 0;
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// Note: scaleFactor is in the range [0.0 - 2.0]
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virtual int32_t SetAudioScaling(float scaleFactor) = 0;
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};
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} // namespace webrtc
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#endif // VOICE_ENGINE_FILE_PLAYER_H_
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