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}
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@ -24,7 +24,7 @@
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namespace webrtc {
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DecisionLogic* DecisionLogic::Create(int fs_hz,
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int output_size_samples,
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size_t output_size_samples,
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NetEqPlayoutMode playout_mode,
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DecoderDatabase* decoder_database,
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const PacketBuffer& packet_buffer,
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@ -56,7 +56,7 @@ DecisionLogic* DecisionLogic::Create(int fs_hz,
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}
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DecisionLogic::DecisionLogic(int fs_hz,
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int output_size_samples,
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size_t output_size_samples,
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NetEqPlayoutMode playout_mode,
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DecoderDatabase* decoder_database,
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const PacketBuffer& packet_buffer,
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@ -95,7 +95,7 @@ void DecisionLogic::SoftReset() {
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timescale_hold_off_ = kMinTimescaleInterval;
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}
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void DecisionLogic::SetSampleRate(int fs_hz, int output_size_samples) {
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void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
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// TODO(hlundin): Change to an enumerator and skip assert.
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assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
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fs_mult_ = fs_hz / 8000;
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@ -104,7 +104,7 @@ void DecisionLogic::SetSampleRate(int fs_hz, int output_size_samples) {
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Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
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const Expand& expand,
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int decoder_frame_length,
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size_t decoder_frame_length,
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const RTPHeader* packet_header,
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Modes prev_mode,
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bool play_dtmf, bool* reset_decoder) {
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@ -123,9 +123,9 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
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}
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}
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const int samples_left = static_cast<int>(
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sync_buffer.FutureLength() - expand.overlap_length());
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const int cur_size_samples =
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const size_t samples_left =
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sync_buffer.FutureLength() - expand.overlap_length();
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const size_t cur_size_samples =
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samples_left + packet_buffer_.NumSamplesInBuffer(decoder_database_,
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decoder_frame_length);
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LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
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@ -153,9 +153,10 @@ void DecisionLogic::ExpandDecision(Operations operation) {
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}
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}
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void DecisionLogic::FilterBufferLevel(int buffer_size_samples,
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void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
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Modes prev_mode) {
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const int elapsed_time_ms = output_size_samples_ / (8 * fs_mult_);
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const int elapsed_time_ms =
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static_cast<int>(output_size_samples_ / (8 * fs_mult_));
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delay_manager_->UpdateCounters(elapsed_time_ms);
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// Do not update buffer history if currently playing CNG since it will bias
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@ -164,7 +165,7 @@ void DecisionLogic::FilterBufferLevel(int buffer_size_samples,
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buffer_level_filter_->SetTargetBufferLevel(
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delay_manager_->base_target_level());
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int buffer_size_packets = 0;
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size_t buffer_size_packets = 0;
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if (packet_length_samples_ > 0) {
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// Calculate size in packets.
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buffer_size_packets = buffer_size_samples / packet_length_samples_;
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