e7edea9759
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).
Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.
This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
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> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163e
TBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
87 lines
2.8 KiB
C++
87 lines
2.8 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 WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
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#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
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#include "webrtc/common_types.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class FileCallback;
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class FilePlayer
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{
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public:
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// The largest decoded frame size in samples (60ms with 32kHz sample rate).
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enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
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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 FilePlayer* CreateFilePlayer(const uint32_t instanceID,
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const FileFormats fileFormat);
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static void DestroyFilePlayer(FilePlayer* player);
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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(
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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(
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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(
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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 = 0,
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const CodecInst* codecInst = NULL) = 0;
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// Note: codecInst is used for pre-encoded files.
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virtual int32_t StartPlayingFile(
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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 = 0,
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const CodecInst* codecInst = NULL) = 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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protected:
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virtual ~FilePlayer() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
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