ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
112 lines
3.8 KiB
C++
112 lines
3.8 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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#include "webrtc/video_frame.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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// Return the time in ms until next video frame should be pulled (by
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// calling GetVideoFromFile(..)).
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// Note: this API reads one video frame from file. This means that it should
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// be called exactly once per GetVideoFromFile(..) API call.
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virtual int32_t TimeUntilNextVideoFrame() { return -1;}
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virtual int32_t StartPlayingVideoFile(
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const char* /*fileName*/,
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bool /*loop*/,
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bool /*videoOnly*/) { return -1;}
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virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const
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{return -1;}
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virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; }
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// Same as GetVideoFromFile(). videoFrame will have the resolution specified
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// by the width outWidth and height outHeight in pixels.
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virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/,
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const uint32_t /*outWidth*/,
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const uint32_t /*outHeight*/) {
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return -1;
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}
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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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