blob: 68fbd54754f51c9254e0cc827508e2e11d14c05f [file] [log] [blame]
// 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.
#include <stdint.h>
#include <string>
#include "base/compiler_specific.h"
#include "base/time/time.h"
#include "build/build_config.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_latency.h"
#include "media/base/audio_point.h"
#include "media/base/channel_layout.h"
#include "media/base/media_export.h"
namespace media {
// Use a struct-in-struct approach to ensure that we can calculate the required
// size as sizeof(Audio{Input,Output}BufferParameters) + #(bytes in audio
// buffer) without using packing. Also align Audio{Input,Output}BufferParameters
// instead of in Audio{Input,Output}Buffer to be able to calculate size like so.
// Use a macro for the alignment value that's the same as
// AudioBus::kChannelAlignment, since MSVC doesn't accept the latter to be used.
#if defined(OS_WIN)
#pragma warning(push)
#pragma warning(disable : 4324) // Disable warning for added padding.
static_assert(AudioBus::kChannelAlignment == PARAMETERS_ALIGNMENT,
"Audio buffer parameters struct alignment not same as AudioBus");
double volume;
uint32_t size;
uint32_t hardware_delay_bytes;
uint32_t id;
bool key_pressed;
uint32_t frames_skipped;
int64_t delay;
int64_t delay_timestamp;
#if defined(OS_WIN)
#pragma warning(pop)
static_assert(sizeof(AudioInputBufferParameters) %
AudioBus::kChannelAlignment ==
"AudioInputBufferParameters not aligned");
static_assert(sizeof(AudioOutputBufferParameters) %
AudioBus::kChannelAlignment ==
"AudioOutputBufferParameters not aligned");
struct MEDIA_EXPORT AudioInputBuffer {
AudioInputBufferParameters params;
int8_t audio[1];
struct MEDIA_EXPORT AudioOutputBuffer {
AudioOutputBufferParameters params;
int8_t audio[1];
class MEDIA_EXPORT AudioParameters {
// TODO(miu): Rename this enum to something that correctly reflects its
// semantics, such as "TransportScheme."
enum Format {
AUDIO_PCM_LINEAR = 0, // PCM is 'raw' amplitude samples.
AUDIO_PCM_LOW_LATENCY, // Linear PCM, low latency requested.
AUDIO_BITSTREAM_AC3, // Compressed AC3 bitstream.
AUDIO_BITSTREAM_EAC3, // Compressed E-AC3 bitstream.
AUDIO_FAKE, // Creates a fake AudioOutputStream object.
AUDIO_FORMAT_LAST = AUDIO_FAKE, // Only used for validation of format.
enum {
// Telephone quality sample rate, mostly for speech-only audio.
kTelephoneSampleRate = 8000,
// CD sampling rate is 44.1 KHz or conveniently 2x2x3x3x5x5x7x7.
kAudioCDSampleRate = 44100,
// Bitmasks to determine whether certain platform (typically hardware) audio
// effects should be enabled.
enum PlatformEffectsMask {
DUCKING = 0x2, // Enables ducking if the OS supports it.
HOTWORD = 0x8,
AudioParameters(Format format,
ChannelLayout channel_layout,
int sample_rate,
int bits_per_sample,
int frames_per_buffer);
// Re-initializes all members.
void Reset(Format format,
ChannelLayout channel_layout,
int sample_rate,
int bits_per_sample,
int frames_per_buffer);
// Checks that all values are in the expected range. All limits are specified
// in media::Limits.
bool IsValid() const;
// Returns a human-readable string describing |*this|. For debugging & test
// output only.
std::string AsHumanReadableString() const;
// Returns size of audio buffer in bytes.
int GetBytesPerBuffer() const;
// Returns the number of bytes representing one second of audio.
int GetBytesPerSecond() const;
// Returns the number of bytes representing a frame of audio.
int GetBytesPerFrame() const;
// Returns the number of microseconds per frame of audio. Intentionally
// reported as a double to surface of partial microseconds per frame, which
// is common for many sample rates. Failing to account for these nanoseconds
// can lead to audio/video sync drift.
double GetMicrosecondsPerFrame() const;
// Returns the duration of this buffer as calculated from frames_per_buffer()
// and sample_rate().
base::TimeDelta GetBufferDuration() const;
// Comparison with other AudioParams.
bool Equals(const AudioParameters& other) const;
// Return true if |format_| is compressed bitstream.
bool IsBitstreamFormat() const;
void set_format(Format format) { format_ = format; }
Format format() const { return format_; }
// A setter for channel_layout_ is intentionally excluded.
ChannelLayout channel_layout() const { return channel_layout_; }
// The number of channels is usually computed from channel_layout_. Setting
// this explictly is only required with CHANNEL_LAYOUT_DISCRETE.
void set_channels_for_discrete(int channels) {
channels == ChannelLayoutToChannelCount(channel_layout_));
channels_ = channels;
int channels() const { return channels_; }
void set_sample_rate(int sample_rate) { sample_rate_ = sample_rate; }
int sample_rate() const { return sample_rate_; }
void set_bits_per_sample(int bits_per_sample) {
bits_per_sample_ = bits_per_sample;
int bits_per_sample() const { return bits_per_sample_; }
void set_frames_per_buffer(int frames_per_buffer) {
frames_per_buffer_ = frames_per_buffer;
int frames_per_buffer() const { return frames_per_buffer_; }
void set_effects(int effects) { effects_ = effects; }
int effects() const { return effects_; }
void set_mic_positions(const std::vector<Point>& mic_positions) {
mic_positions_ = mic_positions;
const std::vector<Point>& mic_positions() const { return mic_positions_; }
void set_latency_tag(AudioLatency::LatencyType latency_tag) {
latency_tag_ = latency_tag;
AudioLatency::LatencyType latency_tag() const { return latency_tag_; }
AudioParameters(const AudioParameters&);
AudioParameters& operator=(const AudioParameters&);
// Creates reasonable dummy parameters in case no device is available.
static AudioParameters UnavailableDeviceParams();
Format format_; // Format of the stream.
ChannelLayout channel_layout_; // Order of surround sound channels.
int channels_; // Number of channels. Value set based on
// |channel_layout|.
int sample_rate_; // Sampling frequency/rate.
int bits_per_sample_; // Number of bits per sample.
int frames_per_buffer_; // Number of frames in a buffer.
int effects_; // Bitmask using PlatformEffectsMask.
// Microphone positions using Cartesian coordinates:
// x: the horizontal dimension, with positive to the right from the camera's
// perspective.
// y: the depth dimension, with positive forward from the camera's
// perspective.
// z: the vertical dimension, with positive upwards.
// Usually, the center of the microphone array will be treated as the origin
// (often the position of the camera).
// An empty vector indicates unknown positions.
std::vector<Point> mic_positions_;
// Optional tag to pass latency info from renderer to browser. Set to
// AudioLatency::LATENCY_COUNT by default, which means "not specified".
AudioLatency::LatencyType latency_tag_;
// Comparison is useful when AudioParameters is used with std structures.
inline bool operator<(const AudioParameters& a, const AudioParameters& b) {
if (a.format() != b.format())
return a.format() < b.format();
if (a.channels() != b.channels())
return a.channels() < b.channels();
if (a.sample_rate() != b.sample_rate())
return a.sample_rate() < b.sample_rate();
if (a.bits_per_sample() != b.bits_per_sample())
return a.bits_per_sample() < b.bits_per_sample();
return a.frames_per_buffer() < b.frames_per_buffer();
} // namespace media