Switch to standard integer types in media/.

media/base/audio_splicer.cc

 // 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 "media/base/audio_splicer.h"
 
 #include <cstdlib>
 #include <deque>
 
 #include "base/logging.h"
@@ skipping 52 matching lines @@
   AudioStreamSanitizer(int samples_per_second,
                        const scoped_refptr<MediaLog>& media_log);
   ~AudioStreamSanitizer();
 
   // Resets the sanitizer state by clearing the output buffers queue, and
   // resetting the timestamp helper.
   void Reset();
 
   // Similar to Reset(), but initializes the timestamp helper with the given
   // parameters.
-  void ResetTimestampState(int64 frame_count, base::TimeDelta base_timestamp);
+  void ResetTimestampState(int64_t frame_count, base::TimeDelta base_timestamp);
 
   // Adds a new buffer full of samples or end of stream buffer to the splicer.
   // Returns true if the buffer was accepted. False is returned if an error
   // occurred.
   bool AddInput(const scoped_refptr<AudioBuffer>& input);
 
   // Returns true if the sanitizer has a buffer to return.
   bool HasNextBuffer() const;
 
   // Removes the next buffer from the output buffer queue and returns it; should
@@ skipping 33 matching lines @@
     int samples_per_second,
     const scoped_refptr<MediaLog>& media_log)
     : output_timestamp_helper_(samples_per_second), media_log_(media_log) {}
 
 AudioStreamSanitizer::~AudioStreamSanitizer() {}
 
 void AudioStreamSanitizer::Reset() {
   ResetTimestampState(0, kNoTimestamp());
 }
 
-void AudioStreamSanitizer::ResetTimestampState(int64 frame_count,
+void AudioStreamSanitizer::ResetTimestampState(int64_t frame_count,
                                                base::TimeDelta base_timestamp) {
   output_buffers_.clear();
   received_end_of_stream_ = false;
   output_timestamp_helper_.SetBaseTimestamp(base_timestamp);
   if (frame_count > 0)
     output_timestamp_helper_.AddFrames(frame_count);
 }
 
 bool AudioStreamSanitizer::AddInput(const scoped_refptr<AudioBuffer>& input) {
   DCHECK(!received_end_of_stream_ || input->end_of_stream());
@@ skipping 410 matching lines @@
     AccurateTrimStart(frames_to_trim, remainder, output_ts_helper);
     CHECK(output_sanitizer_->AddInput(remainder));
   }
 
   // Transfer all remaining buffers out and reset once empty.
   CHECK(post_splice_sanitizer_->DrainInto(output_sanitizer_.get()));
   post_splice_sanitizer_->Reset();
 }
 
 }  // namespace media