// Copyright 2016 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 "media/base/audio_latency.h"

#include <stdint.h>

#include <algorithm>

#include "base/logging.h"
#include "base/time/time.h"
#include "build/build_config.h"
#include "media/base/limits.h"

#if defined(OS_ANDROID)
#include "base/android/build_info.h"
#endif

#if defined(OS_MACOSX)
#include "media/base/mac/audio_latency_mac.h"
#endif

namespace media {

namespace {
#if !defined(OS_WIN)
// Taken from "Bit Twiddling Hacks"
// http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2
uint32_t RoundUpToPowerOfTwo(uint32_t v) {
  v--;
  v |= v >> 1;
  v |= v >> 2;
  v |= v >> 4;
  v |= v >> 8;
  v |= v >> 16;
  v++;
  return v;
}
#endif
}  // namespace

// static
bool AudioLatency::IsResamplingPassthroughSupported(LatencyType type) {
#if defined(OS_CHROMEOS)
  return true;
#elif defined(OS_ANDROID)
  // Only N MR1+ has support for OpenSLES performance modes which allow for
  // power efficient playback. Per the Android audio team, we shouldn't waste
  // cycles on resampling when using the playback mode. See OpenSLESOutputStream
  // for additional implementation details.
  return type == LATENCY_PLAYBACK &&
         base::android::BuildInfo::GetInstance()->sdk_int() >=
             base::android::SDK_VERSION_NOUGAT_MR1;
#else
  return false;
#endif
}

// static
int AudioLatency::GetHighLatencyBufferSize(int sample_rate,
                                           int preferred_buffer_size) {
  // Empirically, we consider 20ms of samples to be high latency.
#if !defined(USE_CRAS)
  const double twenty_ms_size = 2.0 * sample_rate / 100;
#endif

#if defined(OS_WIN)
  preferred_buffer_size = std::max(preferred_buffer_size, 1);

  // Windows doesn't use power of two buffer sizes, so we should always round up
  // to the nearest multiple of the output buffer size.
  const int high_latency_buffer_size =
      std::ceil(twenty_ms_size / preferred_buffer_size) * preferred_buffer_size;
#else
  // On other platforms use the nearest higher power of two buffer size.  For a
  // given sample rate, this works out to:
  //
  //     <= 3200   : 64
  //     <= 6400   : 128
  //     <= 12800  : 256
  //     <= 25600  : 512
  //     <= 51200  : 1024
  //     <= 102400 : 2048
  //     <= 204800 : 4096
  //
  // On Linux, the minimum hardware buffer size is 512, so the lower calculated
  // values are unused.  OSX may have a value as low as 128.
#if defined(USE_CRAS)
  const double eighty_ms_size = 8.0 * sample_rate / 100;
  const int high_latency_buffer_size = RoundUpToPowerOfTwo(eighty_ms_size);
#else
  const int high_latency_buffer_size = RoundUpToPowerOfTwo(twenty_ms_size);
#endif  // defined(USE_CRAS)
#endif  // defined(OS_WIN)

  return std::max(preferred_buffer_size, high_latency_buffer_size);
}

// static
int AudioLatency::GetRtcBufferSize(int sample_rate, int hardware_buffer_size) {
  // Use native hardware buffer size as default. On Windows, we strive to open
  // up using this native hardware buffer size to achieve best
  // possible performance and to ensure that no FIFO is needed on the browser
  // side to match the client request. That is why there is no #if case for
  // Windows below.
  int frames_per_buffer = hardware_buffer_size;

  // No |hardware_buffer_size| is specified, fall back to 10 ms buffer size.
  if (!frames_per_buffer) {
    frames_per_buffer = sample_rate / 100;
    DVLOG(1) << "Using 10 ms sink output buffer size: " << frames_per_buffer;
    return frames_per_buffer;
  }

#if defined(OS_LINUX) || defined(OS_MACOSX) || defined(OS_FUCHSIA)
  // On Linux, MacOS and Fuchsia, the low level IO implementations on the
  // browser side supports all buffer size the clients want. We use the native
  // peer connection buffer size (10ms) to achieve best possible performance.
  frames_per_buffer = sample_rate / 100;
#elif defined(OS_ANDROID)
  // TODO(olka/henrika): This settings are very old, need to be revisited.
  int frames_per_10ms = sample_rate / 100;
  if (frames_per_buffer < 2 * frames_per_10ms) {
    // Examples of low-latency frame sizes and the resulting |buffer_size|:
    //  Nexus 7     : 240 audio frames => 2*480 = 960
    //  Nexus 10    : 256              => 2*441 = 882
    //  Galaxy Nexus: 144              => 2*441 = 882
    frames_per_buffer = 2 * frames_per_10ms;
    DVLOG(1) << "Low-latency output detected on Android";
  }
#endif

  DVLOG(1) << "Using sink output buffer size: " << frames_per_buffer;
  return frames_per_buffer;
}

// static
int AudioLatency::GetInteractiveBufferSize(int hardware_buffer_size) {
#if defined(OS_ANDROID)
  // Always log this because it's relatively hard to get this
  // information out.
  LOG(INFO) << "audioHardwareBufferSize = " << hardware_buffer_size;
#endif

  return hardware_buffer_size;
}

int AudioLatency::GetExactBufferSize(base::TimeDelta duration,
                                     int sample_rate,
                                     int hardware_buffer_size,
                                     int min_hardware_buffer_size,
                                     int max_hardware_buffer_size,
                                     int max_allowed_buffer_size) {
  DCHECK_NE(0, hardware_buffer_size);
  DCHECK_NE(0, max_allowed_buffer_size);
  DCHECK_GE(hardware_buffer_size, min_hardware_buffer_size);
  DCHECK_GE(max_hardware_buffer_size, min_hardware_buffer_size);
  DCHECK(max_hardware_buffer_size == 0 ||
         hardware_buffer_size <= max_hardware_buffer_size);
  DCHECK(max_hardware_buffer_size == 0 ||
         max_hardware_buffer_size <= max_allowed_buffer_size);

  int requested_buffer_size = std::round(duration.InSecondsF() * sample_rate);

  if (min_hardware_buffer_size &&
      requested_buffer_size <= min_hardware_buffer_size)
    return min_hardware_buffer_size;

  if (requested_buffer_size <= hardware_buffer_size)
    return hardware_buffer_size;

#if defined(OS_WIN)
  // On Windows we allow either exactly the minimum buffer size (using
  // IAudioClient3) or multiples of the default buffer size using the previous
  // IAudioClient API.
  const int multiplier = hardware_buffer_size;
#else
  const int multiplier = min_hardware_buffer_size > 0 ? min_hardware_buffer_size
                                                      : hardware_buffer_size;
#endif

  int buffer_size =
      std::ceil(requested_buffer_size / static_cast<double>(multiplier)) *
      multiplier;

  // If the user is requesting a buffer size >= max_hardware_buffer_size then we
  // want the hardware to run at this max and then only return sizes that are
  // multiples of this here so that we don't end up with Web Audio running with
  // a period that's misaligned with the hardware one.
  if (max_hardware_buffer_size && buffer_size >= max_hardware_buffer_size) {
    buffer_size = std::ceil(requested_buffer_size /
                            static_cast<double>(max_hardware_buffer_size)) *
                  max_hardware_buffer_size;
  }

  const int platform_max_buffer_size =
      max_hardware_buffer_size
          ? (max_allowed_buffer_size / max_hardware_buffer_size) *
                max_hardware_buffer_size
          : (max_allowed_buffer_size / multiplier) * multiplier;

  return std::min(buffer_size, platform_max_buffer_size);
}
}  // namespace media