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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@ -106,7 +106,7 @@ class NetEqImpl : public webrtc::NetEq {
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// Returns kOK on success, or kFail in case of an error.
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int GetAudio(size_t max_length,
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int16_t* output_audio,
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int* samples_per_channel,
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size_t* samples_per_channel,
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int* num_channels,
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NetEqOutputType* type) override;
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@ -203,9 +203,9 @@ class NetEqImpl : public webrtc::NetEq {
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protected:
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static const int kOutputSizeMs = 10;
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static const int kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
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static const size_t kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
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// TODO(hlundin): Provide a better value for kSyncBufferSize.
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static const int kSyncBufferSize = 2 * kMaxFrameSize;
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static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
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// Inserts a new packet into NetEq. This is used by the InsertPacket method
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// above. Returns 0 on success, otherwise an error code.
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@ -225,7 +225,7 @@ class NetEqImpl : public webrtc::NetEq {
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// Returns 0 on success, otherwise an error code.
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int GetAudioInternal(size_t max_length,
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int16_t* output,
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int* samples_per_channel,
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size_t* samples_per_channel,
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int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Provides a decision to the GetAudioInternal method. The decision what to
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@ -318,7 +318,7 @@ class NetEqImpl : public webrtc::NetEq {
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// |required_samples| samples. The packets are inserted into |packet_list|.
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// Returns the number of samples that the packets in the list will produce, or
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// -1 in case of an error.
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int ExtractPackets(int required_samples, PacketList* packet_list)
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int ExtractPackets(size_t required_samples, PacketList* packet_list)
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EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Resets various variables and objects to new values based on the sample rate
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@ -375,8 +375,8 @@ class NetEqImpl : public webrtc::NetEq {
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StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
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int fs_hz_ GUARDED_BY(crit_sect_);
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int fs_mult_ GUARDED_BY(crit_sect_);
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int output_size_samples_ GUARDED_BY(crit_sect_);
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int decoder_frame_length_ GUARDED_BY(crit_sect_);
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size_t output_size_samples_ GUARDED_BY(crit_sect_);
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size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
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Modes last_mode_ GUARDED_BY(crit_sect_);
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rtc::scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
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size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
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