// Copyright (c) 2012 The Chromium Authors. All rights reserved. // Use of this source code is governed by a BSD-style license that can be // found in the LICENSE file. #ifndef MEDIA_AUDIO_AUDIO_UTIL_H_ #define MEDIA_AUDIO_AUDIO_UTIL_H_ #include #include #include "base/basictypes.h" #include "media/base/channel_layout.h" #include "media/base/media_export.h" namespace base { class SharedMemory; } namespace media { // For all audio functions 3 audio formats are supported: // 8 bits unsigned 0 to 255. // 16 bit signed (little endian). // 32 bit signed (little endian) // AdjustVolume() does a software volume adjustment of a sample buffer. // The samples are multiplied by the volume, which should range from // 0.0 (mute) to 1.0 (full volume). // Using software allows each audio and video to have its own volume without // affecting the master volume. // In the future the function may be used to adjust the sample format to // simplify hardware requirements and to support a wider variety of input // formats. // The buffer is modified in-place to avoid memory management, as this // function may be called in performance critical code. MEDIA_EXPORT bool AdjustVolume(void* buf, size_t buflen, int channels, int bytes_per_sample, float volume); // MixStreams() mixes 2 audio streams with same sample rate and number of // samples, adjusting volume on one of them. // Dst += Src * volume. MEDIA_EXPORT void MixStreams(void* dst, void* src, size_t buflen, int bytes_per_sample, float volume); // FoldChannels() does a software multichannel folding down to stereo. // Channel order is assumed to be 5.1 Dolby standard which is // front left, front right, center, surround left, surround right. // The subwoofer is ignored. // 6.1 adds a rear center speaker, and 7.1 has 2 rear speakers. These // channels are rare and ignored. // After summing the channels, volume is adjusted and the samples are // clipped to the maximum value. // Volume should normally range from 0.0 (mute) to 1.0 (full volume), but // since clamping is performed a value of more than 1 is allowed to increase // volume. // The buffer is modified in-place to avoid memory management, as this // function may be called in performance critical code. MEDIA_EXPORT bool FoldChannels(void* buf, size_t buflen, int channels, int bytes_per_sample, float volume); // DeinterleaveAudioChannel() takes interleaved audio buffer |source| // of the given |sample_fmt| and |number_of_channels| and extracts // |number_of_frames| data for the given |channel_index| and // puts it in the floating point |destination|. // It returns |true| on success, or |false| if the |sample_fmt| is // not recognized. MEDIA_EXPORT bool DeinterleaveAudioChannel(void* source, float* destination, int channels, int channel_index, int bytes_per_sample, size_t number_of_frames); // InterleaveFloatToInt scales, clips, and interleaves the planar // floating-point audio contained in |source| to the |destination|. // The floating-point data is in a canonical range of -1.0 -> +1.0. // The size of the |source| vector determines the number of channels. // The |destination| buffer is assumed to be large enough to hold the // result. Thus it must be at least size: number_of_frames * source.size() MEDIA_EXPORT void InterleaveFloatToInt(const std::vector& source, void* destination, size_t number_of_frames, int bytes_per_sample); // Returns the default audio output hardware sample-rate. MEDIA_EXPORT int GetAudioHardwareSampleRate(); // Returns the audio input hardware sample-rate for the specified device. MEDIA_EXPORT int GetAudioInputHardwareSampleRate( const std::string& device_id); // Returns the optimal low-latency buffer size for the audio hardware. // This is the smallest buffer size the system can comfortably render // at without glitches. The buffer size is in sample-frames. MEDIA_EXPORT size_t GetAudioHardwareBufferSize(); // Returns the channel layout for the specified audio input device. MEDIA_EXPORT ChannelLayout GetAudioInputHardwareChannelLayout( const std::string& device_id); // Computes a buffer size based on the given |sample_rate|. Must be used in // conjunction with AUDIO_PCM_LINEAR. MEDIA_EXPORT size_t GetHighLatencyOutputBufferSize(int sample_rate); // Functions that handle data buffer passed between processes in the shared // memory. Called on both IPC sides. MEDIA_EXPORT uint32 TotalSharedMemorySizeInBytes(uint32 packet_size); MEDIA_EXPORT uint32 PacketSizeSizeInBytes(uint32 shared_memory_created_size); MEDIA_EXPORT uint32 GetActualDataSizeInBytes(base::SharedMemory* shared_memory, uint32 shared_memory_size); MEDIA_EXPORT void SetActualDataSizeInBytes(base::SharedMemory* shared_memory, uint32 shared_memory_size, uint32 actual_data_size); MEDIA_EXPORT void SetUnknownDataSize(base::SharedMemory* shared_memory, uint32 shared_memory_size); MEDIA_EXPORT bool IsUnknownDataSize(base::SharedMemory* shared_memory, uint32 shared_memory_size); #if defined(OS_WIN) // Does Windows support WASAPI? We are checking in lot of places, and // sometimes check was written incorrectly, so move into separate function. MEDIA_EXPORT bool IsWASAPISupported(); // Returns number of buffers to be used by wave out. MEDIA_EXPORT int NumberOfWaveOutBuffers(); #endif // defined(OS_WIN) } // namespace media #endif // MEDIA_AUDIO_AUDIO_UTIL_H_