// Copyright 2016 The Fuchsia 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 "src/media/audio/audio_core/audio_output.h"

#include <lib/fit/defer.h>
#include <lib/zx/clock.h>

#include <limits>

#include <fbl/auto_lock.h>

#include "src/lib/fxl/time/time_delta.h"
#include "src/media/audio/audio_core/audio_renderer_impl.h"
#include "src/media/audio/audio_core/mixer/mixer.h"
#include "src/media/audio/audio_core/mixer/no_op.h"
#include "src/media/audio/lib/logging/logging.h"

namespace media::audio {

static constexpr fxl::TimeDelta kMaxTrimPeriod = fxl::TimeDelta::FromMilliseconds(10);

AudioOutput::AudioOutput(AudioDeviceManager* manager) : AudioDevice(Type::Output, manager) {
  next_sched_time_ = fxl::TimePoint::Now();
  next_sched_time_known_ = true;
  source_link_refs_.reserve(16u);
}

zx_status_t AudioOutput::Init() {
  zx_status_t res = AudioDevice::Init();
  if (res != ZX_OK) {
    return res;
  }

  mix_timer_ = dispatcher::Timer::Create();
  if (mix_timer_ == nullptr) {
    return ZX_ERR_NO_MEMORY;
  }

  dispatcher::Timer::ProcessHandler process_handler(
      [output = fbl::WrapRefPtr(this)](dispatcher::Timer* timer) -> zx_status_t {
        OBTAIN_EXECUTION_DOMAIN_TOKEN(token, output->mix_domain_);
        output->Process();
        return ZX_OK;
      });

  res = mix_timer_->Activate(mix_domain_, std::move(process_handler));
  if (res != ZX_OK) {
    FXL_PLOG(ERROR, res) << "Failed to activate mix_timer_";
  }

  return res;
}

void AudioOutput::Process() {
  bool mixed = false;
  fxl::TimePoint now = fxl::TimePoint::Now();

  // At this point, we should always know when our implementation would like to be called to do some
  // mixing work next. If we do not know, then we should have already shut down.
  //
  // If the next sched time has not arrived yet, don't attempt to mix anything. Just trim the queues
  // and move on.
  FXL_DCHECK(next_sched_time_known_);
  if (now >= next_sched_time_) {
    // Clear the flag. If the implementation does not set it during the cycle by calling
    // SetNextSchedTime, we consider it an error and shut down.
    next_sched_time_known_ = false;

    // As long as our implementation wants to mix more and has not run into a problem trying to
    // finish the mix job, mix some more.
    do {
      memset(&cur_mix_job_, 0, sizeof(cur_mix_job_));

      if (!StartMixJob(&cur_mix_job_, now)) {
        break;
      }

      // If we have a mix job, then we must have an output producer, and an intermediate buffer
      // allocated, and it must be large enough for the mix job we were given.
      FXL_DCHECK(mix_buf_);
      FXL_DCHECK(output_producer_);
      FXL_DCHECK(cur_mix_job_.buf_frames <= mix_buf_frames_);

      // If we are not muted, actually do the mix. Otherwise, just fill the final buffer with
      // silence. Do not set the 'mixed' flag if we are muted. This is our signal that we still need
      // to trim our sources (something that happens automatically if we mix).
      if (!cur_mix_job_.sw_output_muted) {
        // Fill the intermediate buffer with silence.
        size_t bytes_to_zero =
            sizeof(mix_buf_[0]) * cur_mix_job_.buf_frames * output_producer_->channels();
        std::memset(mix_buf_.get(), 0, bytes_to_zero);

        // Mix each renderer into the intermediate accumulator buffer, then reformat (and clip) into
        // the final output buffer.
        ForEachLink(TaskType::Mix);
        output_producer_->ProduceOutput(mix_buf_.get(), cur_mix_job_.buf, cur_mix_job_.buf_frames);
        mixed = true;
      } else {
        output_producer_->FillWithSilence(cur_mix_job_.buf, cur_mix_job_.buf_frames);
      }

    } while (FinishMixJob(cur_mix_job_));
  }

  if (!next_sched_time_known_) {
    FXL_LOG(ERROR) << "Output failed to schedule next service time. Shutting down!";
    ShutdownSelf();
    return;
  }

  // If we mixed nothing this time, make sure that we trim all of our renderer queues. No matter
  // what is going on with the output hardware, we are not allowed to hold onto the queued data past
  // its presentation time.
  if (!mixed) {
    ForEachLink(TaskType::Trim);
  }

  // Figure out when we should wake up to do more work again. No matter how long our implementation
  // wants to wait, we need to make sure to wake up and periodically trim our input queues.
  fxl::TimePoint max_sched_time = now + kMaxTrimPeriod;
  if (next_sched_time_ > max_sched_time) {
    next_sched_time_ = max_sched_time;
  }

  zx_time_t next_time = static_cast<zx_time_t>(next_sched_time_.ToEpochDelta().ToNanoseconds());
  if (mix_timer_->Arm(next_time) != ZX_OK) {
    ShutdownSelf();
  }
}

zx_status_t AudioOutput::InitializeSourceLink(const fbl::RefPtr<AudioLink>& link) {
  auto mix_bookkeeping = std::make_unique<Bookkeeping>();

  // For now, refuse to link to anything but a packet source. This code does not currently know how
  // to properly handle a ring-buffer source.
  if (link->source_type() != AudioLink::SourceType::Packet) {
    return ZX_ERR_INTERNAL;
  }

  // If we have an output, pick a mixer based on the input and output formats. Otherwise, we only
  // need a NoOp mixer (for the time being).
  auto& packet_link = static_cast<AudioLinkPacketSource&>(*link);
  if (output_producer_) {
    mix_bookkeeping->mixer =
        Mixer::Select(packet_link.format_info().format(), *(output_producer_->format()));
  } else {
    mix_bookkeeping->mixer = std::make_unique<audio::mixer::NoOp>();
  }

  if (mix_bookkeeping->mixer == nullptr) {
    FXL_LOG(ERROR)
        << "*** Audio system mixer cannot convert between formats *** (could not select mixer "
           "while linking to output). Usually, this indicates a 'num_channels' mismatch.";
    return ZX_ERR_NOT_SUPPORTED;
  }

  // The Gain object contains multiple stages. In render, stream gain is "source" gain and device
  // (or system) gain is "dest" gain.
  //
  // The renderer will set this link's source gain once this call returns.
  //
  // Set the dest gain -- device gain retrieved from device settings.
  if (device_settings_ != nullptr) {
    AudioDeviceSettings::GainState cur_gain_state;
    device_settings_->SnapshotGainState(&cur_gain_state);

    mix_bookkeeping->gain.SetDestMute(cur_gain_state.muted);
    mix_bookkeeping->gain.SetDestGain(cur_gain_state.gain_db);
  }
  // Settings should exist but if they don't, we use default DestGain (Unity).

  // Connect usage gain from the source to the gain object.
  auto audio_renderer = fbl::RefPtr<AudioRendererImpl>::Downcast(link->GetSource());
  fuchsia::media::Usage usage;
  usage.set_render_usage(audio_renderer->GetUsage());
  mix_bookkeeping->gain.SetUsage(std::move(usage));

  // Things went well. Stash a reference to our bookkeeping and get out.
  packet_link.set_bookkeeping(std::move(mix_bookkeeping));
  return ZX_OK;
}

// Create our intermediate accumulation buffer.
void AudioOutput::SetupMixBuffer(uint32_t max_mix_frames) {
  FXL_DCHECK(output_producer_->channels() > 0u);
  FXL_DCHECK(max_mix_frames > 0u);
  FXL_DCHECK(static_cast<uint64_t>(max_mix_frames) * output_producer_->channels() <=
             std::numeric_limits<uint32_t>::max());

  mix_buf_frames_ = max_mix_frames;
  mix_buf_.reset(new float[mix_buf_frames_ * output_producer_->channels()]);
}

void AudioOutput::ForEachLink(TaskType task_type) {
  // Make a copy of our currently active set of links so that we don't have to hold onto mutex_ for
  // the entire mix operation.
  {
    fbl::AutoLock links_lock(&links_lock_);
    ZX_DEBUG_ASSERT(source_link_refs_.empty());
    for (auto& link : source_links_) {
      // For now, skip ring-buffer source links. This code cannot mix them yet.
      if (link.source_type() != AudioLink::SourceType::Packet) {
        continue;
      }

      source_link_refs_.emplace_back(fbl::WrapRefPtr(&link));
    }
  }

  // In all cases, release our temporary references upon leaving this method.
  auto cleanup = fit::defer([this]() FXL_NO_THREAD_SAFETY_ANALYSIS { source_link_refs_.clear(); });

  for (const auto& link : source_link_refs_) {
    // Quit early if we should be shutting down.
    if (is_shutting_down()) {
      return;
    }

    // Is the link still valid?  If so, process it.
    if (!link->valid()) {
      continue;
    }

    FXL_DCHECK(link->source_type() == AudioLink::SourceType::Packet);
    FXL_DCHECK(link->GetSource()->type() == AudioObject::Type::AudioRenderer);
    auto packet_link = fbl::RefPtr<AudioLinkPacketSource>::Downcast(link);
    auto audio_renderer = fbl::RefPtr<AudioRendererImpl>::Downcast(link->GetSource());

    // It would be nice to be able to use a dynamic cast for this, but currently we are building
    // with no-rtti
    FXL_DCHECK(packet_link->bookkeeping() != nullptr);
    auto& info = *packet_link->bookkeeping();

    // Ensure the mapping from source-frame to local-time is up-to-date.
    UpdateSourceTrans(audio_renderer, &info);

    bool setup_done = false;
    fbl::RefPtr<AudioPacketRef> pkt_ref;

    bool release_audio_renderer_packet;
    while (true) {
      release_audio_renderer_packet = false;
      // Try to grab the packet queue's front. If it has been flushed since the last time we grabbed
      // it, reset our mixer's internal filter state.
      bool was_flushed;
      pkt_ref = packet_link->LockPendingQueueFront(&was_flushed);
      if (was_flushed) {
        info.mixer->Reset();
      }

      // If the queue is empty, then we are done.
      if (!pkt_ref) {
        break;
      }

      // If we have not set up for this renderer yet, do so. If the setup fails for any reason, stop
      // processing packets for this renderer.
      if (!setup_done) {
        setup_done = (task_type == TaskType::Mix) ? SetupMix(audio_renderer, &info)
                                                  : SetupTrim(audio_renderer, &info);
        if (!setup_done) {
          // Clear our ramps, if we exit with error?
          break;
        }
      }

      // Now process the packet at the front of the renderer's queue. If the packet has been
      // entirely consumed, pop it off the front and proceed to the next. Otherwise, we are done.
      release_audio_renderer_packet = (task_type == TaskType::Mix)
                                          ? ProcessMix(audio_renderer, &info, pkt_ref)
                                          : ProcessTrim(audio_renderer, &info, pkt_ref);

      // If we have mixed enough destination frames, we are done with this mix, regardless of what
      // we should now do with the source packet.
      if ((task_type == TaskType::Mix) &&
          (cur_mix_job_.frames_produced == cur_mix_job_.buf_frames)) {
        break;
      }
      // If we still need to produce more destination data, but could not complete this source
      // packet (we're paused, or the packet is in the future), then we are done.
      if (!release_audio_renderer_packet) {
        break;
      }
      // We did consume this entire source packet, and we should keep mixing.
      pkt_ref.reset();
      packet_link->UnlockPendingQueueFront(release_audio_renderer_packet);
    }

    // Unlock queue (completing packet if needed) and proceed to the next source.
    pkt_ref.reset();
    packet_link->UnlockPendingQueueFront(release_audio_renderer_packet);

    // Note: there is no point in doing this for Trim tasks, but it doesn't hurt anything, and it's
    // easier than adding another function to ForEachLink to run after each renderer is processed,
    // just to set this flag.
    cur_mix_job_.accumulate = true;
  }
}

bool AudioOutput::SetupMix(const fbl::RefPtr<AudioRendererImpl>& audio_renderer,
                           Bookkeeping* info) {
  // If we need to recompose our transformation from destination frame space to source fractional
  // frames, do so now.
  FXL_DCHECK(info);
  UpdateDestTrans(cur_mix_job_, info);
  cur_mix_job_.frames_produced = 0;

  return true;
}

bool AudioOutput::ProcessMix(const fbl::RefPtr<AudioRendererImpl>& audio_renderer,
                             Bookkeeping* info, const fbl::RefPtr<AudioPacketRef>& packet) {
  // Bookkeeping should contain: the rechannel matrix (eventually).

  // Sanity check our parameters.
  FXL_DCHECK(info);
  FXL_DCHECK(packet);

  // We had better have a valid job, or why are we here?
  FXL_DCHECK(cur_mix_job_.buf_frames);
  FXL_DCHECK(cur_mix_job_.frames_produced <= cur_mix_job_.buf_frames);

  // We also must have selected a mixer, or we are in trouble.
  FXL_DCHECK(info->mixer);
  Mixer& mixer = *(info->mixer);

  // If the renderer is currently paused, subject_delta (not just step_size) is zero. This packet
  // may be relevant eventually, but currently it contributes nothing. Tell ForEachLink we are done,
  // but hold the packet for now.
  if (!info->dest_frames_to_frac_source_frames.subject_delta()) {
    return false;
  }

  // Have we produced enough? If so, hold this packet and move to next renderer.
  if (cur_mix_job_.frames_produced >= cur_mix_job_.buf_frames) {
    return false;
  }

  // At this point we know we need to consume some source data, but we don't yet know how much.
  // Here is how many destination frames we still need to produce, for this mix job.
  uint32_t dest_frames_left = cur_mix_job_.buf_frames - cur_mix_job_.frames_produced;
  float* buf = mix_buf_.get() + (cur_mix_job_.frames_produced * output_producer_->channels());

  // Calculate this job's first and last sampling points, in source sub-frames. Use timestamps for
  // the first and last dest frames we need, translated into the source (frac_frame) timeline.
  int64_t frac_source_for_first_mix_job_frame = info->dest_frames_to_frac_source_frames(
      cur_mix_job_.start_pts_of + cur_mix_job_.frames_produced);

  // This represents (in the frac_frame source timeline) the time of the LAST dest frame we need.
  // Without the "-1", this would be the first destination frame of the NEXT job.
  int64_t frac_source_for_final_mix_job_frame =
      frac_source_for_first_mix_job_frame +
      info->dest_frames_to_frac_source_frames.rate().Scale(dest_frames_left - 1);

  // If packet has no frames, there's no need to mix it; it may be skipped.
  if (packet->end_pts() == packet->start_pts()) {
    AUD_VLOG_OBJ(TRACE, audio_renderer.get()) << " skipping an empty packet!";
    return true;
  }

  FXL_DCHECK((packet->end_pts() - packet->start_pts()) >= Mixer::FRAC_ONE);

  // The above two calculated values characterize our demand. Now reason about our supply. Calculate
  // the actual first and final frame times in the source packet.
  int64_t frac_source_for_first_packet_frame = packet->start_pts();
  int64_t frac_source_for_final_packet_frame = packet->end_pts() - Mixer::FRAC_ONE;

  // If this source packet's final audio frame occurs before our filter's negative edge, centered at
  // our first sampling point, then this packet is entirely in the past and may be skipped.
  // Returning true means we're done with the packet (it can be completed) and we would like another
  if (frac_source_for_final_packet_frame <
      (frac_source_for_first_mix_job_frame - mixer.neg_filter_width())) {
    auto clock_mono_late = info->clock_mono_to_frac_source_frames.rate().Inverse().Scale(
        frac_source_for_first_mix_job_frame - mixer.neg_filter_width() -
        frac_source_for_first_packet_frame);
    AUD_LOG_OBJ(ERROR, audio_renderer.get())
        << ", skipping packet [" << frac_source_for_first_packet_frame << ","
        << frac_source_for_final_packet_frame << "]: missed mix-start "
        << frac_source_for_first_mix_job_frame - mixer.neg_filter_width() << " by "
        << static_cast<double>(clock_mono_late) / ZX_MSEC(1) << " ms";
    return true;
  }

  // If this source packet's first audio frame occurs after our filter's positive edge, centered at
  // our final sampling point, then this packet is entirely in the future and should be held.
  // Returning false (based on requirement that packets must be presented in timestamp-chronological
  // order) means that we have consumed all of the available packet "supply" as we can at this time.
  if (frac_source_for_first_packet_frame >
      (frac_source_for_final_mix_job_frame + mixer.pos_filter_width())) {
    return false;
  }

  // If neither of the above, then evidently this source packet intersects our mixer's filter.
  // Compute the offset into the dest buffer where our first generated sample should land, and the
  // offset into the source packet where we should start sampling.
  int64_t frac_source_offset_64 =
      frac_source_for_first_mix_job_frame - frac_source_for_first_packet_frame;
  int64_t dest_offset_64 = 0;
  int64_t frac_source_pos_edge_first_mix_frame =
      frac_source_for_first_mix_job_frame + mixer.pos_filter_width();

  // If the packet's first frame comes after the filter window's positive edge,
  // then we should skip some frames in the destination buffer before starting to produce data.
  if (frac_source_for_first_packet_frame > frac_source_pos_edge_first_mix_frame) {
    const TimelineRate& dest_to_src = info->dest_frames_to_frac_source_frames.rate();
    // The dest_buffer offset is based on the distance from mix job start to packet start (measured
    // in frac_frames), converted into frames in the destination timeline. As we scale the
    // frac_frame delta into dest frames, we want to "round up" any subframes that are present; any
    // src subframes should push our dest frame up to the next integer. To do this, we subtract a
    // single subframe (guaranteeing that the zero-fraction src case will truncate down), then scale
    // the src delta to dest frames (which effectively truncates any resultant fraction in the
    // computed dest frame), then add an additional 'round-up' frame (to account for initial
    // subtract). Because we entered this IF in the first place, we have at least some fractional
    // src delta, thus dest_offset_64 is guaranteed to become greater than zero.
    dest_offset_64 = dest_to_src.Inverse().Scale(frac_source_for_first_packet_frame -
                                                 frac_source_pos_edge_first_mix_frame - 1) +
                     1;
    FXL_DCHECK(dest_offset_64 > 0);

    frac_source_offset_64 += dest_to_src.Scale(dest_offset_64);

    if (abs(frac_source_offset_64) >= (Mixer::FRAC_ONE >> 1)) {
      AUD_LOG_OBJ(WARNING, audio_renderer.get())
          << " skipped " << dest_offset_64 << " output frames; frac_source_offset is 0x" << std::hex
          << frac_source_offset_64;
    } else {
      AUD_VLOG_OBJ(TRACE, audio_renderer.get())
          << " skipped " << dest_offset_64 << " output frames to align with source timestamp";
    }
  }

  FXL_DCHECK(dest_offset_64 >= 0);
  FXL_DCHECK(dest_offset_64 < static_cast<int64_t>(dest_frames_left));
  auto dest_offset = static_cast<uint32_t>(dest_offset_64);

  FXL_DCHECK(frac_source_offset_64 <= std::numeric_limits<int32_t>::max());
  FXL_DCHECK(frac_source_offset_64 >= std::numeric_limits<int32_t>::min());
  auto frac_source_offset = static_cast<int32_t>(frac_source_offset_64);

  // Looks like we are ready to go. Mix.
  FXL_DCHECK(packet->frac_frame_len() <=
             static_cast<uint32_t>(std::numeric_limits<int32_t>::max()));

  FXL_DCHECK(frac_source_offset + mixer.pos_filter_width() >= 0);
  bool consumed_source = false;
  if (frac_source_offset + static_cast<int32_t>(mixer.pos_filter_width()) <
      static_cast<int32_t>(packet->frac_frame_len())) {
    // When calling Mix(), we communicate the resampling rate with three parameters. We augment
    // step_size with rate_modulo and denominator arguments that capture the remaining rate
    // component that cannot be expressed by a 19.13 fixed-point step_size. Note: step_size and
    // frac_source_offset use the same format -- they have the same limitations in what they can and
    // cannot communicate.
    //
    // For perfect position accuracy, just as we track incoming/outgoing fractional source offset,
    // we also need to track the ongoing subframe_position_modulo. This is now added to Mix() and
    // maintained across calls, but not initially set to any value other than zero. For now, we are
    // deferring that work, tracking it with MTWN-128.
    //
    // Q: Why did we solve this issue for Rate but not for initial Position?
    // A: We solved this issue for *rate* because its effect accumulates over time, causing clearly
    // measurable distortion that becomes crippling with larger jobs. For *position*, there is no
    // accumulated magnification over time -- in analyzing the distortion that this should cause,
    // mix job size affects the distortion's frequency but not its amplitude. We expect the effects
    // to be below audible thresholds. Until the effects are measurable and attributable to this
    // jitter, we will defer this work.
    //
    // TODO(mpuryear): integrate bookkeeping into the Mixer itself (MTWN-129).

    auto prev_dest_offset = dest_offset;
    auto prev_frac_source_offset = frac_source_offset;

    // Check whether we are still ramping
    bool ramping = info->gain.IsRamping();
    if (ramping) {
      info->gain.GetScaleArray(info->scale_arr.get(),
                               std::min(dest_frames_left - dest_offset, Bookkeeping::kScaleArrLen),
                               cur_mix_job_.local_to_output->rate());
    }

    consumed_source = info->mixer->Mix(buf, dest_frames_left, &dest_offset, packet->payload(),
                                       packet->frac_frame_len(), &frac_source_offset,
                                       cur_mix_job_.accumulate, info);
    FXL_DCHECK(dest_offset <= dest_frames_left);
    AUD_VLOG_OBJ(SPEW, this) << " consumed from " << std::hex << std::setw(8)
                             << prev_frac_source_offset << " to " << std::setw(8)
                             << frac_source_offset << ", of " << std::setw(8)
                             << packet->frac_frame_len();

    // If src is ramping, advance by delta of dest_offset
    if (ramping) {
      info->gain.Advance(dest_offset - prev_dest_offset, cur_mix_job_.local_to_output->rate());
    }
  } else {
    consumed_source = true;
    FXL_LOG(ERROR) << "Skipping packet -- frac_source_offset 0x" << std::hex << frac_source_offset
                   << " exceeded final src frame's positive window 0x"
                   << (static_cast<int32_t>(packet->frac_frame_len()) - Mixer::FRAC_ONE +
                       mixer.pos_filter_width());
  }

  if (consumed_source) {
    FXL_DCHECK(frac_source_offset + info->mixer->pos_filter_width() >= packet->frac_frame_len());
  }

  cur_mix_job_.frames_produced += dest_offset;

  FXL_DCHECK(cur_mix_job_.frames_produced <= cur_mix_job_.buf_frames);
  return consumed_source;
}

bool AudioOutput::SetupTrim(const fbl::RefPtr<AudioRendererImpl>& audio_renderer,
                            Bookkeeping* info) {
  // Compute the cutoff time used to decide whether to trim packets. ForEachLink has already updated
  // our transformation, no need for us to do so here.
  FXL_DCHECK(info);

  int64_t local_now_ticks = (fxl::TimePoint::Now() - fxl::TimePoint()).ToNanoseconds();

  // RateControlBase guarantees that the transformation into the media timeline is never singular.
  // If a forward transformation fails it must be because of overflow, which should be impossible
  // unless user defined a playback rate where the ratio of media-ticks-to-local-ticks is greater
  // than one.
  trim_threshold_ = info->clock_mono_to_frac_source_frames(local_now_ticks);

  return true;
}

bool AudioOutput::ProcessTrim(const fbl::RefPtr<AudioRendererImpl>& audio_renderer,
                              Bookkeeping* info, const fbl::RefPtr<AudioPacketRef>& pkt_ref) {
  FXL_DCHECK(pkt_ref);

  // If the presentation end of this packet is in the future, stop trimming.
  if (pkt_ref->end_pts() > trim_threshold_) {
    return false;
  }

  return true;
}

void AudioOutput::UpdateSourceTrans(const fbl::RefPtr<AudioRendererImpl>& audio_renderer,
                                    Bookkeeping* bk) {
  FXL_DCHECK(audio_renderer != nullptr);
  uint32_t gen = bk->source_trans_gen_id;

  audio_renderer->SnapshotCurrentTimelineFunction(zx::clock::get_monotonic().get(),
                                                  &bk->clock_mono_to_frac_source_frames, &gen);

  // If local->media transformation hasn't changed since last time, we're done.
  if (bk->source_trans_gen_id == gen) {
    return;
  }

  // Transformation has changed. Update gen; invalidate dest-to-src generation.
  bk->source_trans_gen_id = gen;
  bk->dest_trans_gen_id = kInvalidGenerationId;
}

void AudioOutput::UpdateDestTrans(const MixJob& job, Bookkeeping* bk) {
  // We should only be here if we have a valid mix job. This means a job which supplies a valid
  // transformation from local time to output frames.
  FXL_DCHECK(job.local_to_output);
  FXL_DCHECK(job.local_to_output_gen != kInvalidGenerationId);

  // If generations match, don't re-compute -- just use what we have already.
  if (bk->dest_trans_gen_id == job.local_to_output_gen) {
    return;
  }

  // Assert we can map from local time to fractional renderer frames.
  FXL_DCHECK(bk->source_trans_gen_id != kInvalidGenerationId);

  // Combine the job-supplied local-to-output transformation, with the renderer-supplied mapping of
  // local-to-input-subframe, to produce a transformation which maps from output frames to
  // fractional input frames.
  TimelineFunction& dest = bk->dest_frames_to_frac_source_frames;
  dest = bk->clock_mono_to_frac_source_frames * job.local_to_output->Inverse();

  // Finally, compute the step size in subframes. IOW, every time we move forward one output frame,
  // how many input subframes should we consume. Don't bother doing the multiplications if already
  // we know the numerator is zero.
  FXL_DCHECK(dest.rate().reference_delta());
  if (!dest.rate().subject_delta()) {
    bk->step_size = 0;
    bk->denominator = 0;  // shouldn't also need to clear rate_mod and pos_mod
  } else {
    int64_t tmp_step_size = dest.rate().Scale(1);

    FXL_DCHECK(tmp_step_size >= 0);
    FXL_DCHECK(tmp_step_size <= std::numeric_limits<uint32_t>::max());

    bk->step_size = static_cast<uint32_t>(tmp_step_size);
    bk->denominator = bk->SnapshotDenominatorFromDestTrans();
    bk->rate_modulo = dest.rate().subject_delta() - (bk->denominator * bk->step_size);
  }

  // Done, update our dest_trans generation.
  bk->dest_trans_gen_id = job.local_to_output_gen;
}

}  // namespace media::audio