From 363b85226e822128815fee268358f93d5259a33e Mon Sep 17 00:00:00 2001
From: samrobinson <samrobinson@google.com>
Date: Mon, 20 Feb 2023 11:43:09 +0000
Subject: [PATCH] Pass AudioFormat to ASP#computeNextEncoderInputBufferTimeUs.

This method uses sampleRate, channelCount and pcmEncoding, so passing
AudioFormat is easier.

This will lead into a future change that builds the
encoderInputAudioFormat from encoder.getConfigurationFormat()

PiperOrigin-RevId: 510956177
---
 .../exoplayer2/transformer/AudioSamplePipeline.java      | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

diff --git a/library/transformer/src/main/java/com/google/android/exoplayer2/transformer/AudioSamplePipeline.java b/library/transformer/src/main/java/com/google/android/exoplayer2/transformer/AudioSamplePipeline.java
index f598421fb4..fe4937129e 100644
--- a/library/transformer/src/main/java/com/google/android/exoplayer2/transformer/AudioSamplePipeline.java
+++ b/library/transformer/src/main/java/com/google/android/exoplayer2/transformer/AudioSamplePipeline.java
@@ -326,9 +326,7 @@ import org.checkerframework.dataflow.qual.Pure;
     encoderInputBufferData.put(inputBuffer);
     encoderInputBuffer.timeUs = nextEncoderInputBufferTimeUs;
     computeNextEncoderInputBufferTimeUs(
-        /* bytesWritten= */ encoderInputBufferData.position(),
-        encoderInputAudioFormat.bytesPerFrame,
-        encoderInputAudioFormat.sampleRate);
+        /* bytesWritten= */ encoderInputBufferData.position(), encoderInputAudioFormat);
     encoderInputBuffer.setFlags(0);
     encoderInputBuffer.flip();
     inputBuffer.limit(bufferLimit);
@@ -355,15 +353,14 @@ import org.checkerframework.dataflow.qual.Pure;
     return transformationRequest.buildUpon().setAudioMimeType(actualFormat.sampleMimeType).build();
   }
 
-  private void computeNextEncoderInputBufferTimeUs(
-      long bytesWritten, int bytesPerFrame, int sampleRate) {
+  private void computeNextEncoderInputBufferTimeUs(long bytesWritten, AudioFormat audioFormat) {
     // The calculation below accounts for remainders and rounding. Without that it corresponds to
     // the following:
     // bufferDurationUs = numberOfFramesInBuffer * sampleDurationUs
     //     where numberOfFramesInBuffer = bytesWritten / bytesPerFrame
     //     and   sampleDurationUs       = C.MICROS_PER_SECOND / sampleRate
     long numerator = bytesWritten * C.MICROS_PER_SECOND + encoderBufferDurationRemainder;
-    long denominator = (long) bytesPerFrame * sampleRate;
+    long denominator = (long) audioFormat.bytesPerFrame * audioFormat.sampleRate;
     long bufferDurationUs = numerator / denominator;
     encoderBufferDurationRemainder = numerator - bufferDurationUs * denominator;
     if (encoderBufferDurationRemainder > 0) { // Ceil division result.