Adding new function ReadFrames() that returns the audio frame in planar

media/base/audio_buffer.cc

 // Copyright 2013 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_buffer.h"
 
 #include "base/logging.h"
 #include "media/base/audio_bus.h"
 #include "media/base/buffers.h"
 #include "media/base/limits.h"
@@ skipping 225 matching lines @@
   DCHECK(sample_format_ == kSampleFormatU8 ||
          sample_format_ == kSampleFormatS16 ||
          sample_format_ == kSampleFormatS32);
   int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_);
   int frame_size = channel_count_ * bytes_per_channel;
   const uint8* source_data = data_.get() + source_frame_offset * frame_size;
   dest->FromInterleavedPartial(
       source_data, dest_frame_offset, frames_to_copy, bytes_per_channel);
 }
 
+static const int32 kUint8Bias = 128;
+
+static inline int32 ConvertU8ToS32(uint8 value) {
+  return (static_cast<int32>(value) - kUint8Bias) << 24;
+}
+
+static inline int32 ConvertS16ToS32(int16 value) {
+  return static_cast<int32>(value) << 16;
+}
+
+static inline int32 ConvertF32ToS32(float value) {
+  return static_cast<int32>(value < 0
+                                ? value * ~std::numeric_limits<int32>::min()

[DaleCurtis, 2014/09/25 22:30:44]

I don't think this does what you want; e.g., ~(-5) == 4, instead try -value
without the ~ on the min.

+                                : value * std::numeric_limits<int32>::max());
+}
+
+void AudioBuffer::ReadFramesInterleavedS32(int frames_to_copy,
+                                           int32* dest_data) {
+  DCHECK_EQ(frames_to_copy, channel_count_ * adjusted_frame_count_);
+
+  if (sample_format_ == kSampleFormatU8) {
+    // Unsigned 8-bit w/ bias of 128.
+    const uint8* source_data =
+        reinterpret_cast<const uint8*>(channel_data_[0]) + trim_start_;
+
+    for (int i = 0; i < adjusted_frame_count_ * channel_count_; ++i) {
+      dest_data[i] = ConvertU8ToS32(source_data[i]);
+    }
+  } else if (sample_format_ == kSampleFormatS16) {
+    // Format is interleaved signed16. Convert each value into int32 and insert
+    // into output channel data.
+    const int16* source_data =
+        reinterpret_cast<const int16*>(channel_data_[0]) + trim_start_;
+
+    for (int i = 0; i < adjusted_frame_count_ * channel_count_; ++i) {
+      dest_data[i] = ConvertS16ToS32(source_data[i]);
+    }
+  } else if (sample_format_ == kSampleFormatS32) {
+    // Format is interleaved signed32; just copy the data.
+    const int32* source_data =
+        reinterpret_cast<const int32*>(channel_data_[0]) + trim_start_;
+    memcpy(dest_data,
+           source_data,
+           adjusted_frame_count_ * channel_count_ * sizeof(int32));
+  } else if (sample_format_ == kSampleFormatF32) {
+    // Format is interleaved float. Convert each value into int32 and insert
+    // into output channel data.
+    const float* source_data =
+        reinterpret_cast<const float*>(channel_data_[0]) + trim_start_;
+    for (int i = 0; i < adjusted_frame_count_ * channel_count_; ++i) {
+      dest_data[i] = ConvertF32ToS32(source_data[i]);
+    }
+  } else if (sample_format_ == kSampleFormatPlanarS16) {
+    // Format is planar signed 16 bit. Convert each value into int32 and insert
+    // into output channel data.
+    int dest_data_frame_size = channel_count_;
+    for (int ch = 0; ch < channel_count_; ++ch) {
+      const int16* source_data =
+          reinterpret_cast<const int16*>(channel_data_[ch]) + trim_start_;
+
+      for (int i = 0; i < adjusted_frame_count_; ++i) {
+        dest_data[ch + (i * dest_data_frame_size)] =

[DaleCurtis, 2014/09/25 22:30:44]

A paired incrementing offset is more efficient than computing this
multiplication every time, see AudioBus::FromInterleavedInternal. Minor in any
case, so up to you.

+            ConvertS16ToS32(source_data[i]);
+      }
+    }
+  } else if (sample_format_ == kSampleFormatPlanarF32) {
+    // Format is planar float. Convert each value into int32 and insert into
+    // output channel data.
+    int dest_data_frame_size = channel_count_;
+    for (int ch = 0; ch < channel_count_; ++ch) {
+      const float* source_data =
+          reinterpret_cast<const float*>(channel_data_[ch]) + trim_start_;
+
+      for (int i = 0; i < adjusted_frame_count_; ++i) {
+        dest_data[ch + (i * dest_data_frame_size)] =

[DaleCurtis, 2014/09/25 22:30:44]

Ditto.

+            ConvertF32ToS32(source_data[i]);
+      }
+    }
+  } else {
+    NOTREACHED();

[DaleCurtis, 2014/09/25 22:30:44]

If you prefer you could use case statement since it explictly has to list all
possible types.

+  }
+}
+
 void AudioBuffer::TrimStart(int frames_to_trim) {
   CHECK_GE(frames_to_trim, 0);
   CHECK_LE(frames_to_trim, adjusted_frame_count_);
 
   // Adjust the number of frames in this buffer and where the start really is.
   adjusted_frame_count_ -= frames_to_trim;
   trim_start_ += frames_to_trim;
 
   // Adjust timestamp_ and duration_ to reflect the smaller number of frames.
   const base::TimeDelta old_duration = duration_;
@@ skipping 47 matching lines @@
     }
   } else {
     CHECK_EQ(frames_to_copy, 0);
   }
 
   // Trim the leftover data off the end of the buffer and update duration.
   TrimEnd(frames_to_trim);
 }
 
 }  // namespace media