Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

112 lines
3.8 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
namespace webrtc {
class FileCallback;
class FilePlayer
{
public:
// The largest decoded frame size in samples (60ms with 32kHz sample rate).
enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
// Note: will return NULL for unsupported formats.
static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
const FileFormats fileFormat);
static void DestroyFilePlayer(FilePlayer* player);
// Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
// will be set to the number of samples read (not the number of samples per
// channel).
virtual int Get10msAudioFromFile(
int16_t* outBuffer,
size_t& lengthInSamples,
int frequencyInHz) = 0;
// Register callback for receiving file playing notifications.
virtual int32_t RegisterModuleFileCallback(
FileCallback* callback) = 0;
// API for playing audio from fileName to channel.
// Note: codecInst is used for pre-encoded files.
virtual int32_t StartPlayingFile(
const char* fileName,
bool loop,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition = 0,
const CodecInst* codecInst = NULL) = 0;
// Note: codecInst is used for pre-encoded files.
virtual int32_t StartPlayingFile(
InStream& sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition = 0,
const CodecInst* codecInst = NULL) = 0;
virtual int32_t StopPlayingFile() = 0;
virtual bool IsPlayingFile() const = 0;
virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
// Set audioCodec to the currently used audio codec.
virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
virtual int32_t Frequency() const = 0;
// Note: scaleFactor is in the range [0.0 - 2.0]
virtual int32_t SetAudioScaling(float scaleFactor) = 0;
// Return the time in ms until next video frame should be pulled (by
// calling GetVideoFromFile(..)).
// Note: this API reads one video frame from file. This means that it should
// be called exactly once per GetVideoFromFile(..) API call.
virtual int32_t TimeUntilNextVideoFrame() { return -1;}
virtual int32_t StartPlayingVideoFile(
const char* /*fileName*/,
bool /*loop*/,
bool /*videoOnly*/) { return -1;}
virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const
{return -1;}
virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; }
// Same as GetVideoFromFile(). videoFrame will have the resolution specified
// by the width outWidth and height outHeight in pixels.
virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/,
const uint32_t /*outWidth*/,
const uint32_t /*outHeight*/) {
return -1;
}
protected:
virtual ~FilePlayer() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_