83d27683a8
In particular, PlayoutStereoData and StartPlayingAudioFile. This also eliminates the dependency on system_wrappers FileWrapper. Bug: None Change-Id: I61df1eea1ad5f5035e36c8229febbf3668808f65 Reviewed-on: https://webrtc-review.googlesource.com/28121 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21038}
85 lines
3.4 KiB
C++
85 lines
3.4 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 MODULES_MEDIA_FILE_MEDIA_FILE_H_
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#define MODULES_MEDIA_FILE_MEDIA_FILE_H_
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/include/module.h"
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#include "modules/include/module_common_types.h"
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#include "modules/media_file/media_file_defines.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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class MediaFile : public Module
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{
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public:
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// Factory method. Constructor disabled. id is the identifier for the
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// MediaFile instance.
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static MediaFile* CreateMediaFile(const int32_t id);
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static void DestroyMediaFile(MediaFile* module);
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// Put 10-60ms of audio data from file into the audioBuffer depending on
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// codec frame size. dataLengthInBytes is both an input and output
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// parameter. As input parameter it indicates the size of audioBuffer.
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// As output parameter it indicates the number of bytes written to
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// audioBuffer.
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// Note: This API only play mono audio but can be used on file containing
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// audio with more channels (in which case the audio will be converted to
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// mono).
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virtual int32_t PlayoutAudioData(
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int8_t* audioBuffer,
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size_t& dataLengthInBytes) = 0;
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// Prepare for playing audio from stream.
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// FileCallback::PlayNotification(..) will be called after
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// notificationTimeMs of the file has been played if notificationTimeMs is
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// greater than zero. format specifies the type of file fileName refers to.
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// codecInst specifies the encoding of the audio data. Note that
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// file formats that contain this information (like WAV files) don't need to
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// provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
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// specify what part of the file should be read. From startPointMs ms to
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// stopPointMs ms.
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// Note: codecInst.channels should be set to 2 for stereo (and 1 for
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// mono). Stereo audio is only supported for WAV files.
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virtual int32_t StartPlayingAudioStream(
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InStream& stream,
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const uint32_t notificationTimeMs = 0,
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const FileFormats format = kFileFormatPcm16kHzFile,
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const CodecInst* codecInst = NULL,
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const uint32_t startPointMs = 0,
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const uint32_t stopPointMs = 0) = 0;
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// Stop playing from file or stream.
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virtual int32_t StopPlaying() = 0;
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// Return true if playing.
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virtual bool IsPlaying() = 0;
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// Set durationMs to the number of ms that has been played from file.
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virtual int32_t PlayoutPositionMs(
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uint32_t& durationMs) const = 0;
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// Register callback to receive media file related notifications. Disables
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// callbacks if callback is NULL.
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virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0;
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// Update codecInst according to the current audio codec being used for
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// reading or writing.
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virtual int32_t codec_info(CodecInst& codecInst) const = 0;
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protected:
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MediaFile() {}
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virtual ~MediaFile() {}
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};
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} // namespace webrtc
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#endif // MODULES_MEDIA_FILE_MEDIA_FILE_H_
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